Release v4.0-b1, 4.0 beta1, v4.0.206, 144126 lines.

* fix memory leak when replace rtp packet in cache.

.circleci/config.yml

@@ -1,20 +0,0 @@
-version: 2
-jobs:
-  build:
-    docker:
-      - image: ossrs/srs:dev
-    steps:
-      - checkout
-      - run: cd trunk && ./configure --with-utest && make
-  test:
-    docker:
-      - image: ossrs/srs:dev
-    steps:
-      - checkout
-      - run: cd trunk && ./configure --with-utest --gcov && make && ./objs/srs_utest && bash auto/coverage.sh
-workflows:
-  version: 2
-  build_and_test:
-    jobs:
-      - build
-      - test

.github/FUNDING.yml

@@ -0,0 +1,12 @@
+# These are supported funding model platforms
+
+github: # Replace with up to 4 GitHub Sponsors-enabled usernames e.g., [user1, user2]
+patreon: # Replace with a single Patreon username
+open_collective: srs-server
+ko_fi: # Replace with a single Ko-fi username
+tidelift: # Replace with a single Tidelift platform-name/package-name e.g., npm/babel
+community_bridge: # Replace with a single Community Bridge project-name e.g., cloud-foundry
+liberapay: # Replace with a single Liberapay username
+issuehunt: # Replace with a single IssueHunt username
+otechie: # Replace with a single Otechie username
+custom: # Replace with up to 4 custom sponsorship URLs e.g., ['link1', 'link2']

.github/ISSUE_TEMPLATE

@@ -3,22 +3,30 @@ name: File bug
 about: File bug to improve SRS
 title: ''
 labels: ''
-assignees: ''
+assignees: 'winlinvip'
 
 ---
 
+> 注意：提问前，请先看FAQ(Please read FAQ before file an issue) https://github.com/ossrs/srs/issues/2716
+
 **描述(Description)**
 
 > 描述你遇到了什么问题(Please description your issue here)
 
 1. SRS版本(Version): `xxxxxx`
+
 1. SRS的日志如下(Log):
 ```
+
 xxxxxxxxxxxx
+
 ```
+
 1. SRS的配置如下(Config):
 ```
+
 xxxxxxxxxxxx
+
 ```
 
 **重现(Replay)**

.github/workflows/codeql-analysis.yml

@@ -0,0 +1,31 @@
+name: "CodeQL"
+
+# @see https://docs.github.com/en/actions/reference/workflow-syntax-for-github-actions#onpushpull_requestbranchestags
+on: [push, pull_request]
+
+jobs:
+  analyze:
+    name: actions-codeql-analyze
+    runs-on: ubuntu-latest
+
+    strategy:
+      fail-fast: false
+      matrix:
+        language: [ 'cpp' ]
+
+    steps:
+      - name: Checkout repository
+        uses: actions/checkout@v2
+
+      # Initializes the CodeQL tools for scanning.
+      - name: Initialize CodeQL
+        uses: github/codeql-action/init@v1
+        with:
+          languages: ${{ matrix.language }}
+
+      - name: Build SRS
+        run: |
+         cd trunk && ./configure && make
+
+      - name: Perform CodeQL Analysis
+        uses: github/codeql-action/analyze@v1

.github/workflows/release.yml

@@ -0,0 +1,167 @@
+name: "Release"
+
+# @see https://docs.github.com/en/actions/reference/workflow-syntax-for-github-actions#onpushpull_requestbranchestags
+on:
+  push:
+    tags:
+      - v4*
+
+jobs:
+  k8s:
+    name: release-k8s
+    runs-on: ubuntu-20.04
+
+    steps:
+      ################################################################
+      # Git checkout
+      - name: Checkout repository
+        uses: actions/checkout@v2
+      # The github.ref is, for example, refs/tags/v4.0.145 or refs/tags/v4.0-r8
+      # Generate variables like:
+      #   SRS_TAG=v4.0-r8
+      #   SRS_TAG=v4.0.145
+      #   SRS_VERSION=4.0.145
+      #   SRS_VERSION=4.0-r8
+      #   SRS_MAJOR=4
+      # @see https://docs.github.com/en/actions/reference/workflow-commands-for-github-actions#setting-an-environment-variable
+      - name: Generate varaiables
+        run: |
+          SRS_TAG=$(echo ${{ github.ref }}| awk -F '/' '{print $3}')
+          echo "SRS_TAG=$SRS_TAG" >> $GITHUB_ENV
+          SRS_VERSION=$(echo ${SRS_TAG}| sed 's/^v//g')
+          echo "SRS_VERSION=$SRS_VERSION" >> $GITHUB_ENV
+          SRS_MAJOR=$(echo $SRS_TAG| cut -c 2)
+          echo "SRS_MAJOR=$SRS_MAJOR" >> $GITHUB_ENV
+      # Create source tar for release
+      # Generate variables like:
+      #   SRS_SOURCE_TAR=srs-server-4.0.145.tar.gz
+      #   SRS_SOURCE_MD5=83e38700a80a26e30b2df054e69956e5
+      - name: Create source tar.gz
+        run: |
+          DEST_DIR=srs-server-$SRS_VERSION && mkdir -p $DEST_DIR &&
+          cp README.md $DEST_DIR && cp LICENSE $DEST_DIR && cp -R trunk $DEST_DIR/trunk &&
+          (cd $DEST_DIR/trunk/3rdparty && rm -rf *.zip openssl-*.gz srs-bench) &&
+          tar zcf ${DEST_DIR}.tar.gz ${DEST_DIR} && du -sh ${DEST_DIR}* && rm -rf ${DEST_DIR} &&
+          echo "SRS_SOURCE_TAR=${DEST_DIR}.tar.gz" >> $GITHUB_ENV &&
+          echo "SRS_SOURCE_MD5=$(md5sum ${DEST_DIR}.tar.gz| awk '{print $1}')" >> $GITHUB_ENV
+      # Create package tar for release
+      # Generate variables like:
+      #   SRS_PACKAGE_ZIP=SRS-CentOS7-x86_64-4.0.145.zip
+      #   SRS_PACKAGE_MD5=3880a26e30b283edf05700a4e69956e5
+      - name: Create package zip
+        run: |
+          docker build --tag srs:pkg --build-arg version=$SRS_VERSION -f trunk/Dockerfile.pkg . &&
+          SRS_PACKAGE_ZIP=SRS-CentOS7-x86_64-$SRS_VERSION.zip &&
+          docker run --rm -v $(pwd):/output srs:pkg cp objs/$SRS_PACKAGE_ZIP /output/ &&
+          du -sh $SRS_PACKAGE_ZIP &&
+          echo "SRS_PACKAGE_ZIP=$SRS_PACKAGE_ZIP" >> $GITHUB_ENV &&
+          echo "SRS_PACKAGE_MD5=$(md5sum $SRS_PACKAGE_ZIP| awk '{print $1}')" >> $GITHUB_ENV
+
+      ################################################################
+      # Tests
+      - name: Build test image
+        run: docker build --tag srs:test -f trunk/Dockerfile.test .
+      # For utest
+      - name: Run SRS utest
+        run: docker run --rm srs:test bash -c 'make utest && ./objs/srs_utest'
+      # For regression-test
+      - name: Run SRS regression-test
+        run: docker run --rm srs:test bash -c 'make && ./objs/srs -c conf/regression-test.conf && cd 3rdparty/srs-bench && make && ./objs/srs_test -test.v'
+
+      ################################################################
+      # Build
+      # Build SRS image
+      - name: Build SRS docker image
+        run: |
+          echo "Release ossrs/srs:$SRS_TAG"
+          docker build --tag ossrs/srs:$SRS_TAG -f trunk/Dockerfile .
+
+      ################################################################
+      # Docker
+      - name: Login docker hub
+        uses: docker/login-action@v1
+        with:
+          username: "${{ secrets.DOCKER_USERNAME }}"
+          password: "${{ secrets.DOCKER_PASSWORD }}"
+      - name: Push to docker hub
+        run: |
+          docker push ossrs/srs:$SRS_TAG
+          docker tag ossrs/srs:$SRS_TAG ossrs/srs:$SRS_MAJOR
+          docker push ossrs/srs:$SRS_MAJOR
+      # Aliyun ACR
+      - name: Login Aliyun docker hub
+        uses: aliyun/acr-login@v1
+        with:
+          login-server: https://registry.cn-hangzhou.aliyuncs.com
+          username: "${{ secrets.ACR_USERNAME }}"
+          password: "${{ secrets.ACR_PASSWORD }}"
+      - name: Push to Aliyun docker hub
+        run: |
+          docker tag ossrs/srs:$SRS_TAG registry.cn-hangzhou.aliyuncs.com/ossrs/srs:$SRS_TAG
+          docker push registry.cn-hangzhou.aliyuncs.com/ossrs/srs:$SRS_TAG
+          docker tag ossrs/srs:$SRS_TAG registry.cn-hangzhou.aliyuncs.com/ossrs/srs:$SRS_VERSION
+          docker push registry.cn-hangzhou.aliyuncs.com/ossrs/srs:$SRS_VERSION
+          docker tag ossrs/srs:$SRS_TAG registry.cn-hangzhou.aliyuncs.com/ossrs/srs:v$SRS_MAJOR
+          docker push registry.cn-hangzhou.aliyuncs.com/ossrs/srs:v$SRS_MAJOR
+          docker tag ossrs/srs:$SRS_TAG registry.cn-hangzhou.aliyuncs.com/ossrs/srs:$SRS_MAJOR
+          docker push registry.cn-hangzhou.aliyuncs.com/ossrs/srs:$SRS_MAJOR
+
+      ################################################################
+      # K8S
+      - name: Setup KUBCONFIG for Aliyun ACK
+        run: |-
+          KUBECONFIG=$RUNNER_TEMP/kubeconfig_$(date +%s)
+          echo "${{ secrets.KUBCONFIG }}" > $KUBECONFIG
+          echo "KUBECONFIG=$KUBECONFIG" >> $GITHUB_ENV
+      # K8S for SRS 4.0
+      - name: Release SRS 4.0 to Aliyun ACK
+        if: ${{ startsWith(github.ref, 'refs/tags/v4') }}
+        run: |-
+          kubectl set image deploy/srs4-deploy srs4=registry.cn-hangzhou.aliyuncs.com/ossrs/srs:$SRS_TAG
+          kubectl describe deploy/srs4-deploy
+
+      ################################################################
+      # Create release.
+      - name: Create release
+        id: create_release
+        uses: actions/create-release@v1
+        env:
+          GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}
+        with:
+          tag_name: ${{ github.ref }}
+          release_name: Release ${{ github.ref }}
+          body: |
+            ${{ github.event.head_commit.message }}
+            ## Resource
+            * Source: ${{ env.SRS_SOURCE_MD5 }} [${{ env.SRS_SOURCE_TAR }}](https://github.com/ossrs/srs/releases/download/${{ env.SRS_TAG }}/${{ env.SRS_SOURCE_TAR }})
+            * Binary: ${{ env.SRS_PACKAGE_MD5 }} [${{ env.SRS_PACKAGE_ZIP }}](https://github.com/ossrs/srs/releases/download/${{ env.SRS_TAG }}/${{ env.SRS_PACKAGE_ZIP }})
+            ## Docker
+            * China: [docker pull registry.cn-hangzhou.aliyuncs.com/ossrs/srs:${{ env.SRS_MAJOR }}](https://github.com/ossrs/srs/wiki/v4_CN_Home#docker)
+            * China: [docker pull registry.cn-hangzhou.aliyuncs.com/ossrs/srs:${{ env.SRS_TAG }}](https://github.com/ossrs/srs/wiki/v4_CN_Home#docker)
+            * Global: [docker pull ossrs/srs:${{ env.SRS_MAJOR }}](https://github.com/ossrs/srs/wiki/v4_EN_Home#docker)
+            * Global: [docker pull ossrs/srs:${{ env.SRS_TAG }}](https://github.com/ossrs/srs/wiki/v4_EN_Home#docker)
+            ## Doc
+            * [快速入门](https://github.com/ossrs/srs/wiki/v4_CN_Home#getting-started) or [Getting Started](https://github.com/ossrs/srs/wiki/v4_EN_Home#getting-started)
+            * [中文Wiki首页](https://github.com/ossrs/srs/wiki/v4_CN_Home) or [Wiki home](https://github.com/ossrs/srs/wiki/v4_EN_Home)
+            * [FAQ](https://github.com/ossrs/srs/issues/2716), [Features](https://github.com/ossrs/srs/blob/4.0release/trunk/doc/Features.md#features) or [ChangeLogs](https://github.com/ossrs/srs/blob/4.0release/trunk/doc/CHANGELOG.md#changelog)
+          draft: false
+          prerelease: false
+      # Upload release source files
+      - name: Upload Release Assets Source
+        id: upload-release-assets-source
+        uses: dwenegar/upload-release-assets@v1
+        env:
+          GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}
+        with:
+          release_id: ${{ steps.create_release.outputs.id }}
+          assets_path: ${{ env.SRS_SOURCE_TAR }}
+      # Upload release package files
+      - name: Upload Release Assets Package
+        id: upload-release-assets-package
+        uses: dwenegar/upload-release-assets@v1
+        env:
+          GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}
+        with:
+          release_id: ${{ steps.create_release.outputs.id }}
+          assets_path: ${{ env.SRS_PACKAGE_ZIP }}
+

.github/workflows/test.yml

@@ -0,0 +1,138 @@
+name: "Test"
+
+# @see https://docs.github.com/en/actions/reference/workflow-syntax-for-github-actions#onpushpull_requestbranchestags
+on: [push, pull_request]
+
+jobs:
+  build-centos:
+    name: actions-test-build-centos
+    runs-on: ubuntu-20.04
+
+    steps:
+      - name: Checkout repository
+        uses: actions/checkout@v2
+
+      # Build for CentOS7
+      - name: Build on CentOS7, baseline
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target centos7-baseline .
+      - name: Build on CentOS7, with SRT
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target centos7-srt .
+      - name: Build on CentOS7, without WebRTC
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target centos7-no-webrtc .
+      - name: Build on CentOS7, without ASM
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target centos7-no-asm .
+      - name: Build on CentOS7, C++98, no FFmpeg
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target centos7-ansi-no-ffmpeg .
+      # Build for CentOS8
+      - name: Build on CentOS8, baseline
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target centos8-baseline .
+      - name: Build on CentOS8, with SRT
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target centos8-srt .
+
+  build-ubuntu:
+    name: actions-test-build-ubuntu
+    runs-on: ubuntu-20.04
+
+    steps:
+      - name: Checkout repository
+        uses: actions/checkout@v2
+
+      # Build for Ubuntu16
+      - name: Build on Ubuntu16, baseline
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target ubuntu16-baseline .
+      - name: Build on Ubuntu16, with SRT
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target ubuntu16-srt .
+      # Build for Ubuntu18
+      - name: Build on Ubuntu18, baseline
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target ubuntu18-baseline .
+      - name: Build on Ubuntu18, with SRT
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target ubuntu18-srt .
+      # Build for Ubuntu20
+      - name: Build on Ubuntu20, baseline
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target ubuntu20-baseline .
+      - name: Build on Ubuntu20, with SRT
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target ubuntu20-srt .
+
+  build-cross:
+    name: actions-test-build-cross
+    runs-on: ubuntu-20.04
+
+    steps:
+      - name: Checkout repository
+        uses: actions/checkout@v2
+
+      # Cross Build for ARMv7
+      - name: Cross Build for ARMv7
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target ubuntu16-cross-armv7 .
+      # Cross Build for AARCH64
+      - name: Cross Build for AARCH64
+        run: DOCKER_BUILDKIT=1 docker build -f trunk/Dockerfile.builds --target ubuntu16-cross-aarch64 .
+
+  build:
+    name: actions-test-build
+    needs:
+      - build-centos
+      - build-ubuntu
+      - build-cross
+    runs-on: ubuntu-20.04
+
+    steps:
+      - run: echo 'Build done'
+
+  utest:
+    name: actions-test-utest
+    runs-on: ubuntu-20.04
+
+    steps:
+      - name: Checkout repository
+        uses: actions/checkout@v2
+
+      ################################################################
+      # Tests
+      - name: Build test image
+        run: docker build --tag srs:test -f trunk/Dockerfile.test .
+      # For utest
+      - name: Run SRS utest
+        run: docker run --rm srs:test bash -c 'make utest && ./objs/srs_utest'
+      # For regression-test
+      - name: Run SRS regression-test
+        run: docker run --rm srs:test bash -c 'make && ./objs/srs -c conf/regression-test.conf && cd 3rdparty/srs-bench && make && ./objs/srs_test -test.v'
+
+  coverage:
+    name: actions-test-coverage
+    runs-on: ubuntu-20.04
+
+    steps:
+      - name: Checkout repository
+        uses: actions/checkout@v2
+
+      ################################################################
+      # Tests
+      - name: Build coverage image
+        run: docker build --tag srs:cov -f trunk/Dockerfile.cov .
+      # For coverage
+      - name: Run SRS covergae
+        if: ${{ startsWith(github.ref, 'refs/heads/') || startsWith(github.ref, 'refs/pull/') }}
+        run: |
+          # The hash of commit.
+          SRS_SHA=${{ github.sha }}
+          # Note that the root of SRS, must contains .git, for report fixing.
+          SRS_PROJECT=/srs
+          # The github.ref is, for example, refs/heads/4.0release
+          SRS_BRANCH=$(echo ${{ github.ref }}| awk -F 'refs/heads/' '{print $2}'| awk -F '/' '{print $1}')
+          # The github.ref is, for example, refs/pull/2536/merge
+          SRS_PR=$(echo ${{ github.ref }}| awk -F 'refs/pull/' '{print $2}'| awk -F '/' '{print $1}')
+          echo "For ref=${{ github.ref }}, sha=${{ github.sha }}, SRS_BRANCH=$SRS_BRANCH, SRS_PR=$SRS_PR, SRS_SHA=$SRS_SHA, SRS_PROJECT=$SRS_PROJECT"
+          docker run --rm --env SRS_BRANCH=$SRS_BRANCH --env SRS_PR=$SRS_PR --env SRS_SHA=$SRS_SHA --env SRS_PROJECT=$SRS_PROJECT \
+            srs:cov bash -c 'make utest && ./objs/srs_utest && bash auto/codecov.sh'
+
+  done:
+    name: actions-test-done
+    needs:
+      - build
+      - utest
+      - coverage
+    runs-on: ubuntu-20.04
+
+    steps:
+      - run: echo 'All done'

.gitignore

@@ -16,13 +16,14 @@
 *.pyc
 *.swp
 .DS_Store
+.vscode
+.vscode/*
 /trunk/Makefile
 /trunk/objs
 /trunk/src/build-qt-Desktop-Debug
 /trunk/research/librtmp/objs
 /trunk/3rdparty/ccache/ccache-3.1.9
 /trunk/3rdparty/gprof/graphviz-2.36.0
-/trunk/research/api-server/static-dir/crossdomain.xml
 /trunk/research/api-server/static-dir/forward
 /trunk/research/api-server/static-dir/live
 /trunk/research/api-server/static-dir/players
@@ -33,3 +34,10 @@
 .idea
 .DS_Store
 
+cmake-build-debug
+/trunk/ide/srs_clion/CMakeCache.txt
+/trunk/ide/srs_clion/CMakeFiles
+/trunk/ide/srs_clion/Makefile
+/trunk/ide/srs_clion/cmake_install.cmake
+/trunk/ide/srs_clion/srs
+/trunk/ide/srs_clion/Testing/

.run/srs-ingest.run.xml

@@ -0,0 +1,7 @@
+<component name="ProjectRunConfigurationManager">
+  <configuration default="false" name="srs-ingest" type="CMakeRunConfiguration" factoryName="Application" PROGRAM_PARAMS="-c conf/clion-ingest.conf" REDIRECT_INPUT="false" ELEVATE="false" USE_EXTERNAL_CONSOLE="false" WORKING_DIR="file://$CMakeCurrentBuildDir$/../../../" PASS_PARENT_ENVS_2="true" PROJECT_NAME="srs" TARGET_NAME="srs" CONFIG_NAME="Debug" RUN_TARGET_PROJECT_NAME="srs" RUN_TARGET_NAME="srs">
+    <method v="2">
+      <option name="com.jetbrains.cidr.execution.CidrBuildBeforeRunTaskProvider$BuildBeforeRunTask" enabled="true" />
+    </method>
+  </configuration>
+</component>

.run/srs.run.xml

@@ -0,0 +1,7 @@
+<component name="ProjectRunConfigurationManager">
+  <configuration default="false" name="srs" type="CMakeRunConfiguration" factoryName="Application" PROGRAM_PARAMS="-c conf/clion.conf" REDIRECT_INPUT="false" ELEVATE="false" USE_EXTERNAL_CONSOLE="false" WORKING_DIR="file://$CMakeCurrentBuildDir$/../../../" PASS_PARENT_ENVS_2="true" PROJECT_NAME="srs" TARGET_NAME="srs" CONFIG_NAME="Debug" RUN_TARGET_PROJECT_NAME="srs" RUN_TARGET_NAME="srs">
+    <method v="2">
+      <option name="com.jetbrains.cidr.execution.CidrBuildBeforeRunTaskProvider$BuildBeforeRunTask" enabled="true" />
+    </method>
+  </configuration>
+</component>

LICENSE

@@ -1,6 +1,6 @@
 The MIT License (MIT)
 
-Copyright (c) 2013-2020 Winlin
+Copyright (c) 2013-2021 Winlin
 
 Permission is hereby granted, free of charge, to any person obtaining a copy of
 this software and associated documentation files (the "Software"), to deal in
@@ -19,3 +19,11 @@ COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
 IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
 CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 
+---------------------------------------------------------------------------
+Note:
+Individual files contain the following tag instead of the full license text.
+
+	SPDX-License-Identifier: MIT
+
+This enables machine processing of license information based on the SPDX
+License Identifiers that are here available: http://spdx.org/licenses/

trunk/.gitignore

@@ -13,6 +13,7 @@
 /ide/srs_xcode/srs_xcode.xcodeproj/xcuserdata/
 /research/aac/
 /research/api-server/static-dir/mse
+/research/api-server/static-dir/crossdomain.xml
 /research/bat/
 /research/big/
 /research/bitch/
@@ -34,11 +35,18 @@
 /research/speex/
 /test/
 .DS_Store
-srs
+./srs
 *.dSYM/
 *.gcov
+*.gcno
 *.ts
 *.h264
 *.264
-3rdparty/ffmpeg-4.2-fit
+/webrtc
+/configure.sh
+/janus
+*.dump
 bug
+/research/thread-model/thread-local
+*.gcp
+*.svg

trunk/3rdparty/README.md

@@ -0,0 +1,73 @@
+http-parser-2.1.zip
+* for srs to support http callback.
+* https://github.com/joyent/http-parser
+* https://github.com/ossrs/srs/wiki/LicenseMixing#http-parser
+
+nginx-1.5.7.zip
+* http://nginx.org/
+* for srs to support hls streaming.
+
+srt-1-fit
+srt-1.4.1.tar.gz
+* https://github.com/Haivision/srt/releases/tag/v1.4.1
+* https://github.com/ossrs/srs/wiki/LicenseMixing#srt
+
+openssl-1.1-fit
+openssl-1.1.1b.tar.gz
+* http://www.openssl.org/source/openssl-1.1.1b.tar.gz
+
+openssl-1.1.0e.zip
+openssl-OpenSSL_1_0_2u.tar.gz
+* http://www.openssl.org/source/openssl-1.1.0e.tar.gz
+* openssl for SRS(with-ssl) RTMP complex handshake to delivery h264+aac stream.
+* SRTP depends on openssl 1.0.*, so we use both ssl versions.
+* https://github.com/ossrs/srs/wiki/LicenseMixing#openssl
+
+CherryPy-3.2.4.zip
+* sample api server for srs.
+* https://pypi.python.org/pypi/CherryPy/3.2.4
+
+libsrtp-2.3.0.tar.gz
+* For WebRTC, SRTP to encrypt and decrypt RTP.
+* https://github.com/cisco/libsrtp/releases/tag/v2.3.0
+
+ffmpeg-4.2.tar.gz
+opus-1.3.1.tar.gz
+* http://ffmpeg.org/releases/ffmpeg-4.2.tar.gz
+* https://github.com/xiph/opus/releases/tag/v1.3.1
+* To support RTMP/WebRTC transcoding.
+* https://github.com/ossrs/srs/wiki/LicenseMixing#ffmpeg
+    
+gtest-1.6.0.zip
+* google test framework.
+* https://code.google.com/p/googletest/downloads/list
+    
+gperftools-2.1.zip
+* gperf tools for performance benchmark.
+* https://code.google.com/p/gperftools/downloads/list
+
+st-srs
+st-1.9.zip
+state-threads
+state-threads-1.9.1.tar.gz
+* Patched ST from https://github.com/ossrs/state-threads
+* https://github.com/ossrs/srs/wiki/LicenseMixing#state-threads
+
+JSON
+* https://github.com/udp/json-parser
+* https://github.com/ossrs/srs/wiki/LicenseMixing#json
+
+USRSCTP
+* https://github.com/ossrs/srs/wiki/LicenseMixing#usrsctp
+
+links:
+* state-threads:
+        https://github.com/ossrs/state-threads
+* x264: 
+        ftp://ftp.videolan.org/pub/videolan/x264/snapshots/x264-snapshot-20131129-2245-stable.tar.bz2
+* lame: 
+        http://nchc.dl.sourceforge.net/project/lame/lame/3.99/lame-3.99.5.tar.gz
+* yasm:
+        http://www.tortall.net/projects/yasm/releases/yasm-1.2.0.tar.gz
+* speex:
+        http://downloads.xiph.org/releases/speex/speex-1.2rc1.tar.gz

trunk/3rdparty/ccache/build_ccache.sh

@@ -1,17 +0,0 @@
-#!/bin/bash
-
-# check exists.
-if [[ -f /usr/local/bin/ccache ]]; then 
-    echo "ccache is ok";
-    exit 0;
-fi
-
-# check sudoer.
-sudo echo "ok" > /dev/null 2>&1; 
-ret=$?; if [[ 0 -ne ${ret} ]]; then echo "you must be sudoer"; exit 1; fi
-
-unzip ccache-3.1.9.zip && cd ccache-3.1.9 && ./configure && make
-ret=$?; if [[ $ret -ne 0 ]]; then echo "build ccache failed."; exit $ret; fi
-
-sudo cp ccache /usr/local/bin && sudo ln -s ccache /usr/local/bin/gcc && sudo ln -s ccache /usr/local/bin/g++ && sudo ln -s ccache /usr/local/bin/cc && sudo ln -s ccache /usr/local/bin/c++
-ret=$?; if [[ $ret -ne 0 ]]; then echo "install ccache failed."; exit $ret; fi

trunk/3rdparty/ccache/readme.txt

@@ -1,11 +0,0 @@
-ccache是samba组织提供的加速编译过程的工具，
-使用虚拟机编译可以考虑用这个工具，让编译过程飞快。
-
-链接：
-    http://ccache.samba.org/
-    http://samba.org/ftp/ccache/ccache-3.1.9.tar.xz
-    http://ccache.samba.org/manual.html
-
-安装方法：
-    bash build_ccache.sh
-注意：要求以sudoer执行，要修改文件。

trunk/3rdparty/ffmpeg-4-fit/.gitignore

@@ -0,0 +1,21 @@
+# Ignore files.
+*.d
+*.pc
+*.o
+_release
+
+# We should remove bellow files in depends.sh
+doc
+config.asm
+config.h
+ffbuild/config.fate
+ffbuild/config.log
+ffbuild/config.mak
+ffbuild/config.sh
+ffbuild/.config
+libavutil/lib.version
+libavutil/libavutil.version
+libavutil/ffversion.h
+libavutil/avconfig.h
+libavcodec/libavcodec.version
+libswresample/libswresample.version

trunk/3rdparty/ffmpeg-4-fit/LICENSE.md

@@ -0,0 +1,120 @@
+# License
+
+Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
+or later (LGPL v2.1+). Read the file `COPYING.LGPLv2.1` for details. Some other
+files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
+FFmpeg.
+
+Some optional parts of FFmpeg are licensed under the GNU General Public License
+version 2 or later (GPL v2+). See the file `COPYING.GPLv2` for details. None of
+these parts are used by default, you have to explicitly pass `--enable-gpl` to
+configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
+
+Specifically, the GPL parts of FFmpeg are:
+
+- libpostproc
+- optional x86 optimization in the files
+    - `libavcodec/x86/flac_dsp_gpl.asm`
+    - `libavcodec/x86/idct_mmx.c`
+    - `libavfilter/x86/vf_removegrain.asm`
+- the following building and testing tools
+    - `compat/solaris/make_sunver.pl`
+    - `doc/t2h.pm`
+    - `doc/texi2pod.pl`
+    - `libswresample/swresample-test.c`
+    - `tests/checkasm/*`
+    - `tests/tiny_ssim.c`
+- the following filters in libavfilter:
+    - `vf_blackframe.c`
+    - `vf_boxblur.c`
+    - `vf_colormatrix.c`
+    - `vf_cover_rect.c`
+    - `vf_cropdetect.c`
+    - `vf_delogo.c`
+    - `vf_eq.c`
+    - `vf_find_rect.c`
+    - `vf_fspp.c`
+    - `vf_geq.c`
+    - `vf_histeq.c`
+    - `vf_hqdn3d.c`
+    - `vf_interlace.c`
+    - `vf_kerndeint.c`
+    - `vf_mcdeint.c`
+    - `vf_mpdecimate.c`
+    - `vf_owdenoise.c`
+    - `vf_perspective.c`
+    - `vf_phase.c`
+    - `vf_pp.c`
+    - `vf_pp7.c`
+    - `vf_pullup.c`
+    - `vf_repeatfields.c`
+    - `vf_sab.c`
+    - `vf_smartblur.c`
+    - `vf_spp.c`
+    - `vf_stereo3d.c`
+    - `vf_super2xsai.c`
+    - `vf_tinterlace.c`
+    - `vf_uspp.c`
+    - `vsrc_mptestsrc.c`
+
+Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
+the configure parameter `--enable-version3` will activate this licensing option
+for you. Read the file `COPYING.LGPLv3` or, if you have enabled GPL parts,
+`COPYING.GPLv3` to learn the exact legal terms that apply in this case.
+
+There are a handful of files under other licensing terms, namely:
+
+* The files `libavcodec/jfdctfst.c`, `libavcodec/jfdctint_template.c` and
+  `libavcodec/jrevdct.c` are taken from libjpeg, see the top of the files for
+  licensing details. Specifically note that you must credit the IJG in the
+  documentation accompanying your program if you only distribute executables.
+  You must also indicate any changes including additions and deletions to
+  those three files in the documentation.
+* `tests/reference.pnm` is under the expat license.
+
+
+## External libraries
+
+FFmpeg can be combined with a number of external libraries, which sometimes
+affect the licensing of binaries resulting from the combination.
+
+### Compatible libraries
+
+The following libraries are under GPL:
+- frei0r
+- libcdio
+- librubberband
+- libvidstab
+- libx264
+- libx265
+- libxavs
+- libxvid
+
+When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
+passing `--enable-gpl` to configure.
+
+The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
+license is incompatible with the LGPL v2.1 and the GPL v2, but not with
+version 3 of those licenses. So to combine these libraries with FFmpeg, the
+license version needs to be upgraded by passing `--enable-version3` to configure.
+
+### Incompatible libraries
+
+There are certain libraries you can combine with FFmpeg whose licenses are not
+compatible with the GPL and/or the LGPL. If you wish to enable these
+libraries, even in circumstances that their license may be incompatible, pass
+`--enable-nonfree` to configure. But note that if you enable any of these
+libraries the resulting binary will be under a complex license mix that is
+more restrictive than the LGPL and that may result in additional obligations.
+It is possible that these restrictions cause the resulting binary to be
+unredistributable.
+
+The Fraunhofer FDK AAC and OpenSSL libraries are under licenses which are
+incompatible with the GPLv2 and v3. To the best of our knowledge, they are
+compatible with the LGPL.
+
+The NVENC library, while its header file is licensed under the compatible MIT
+license, requires a proprietary binary blob at run time, and is deemed to be
+incompatible with the GPL. We are not certain if it is compatible with the
+LGPL, but we require `--enable-nonfree` even with LGPL configurations in case
+it is not.

trunk/3rdparty/ffmpeg-4-fit/Makefile

@@ -0,0 +1,167 @@
+MAIN_MAKEFILE=1
+include ffbuild/config.mak
+
+vpath %.c    $(SRC_PATH)
+vpath %.cpp  $(SRC_PATH)
+vpath %.h    $(SRC_PATH)
+vpath %.inc  $(SRC_PATH)
+vpath %.m    $(SRC_PATH)
+vpath %.S    $(SRC_PATH)
+vpath %.asm  $(SRC_PATH)
+vpath %.rc   $(SRC_PATH)
+vpath %.v    $(SRC_PATH)
+vpath %.texi $(SRC_PATH)
+vpath %.cu   $(SRC_PATH)
+vpath %.ptx  $(SRC_PATH)
+vpath %/fate_config.sh.template $(SRC_PATH)
+
+TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64 audiomatch
+HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
+
+# $(FFLIBS-yes) needs to be in linking order
+FFLIBS-$(CONFIG_AVDEVICE)   += avdevice
+FFLIBS-$(CONFIG_AVFILTER)   += avfilter
+FFLIBS-$(CONFIG_AVFORMAT)   += avformat
+FFLIBS-$(CONFIG_AVCODEC)    += avcodec
+FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
+FFLIBS-$(CONFIG_POSTPROC)   += postproc
+FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
+FFLIBS-$(CONFIG_SWSCALE)    += swscale
+
+FFLIBS := avutil
+
+DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
+
+SKIPHEADERS = compat/w32pthreads.h
+
+# first so "all" becomes default target
+all: all-yes
+
+include $(SRC_PATH)/ffbuild/common.mak
+
+FF_EXTRALIBS := $(FFEXTRALIBS)
+FF_DEP_LIBS  := $(DEP_LIBS)
+FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
+
+$(TOOLS): %$(EXESUF): %.o
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS)
+
+target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
+
+tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
+tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
+tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
+tools/target_dec_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
+
+CONFIGURABLE_COMPONENTS =                                           \
+    $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))                 \
+    $(SRC_PATH)/libavcodec/bitstream_filters.c                      \
+    $(SRC_PATH)/libavcodec/parsers.c                                \
+    $(SRC_PATH)/libavformat/protocols.c                             \
+
+config.h: ffbuild/.config
+ffbuild/.config: $(CONFIGURABLE_COMPONENTS)
+	@-tput bold 2>/dev/null
+	@-printf '\nWARNING: $(?) newer than config.h, rerun configure\n\n'
+	@-tput sgr0 2>/dev/null
+
+SUBDIR_VARS := CLEANFILES FFLIBS HOSTPROGS TESTPROGS TOOLS               \
+               HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
+               ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
+               ALTIVEC-OBJS VSX-OBJS MMX-OBJS X86ASM-OBJS                \
+               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSP-OBJS MSA-OBJS         \
+               MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
+
+define RESET
+$(1) :=
+$(1)-yes :=
+endef
+
+define DOSUBDIR
+$(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
+SUBDIR := $(1)/
+include $(SRC_PATH)/$(1)/Makefile
+-include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
+-include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
+include $(SRC_PATH)/ffbuild/library.mak
+endef
+
+$(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
+
+libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
+
+$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
+ifeq ($(STRIPTYPE),direct)
+	$(STRIP) -o $@ $<
+else
+	$(CP) $< $@
+	$(STRIP) $@
+endif
+
+%$(PROGSSUF)_g$(EXESUF): $(FF_DEP_LIBS)
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
+
+VERSION_SH  = $(SRC_PATH)/ffbuild/version.sh
+GIT_LOG     = $(SRC_PATH)/.git/logs/HEAD
+
+.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) ffbuild/config.mak
+.version: M=@
+
+libavutil/ffversion.h .version:
+	$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
+	$(Q)touch .version
+
+# force version.sh to run whenever version might have changed
+-include .version
+
+install: install-libs install-headers
+
+install-libs: install-libs-yes
+
+install-data: $(DATA_FILES)
+	$(Q)mkdir -p "$(DATADIR)"
+	$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
+
+uninstall: uninstall-data uninstall-headers uninstall-libs uninstall-pkgconfig
+
+uninstall-data:
+	$(RM) -r "$(DATADIR)"
+
+clean::
+	$(RM) $(CLEANSUFFIXES)
+	$(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES))
+	$(RM) -r coverage-html
+	$(RM) -rf coverage.info coverage.info.in lcov
+
+distclean:: clean
+	$(RM) .version avversion.h config.asm config.h mapfile  \
+		ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
+		version.h libavutil/ffversion.h libavcodec/codec_names.h \
+		libavcodec/bsf_list.c libavformat/protocol_list.c \
+		libavcodec/codec_list.c libavcodec/parser_list.c \
+		libavformat/muxer_list.c libavformat/demuxer_list.c
+ifeq ($(SRC_LINK),src)
+	$(RM) src
+endif
+	$(RM) -rf doc/examples/pc-uninstalled
+
+config:
+	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
+
+build: all alltools examples testprogs
+check: all alltools examples testprogs fate
+
+$(sort $(OBJDIRS)):
+	$(Q)mkdir -p $@
+
+# Dummy rule to stop make trying to rebuild removed or renamed headers
+%.h:
+	@:
+
+# Disable suffix rules.  Most of the builtin rules are suffix rules,
+# so this saves some time on slow systems.
+.SUFFIXES:
+
+.PHONY: all all-yes alltools build check config testprogs
+.PHONY: *clean install* uninstall*

trunk/3rdparty/ffmpeg-4-fit/compat/atomics/gcc/stdatomic.h

@@ -0,0 +1,173 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/*
+ * based on vlc_atomic.h from VLC
+ * Copyright (C) 2010 Rémi Denis-Courmont
+ */
+
+#ifndef COMPAT_ATOMICS_GCC_STDATOMIC_H
+#define COMPAT_ATOMICS_GCC_STDATOMIC_H
+
+#include <stddef.h>
+#include <stdint.h>
+
+#define ATOMIC_FLAG_INIT 0
+
+#define ATOMIC_VAR_INIT(value) (value)
+
+#define atomic_init(obj, value) \
+do {                            \
+    *(obj) = (value);           \
+} while(0)
+
+#define kill_dependency(y) ((void)0)
+
+#define atomic_thread_fence(order) \
+    __sync_synchronize()
+
+#define atomic_signal_fence(order) \
+    ((void)0)
+
+#define atomic_is_lock_free(obj) 0
+
+typedef         _Bool      atomic_flag;
+typedef         _Bool      atomic_bool;
+typedef          char      atomic_char;
+typedef   signed char      atomic_schar;
+typedef unsigned char      atomic_uchar;
+typedef          short     atomic_short;
+typedef unsigned short     atomic_ushort;
+typedef          int       atomic_int;
+typedef unsigned int       atomic_uint;
+typedef          long      atomic_long;
+typedef unsigned long      atomic_ulong;
+typedef          long long atomic_llong;
+typedef unsigned long long atomic_ullong;
+typedef          wchar_t   atomic_wchar_t;
+typedef       int_least8_t atomic_int_least8_t;
+typedef      uint_least8_t atomic_uint_least8_t;
+typedef      int_least16_t atomic_int_least16_t;
+typedef     uint_least16_t atomic_uint_least16_t;
+typedef      int_least32_t atomic_int_least32_t;
+typedef     uint_least32_t atomic_uint_least32_t;
+typedef      int_least64_t atomic_int_least64_t;
+typedef     uint_least64_t atomic_uint_least64_t;
+typedef       int_fast8_t atomic_int_fast8_t;
+typedef      uint_fast8_t atomic_uint_fast8_t;
+typedef      int_fast16_t atomic_int_fast16_t;
+typedef     uint_fast16_t atomic_uint_fast16_t;
+typedef      int_fast32_t atomic_int_fast32_t;
+typedef     uint_fast32_t atomic_uint_fast32_t;
+typedef      int_fast64_t atomic_int_fast64_t;
+typedef     uint_fast64_t atomic_uint_fast64_t;
+typedef          intptr_t atomic_intptr_t;
+typedef         uintptr_t atomic_uintptr_t;
+typedef            size_t atomic_size_t;
+typedef         ptrdiff_t atomic_ptrdiff_t;
+typedef          intmax_t atomic_intmax_t;
+typedef         uintmax_t atomic_uintmax_t;
+
+#define atomic_store(object, desired)   \
+do {                                    \
+    *(object) = (desired);              \
+    __sync_synchronize();               \
+} while (0)
+
+#define atomic_store_explicit(object, desired, order) \
+    atomic_store(object, desired)
+
+#define atomic_load(object) \
+    (__sync_synchronize(), *(object))
+
+#define atomic_load_explicit(object, order) \
+    atomic_load(object)
+
+#define atomic_exchange(object, desired)                            \
+({                                                                  \
+    __typeof__(object) _obj = (object);                             \
+    __typeof__(*object) _old;                                       \
+    do                                                              \
+        _old = atomic_load(_obj);                                   \
+    while (!__sync_bool_compare_and_swap(_obj, _old, (desired)));   \
+    _old;                                                           \
+})
+
+#define atomic_exchange_explicit(object, desired, order) \
+    atomic_exchange(object, desired)
+
+#define atomic_compare_exchange_strong(object, expected, desired)   \
+({                                                                  \
+    __typeof__(object) _exp = (expected);                           \
+    __typeof__(*object) _old = *_exp;                               \
+    *_exp = __sync_val_compare_and_swap((object), _old, (desired)); \
+    *_exp == _old;                                                  \
+})
+
+#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
+    atomic_compare_exchange_strong(object, expected, desired)
+
+#define atomic_compare_exchange_weak(object, expected, desired) \
+    atomic_compare_exchange_strong(object, expected, desired)
+
+#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
+    atomic_compare_exchange_weak(object, expected, desired)
+
+#define atomic_fetch_add(object, operand) \
+    __sync_fetch_and_add(object, operand)
+
+#define atomic_fetch_add_explicit(object, operand, order) \
+    atomic_fetch_add(object, operand)
+
+#define atomic_fetch_sub(object, operand) \
+    __sync_fetch_and_sub(object, operand)
+
+#define atomic_fetch_sub_explicit(object, operand, order) \
+    atomic_fetch_sub(object, operand)
+
+#define atomic_fetch_or(object, operand) \
+    __sync_fetch_and_or(object, operand)
+
+#define atomic_fetch_or_explicit(object, operand, order) \
+    atomic_fetch_or(object, operand)
+
+#define atomic_fetch_xor(object, operand) \
+    __sync_fetch_and_xor(object, operand)
+
+#define atomic_fetch_xor_explicit(object, operand, order) \
+    atomic_fetch_xor(object, operand)
+
+#define atomic_fetch_and(object, operand) \
+    __sync_fetch_and_and(object, operand)
+
+#define atomic_fetch_and_explicit(object, operand, order) \
+    atomic_fetch_and(object, operand)
+
+#define atomic_flag_test_and_set(object) \
+    atomic_exchange(object, 1)
+
+#define atomic_flag_test_and_set_explicit(object, order) \
+    atomic_flag_test_and_set(object)
+
+#define atomic_flag_clear(object) \
+    atomic_store(object, 0)
+
+#define atomic_flag_clear_explicit(object, order) \
+    atomic_flag_clear(object)
+
+#endif /* COMPAT_ATOMICS_GCC_STDATOMIC_H */

trunk/3rdparty/ffmpeg-4-fit/compat/va_copy.h

@@ -0,0 +1,34 @@
+/*
+ * MSVC Compatible va_copy macro
+ * Copyright (c) 2012 Derek Buitenhuis
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef COMPAT_VA_COPY_H
+#define COMPAT_VA_COPY_H
+
+#include <stdarg.h>
+
+#if !defined(va_copy) && defined(_MSC_VER)
+#define va_copy(dst, src) ((dst) = (src))
+#endif
+#if !defined(va_copy) && defined(__GNUC__) && __GNUC__ < 3
+#define va_copy(dst, src) __va_copy(dst, src)
+#endif
+
+#endif /* COMPAT_VA_COPY_H */

trunk/3rdparty/ffmpeg-4-fit/ffbuild/arch.mak

@@ -0,0 +1,17 @@
+OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
+OBJS-$(HAVE_ARMV6)   += $(ARMV6-OBJS)   $(ARMV6-OBJS-yes)
+OBJS-$(HAVE_ARMV8)   += $(ARMV8-OBJS)   $(ARMV8-OBJS-yes)
+OBJS-$(HAVE_VFP)     += $(VFP-OBJS)     $(VFP-OBJS-yes)
+OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
+
+OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
+OBJS-$(HAVE_MIPSDSP)   += $(MIPSDSP-OBJS)    $(MIPSDSP-OBJS-yes)
+OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
+OBJS-$(HAVE_MSA)       += $(MSA-OBJS)        $(MSA-OBJS-yes)
+OBJS-$(HAVE_MMI)   += $(MMI-OBJS)   $(MMI-OBJS-yes)
+
+OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
+OBJS-$(HAVE_VSX)     += $(VSX-OBJS) $(VSX-OBJS-yes)
+
+OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
+OBJS-$(HAVE_X86ASM)  += $(X86ASM-OBJS)  $(X86ASM-OBJS-yes)

trunk/3rdparty/ffmpeg-4-fit/ffbuild/common.mak

@@ -0,0 +1,176 @@
+#
+# common bits used by all libraries
+#
+
+DEFAULT_X86ASMD=.dbg
+
+ifeq ($(DBG),1)
+X86ASMD=$(DEFAULT_X86ASMD)
+else
+X86ASMD=
+endif
+
+ifndef SUBDIR
+
+ifndef V
+Q      = @
+ECHO   = printf "$(1)\t%s\n" $(2)
+BRIEF  = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC
+SILENT = DEPCC DEPHOSTCC DEPAS DEPX86ASM RANLIB RM
+
+MSG    = $@
+M      = @$(call ECHO,$(TAG),$@);
+$(foreach VAR,$(BRIEF), \
+    $(eval override $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
+$(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
+$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
+endif
+
+ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
+
+# NASM requires -I path terminated with /
+IFLAGS     := -I. -I$(SRC_LINK)/
+CPPFLAGS   := $(IFLAGS) $(CPPFLAGS)
+CFLAGS     += $(ECFLAGS)
+CCFLAGS     = $(CPPFLAGS) $(CFLAGS)
+OBJCFLAGS  += $(EOBJCFLAGS)
+OBJCCFLAGS  = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
+ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
+CXXFLAGS   := $(CPPFLAGS) $(CFLAGS) $(CXXFLAGS)
+X86ASMFLAGS += $(IFLAGS:%=%/) -I$(<D)/ -Pconfig.asm
+NVCCFLAGS  += -ptx
+
+HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
+LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
+
+define COMPILE
+       $(call $(1)DEP,$(1))
+       $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
+endef
+
+COMPILE_C = $(call COMPILE,CC)
+COMPILE_CXX = $(call COMPILE,CXX)
+COMPILE_S = $(call COMPILE,AS)
+COMPILE_M = $(call COMPILE,OBJCC)
+COMPILE_X86ASM = $(call COMPILE,X86ASM)
+COMPILE_HOSTC = $(call COMPILE,HOSTCC)
+COMPILE_NVCC = $(call COMPILE,NVCC)
+
+%.o: %.c
+	$(COMPILE_C)
+
+%.o: %.cpp
+	$(COMPILE_CXX)
+
+%.o: %.m
+	$(COMPILE_M)
+
+%.s: %.c
+	$(CC) $(CCFLAGS) -S -o $@ $<
+
+%.o: %.S
+	$(COMPILE_S)
+
+%_host.o: %.c
+	$(COMPILE_HOSTC)
+
+%$(DEFAULT_X86ASMD).asm: %.asm
+	$(DEPX86ASM) $(X86ASMFLAGS) -M -o $@ $< > $(@:.asm=.d)
+	$(X86ASM) $(X86ASMFLAGS) -e $< | sed '/^%/d;/^$$/d;' > $@
+
+%.o: %.asm
+	$(COMPILE_X86ASM)
+	-$(if $(ASMSTRIPFLAGS), $(STRIP) $(ASMSTRIPFLAGS) $@)
+
+%.o: %.rc
+	$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
+
+%.i: %.c
+	$(CC) $(CCFLAGS) $(CC_E) $<
+
+%.h.c:
+	$(Q)echo '#include "$*.h"' >$@
+
+%.ptx: %.cu
+	$(COMPILE_NVCC)
+
+%.ptx.c: %.ptx
+	$(Q)sh $(SRC_PATH)/compat/cuda/ptx2c.sh $@ $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
+
+%.c %.h %.pc %.ver %.version: TAG = GEN
+
+# Dummy rule to stop make trying to rebuild removed or renamed headers
+%.h:
+	@:
+
+# Disable suffix rules.  Most of the builtin rules are suffix rules,
+# so this saves some time on slow systems.
+.SUFFIXES:
+
+# Do not delete intermediate files from chains of implicit rules
+$(OBJS):
+endif
+
+include $(SRC_PATH)/ffbuild/arch.mak
+
+OBJS      += $(OBJS-yes)
+SLIBOBJS  += $(SLIBOBJS-yes)
+FFLIBS    := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
+TESTPROGS += $(TESTPROGS-yes)
+
+LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
+FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(foreach lib,EXTRALIBS-$(NAME) $(FFLIBS:%=EXTRALIBS-%),$($(lib))) $(EXTRALIBS)
+
+OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
+SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
+TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)tests/%) $(TESTPROGS:%=$(SUBDIR)tests/%.o)
+TESTPROGS := $(TESTPROGS:%=$(SUBDIR)tests/%$(EXESUF))
+HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
+HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
+TOOLS     += $(TOOLS-yes)
+TOOLOBJS  := $(TOOLS:%=tools/%.o)
+TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
+HEADERS   += $(HEADERS-yes)
+
+PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(2)LIBNAME))
+DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib),$(CONFIG_SHARED:yes=S)))
+STATIC_DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
+
+SRC_DIR    := $(SRC_PATH)/lib$(NAME)
+ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
+SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
+SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
+HOBJS        = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
+PTXOBJS      = $(filter %.ptx.o,$(OBJS))
+$(HOBJS):     CCFLAGS += $(CFLAGS_HEADERS)
+checkheaders: $(HOBJS)
+.SECONDARY:   $(HOBJS:.o=.c) $(PTXOBJS:.o=.c) $(PTXOBJS:.o=)
+
+alltools: $(TOOLS)
+
+$(HOSTOBJS): %.o: %.c
+	$(COMPILE_HOSTC)
+
+$(HOSTPROGS): %$(HOSTEXESUF): %.o
+	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTEXTRALIBS)
+
+$(OBJS):     | $(sort $(dir $(OBJS)))
+$(HOBJS):    | $(sort $(dir $(HOBJS)))
+$(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
+$(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
+$(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
+$(TOOLOBJS): | tools
+
+OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
+
+CLEANSUFFIXES     = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~
+LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
+
+define RULES
+clean::
+	$(RM) $(HOSTPROGS) $(TESTPROGS) $(TOOLS)
+endef
+
+$(eval $(RULES))
+
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_X86ASMD).d)

trunk/3rdparty/ffmpeg-4-fit/ffbuild/library.mak

@@ -0,0 +1,107 @@
+include $(SRC_PATH)/ffbuild/common.mak
+
+ifeq (,$(filter %clean config,$(MAKECMDGOALS)))
+-include $(SUBDIR)lib$(NAME).version
+endif
+
+LIBVERSION := $(lib$(NAME)_VERSION)
+LIBMAJOR   := $(lib$(NAME)_VERSION_MAJOR)
+LIBMINOR   := $(lib$(NAME)_VERSION_MINOR)
+INCINSTDIR := $(INCDIR)/lib$(NAME)
+
+INSTHEADERS := $(INSTHEADERS) $(HEADERS:%=$(SUBDIR)%)
+
+all-$(CONFIG_STATIC): $(SUBDIR)$(LIBNAME)  $(SUBDIR)lib$(FULLNAME).pc
+all-$(CONFIG_SHARED): $(SUBDIR)$(SLIBNAME) $(SUBDIR)lib$(FULLNAME).pc
+
+LIBOBJS := $(OBJS) $(SUBDIR)%.h.o $(TESTOBJS)
+$(LIBOBJS) $(LIBOBJS:.o=.s) $(LIBOBJS:.o=.i):   CPPFLAGS += -DHAVE_AV_CONFIG_H
+
+$(SUBDIR)$(LIBNAME): $(OBJS)
+	$(RM) $@
+	$(AR) $(ARFLAGS) $(AR_O) $^
+	$(RANLIB) $@
+
+install-headers: install-lib$(NAME)-headers install-lib$(NAME)-pkgconfig
+
+install-libs-$(CONFIG_STATIC): install-lib$(NAME)-static
+install-libs-$(CONFIG_SHARED): install-lib$(NAME)-shared
+
+define RULES
+$(TOOLS):     THISLIB = $(FULLNAME:%=$(LD_LIB))
+$(TESTPROGS): THISLIB = $(SUBDIR)$(LIBNAME)
+
+$(LIBOBJS): CPPFLAGS += -DBUILDING_$(NAME)
+
+$(TESTPROGS) $(TOOLS): %$(EXESUF): %.o
+	$$(LD) $(LDFLAGS) $(LDEXEFLAGS) $$(LD_O) $$(filter %.o,$$^) $$(THISLIB) $(FFEXTRALIBS) $$(EXTRALIBS-$$(*F)) $$(ELIBS)
+
+$(SUBDIR)lib$(NAME).version: $(SUBDIR)version.h | $(SUBDIR)
+	$$(M) $$(SRC_PATH)/ffbuild/libversion.sh $(NAME) $$< > $$@
+
+$(SUBDIR)lib$(FULLNAME).pc: $(SUBDIR)version.h ffbuild/config.sh | $(SUBDIR)
+	$$(M) $$(SRC_PATH)/ffbuild/pkgconfig_generate.sh $(NAME) "$(DESC)"
+
+$(SUBDIR)lib$(NAME).ver: $(SUBDIR)lib$(NAME).v $(OBJS)
+	$$(M)sed 's/MAJOR/$(lib$(NAME)_VERSION_MAJOR)/' $$< | $(VERSION_SCRIPT_POSTPROCESS_CMD) > $$@
+
+$(SUBDIR)$(SLIBNAME): $(SUBDIR)$(SLIBNAME_WITH_MAJOR)
+	$(Q)cd ./$(SUBDIR) && $(LN_S) $(SLIBNAME_WITH_MAJOR) $(SLIBNAME)
+
+$(SUBDIR)$(SLIBNAME_WITH_MAJOR): $(OBJS) $(SLIBOBJS) $(SUBDIR)lib$(NAME).ver
+	$(SLIB_CREATE_DEF_CMD)
+	$$(LD) $(SHFLAGS) $(LDFLAGS) $(LDSOFLAGS) $$(LD_O) $$(filter %.o,$$^) $(FFEXTRALIBS)
+	$(SLIB_EXTRA_CMD)
+
+ifdef SUBDIR
+$(SUBDIR)$(SLIBNAME_WITH_MAJOR): $(DEP_LIBS)
+endif
+
+clean::
+	$(RM) $(addprefix $(SUBDIR),$(CLEANFILES) $(CLEANSUFFIXES) $(LIBSUFFIXES)) \
+	    $(CLEANSUFFIXES:%=$(SUBDIR)$(ARCH)/%) $(CLEANSUFFIXES:%=$(SUBDIR)tests/%)
+
+install-lib$(NAME)-shared: $(SUBDIR)$(SLIBNAME)
+	$(Q)mkdir -p "$(SHLIBDIR)"
+	$$(INSTALL) -m 755 $$< "$(SHLIBDIR)/$(SLIB_INSTALL_NAME)"
+	$$(STRIP) "$(SHLIBDIR)/$(SLIB_INSTALL_NAME)"
+	$(Q)$(foreach F,$(SLIB_INSTALL_LINKS),(cd "$(SHLIBDIR)" && $(LN_S) $(SLIB_INSTALL_NAME) $(F));)
+	$(if $(SLIB_INSTALL_EXTRA_SHLIB),$$(INSTALL) -m 644 $(SLIB_INSTALL_EXTRA_SHLIB:%=$(SUBDIR)%) "$(SHLIBDIR)")
+	$(if $(SLIB_INSTALL_EXTRA_LIB),$(Q)mkdir -p "$(LIBDIR)")
+	$(if $(SLIB_INSTALL_EXTRA_LIB),$$(INSTALL) -m 644 $(SLIB_INSTALL_EXTRA_LIB:%=$(SUBDIR)%) "$(LIBDIR)")
+
+install-lib$(NAME)-static: $(SUBDIR)$(LIBNAME)
+	$(Q)mkdir -p "$(LIBDIR)"
+	$$(INSTALL) -m 644 $$< "$(LIBDIR)"
+	$(LIB_INSTALL_EXTRA_CMD)
+
+install-lib$(NAME)-headers: $(addprefix $(SUBDIR),$(HEADERS) $(BUILT_HEADERS))
+	$(Q)mkdir -p "$(INCINSTDIR)"
+	$$(INSTALL) -m 644 $$^ "$(INCINSTDIR)"
+
+install-lib$(NAME)-pkgconfig: $(SUBDIR)lib$(FULLNAME).pc
+	$(Q)mkdir -p "$(PKGCONFIGDIR)"
+	$$(INSTALL) -m 644 $$^ "$(PKGCONFIGDIR)"
+
+uninstall-libs::
+	-$(RM) "$(SHLIBDIR)/$(SLIBNAME_WITH_MAJOR)" \
+	       "$(SHLIBDIR)/$(SLIBNAME)"            \
+	       "$(SHLIBDIR)/$(SLIBNAME_WITH_VERSION)"
+	-$(RM)  $(SLIB_INSTALL_EXTRA_SHLIB:%="$(SHLIBDIR)/%")
+	-$(RM)  $(SLIB_INSTALL_EXTRA_LIB:%="$(LIBDIR)/%")
+	-$(RM) "$(LIBDIR)/$(LIBNAME)"
+
+uninstall-headers::
+	$(RM) $(addprefix "$(INCINSTDIR)/",$(HEADERS) $(BUILT_HEADERS))
+	-rmdir "$(INCINSTDIR)"
+
+uninstall-pkgconfig::
+	$(RM) "$(PKGCONFIGDIR)/lib$(FULLNAME).pc"
+endef
+
+$(eval $(RULES))
+
+$(TOOLS):     $(DEP_LIBS) $(SUBDIR)$($(CONFIG_SHARED:yes=S)LIBNAME)
+$(TESTPROGS): $(DEP_LIBS) $(SUBDIR)$(LIBNAME)
+
+testprogs: $(TESTPROGS)

trunk/3rdparty/ffmpeg-4-fit/ffbuild/libversion.sh

@@ -0,0 +1,13 @@
+toupper(){
+    echo "$@" | tr abcdefghijklmnopqrstuvwxyz ABCDEFGHIJKLMNOPQRSTUVWXYZ
+}
+
+name=lib$1
+ucname=$(toupper ${name})
+file=$2
+
+eval $(awk "/#define ${ucname}_VERSION_M/ { print \$2 \"=\" \$3 }" "$file")
+eval ${ucname}_VERSION=\$${ucname}_VERSION_MAJOR.\$${ucname}_VERSION_MINOR.\$${ucname}_VERSION_MICRO
+eval echo "${name}_VERSION=\$${ucname}_VERSION"
+eval echo "${name}_VERSION_MAJOR=\$${ucname}_VERSION_MAJOR"
+eval echo "${name}_VERSION_MINOR=\$${ucname}_VERSION_MINOR"

trunk/3rdparty/ffmpeg-4-fit/ffbuild/pkgconfig_generate.sh

@@ -0,0 +1,62 @@
+#!/bin/sh
+
+. ffbuild/config.sh
+
+if test "$shared" = "yes"; then
+    shared=true
+else
+    shared=false
+fi
+
+shortname=$1
+name=lib${shortname}
+fullname=${name}${build_suffix}
+comment=$2
+libs=$(eval echo \$extralibs_${shortname})
+deps=$(eval echo \$${shortname}_deps)
+
+for dep in $deps; do
+    depname=lib${dep}
+    fulldepname=${depname}${build_suffix}
+    . ${depname}/${depname}.version
+    depversion=$(eval echo \$${depname}_VERSION)
+    requires="$requires ${fulldepname} >= ${depversion}, "
+done
+requires=${requires%, }
+
+version=$(grep ${name}_VERSION= $name/${name}.version | cut -d= -f2)
+
+cat <<EOF > $name/$fullname.pc
+prefix=$prefix
+exec_prefix=\${prefix}
+libdir=$libdir
+includedir=$incdir
+
+Name: $fullname
+Description: $comment
+Version: $version
+Requires: $($shared || echo $requires)
+Requires.private: $($shared && echo $requires)
+Conflicts:
+Libs: -L\${libdir} $rpath -l${fullname#lib} $($shared || echo $libs)
+Libs.private: $($shared && echo $libs)
+Cflags: -I\${includedir}
+EOF
+
+mkdir -p doc/examples/pc-uninstalled
+includedir=${source_path}
+[ "$includedir" = . ] && includedir="\${pcfiledir}/../../.."
+    cat <<EOF > doc/examples/pc-uninstalled/${name}-uninstalled.pc
+prefix=
+exec_prefix=
+libdir=\${pcfiledir}/../../../$name
+includedir=${source_path}
+
+Name: $fullname
+Description: $comment
+Version: $version
+Requires: $requires
+Conflicts:
+Libs: -L\${libdir} -Wl,-rpath,\${libdir} -l${fullname#lib} $($shared || echo $libs)
+Cflags: -I\${includedir}
+EOF

trunk/3rdparty/ffmpeg-4-fit/ffbuild/version.sh

@@ -0,0 +1,63 @@
+#!/bin/sh
+
+# Usage: version.sh <ffmpeg-root-dir> <output-version.h> <extra-version>
+
+# check for git short hash
+if ! test "$revision"; then
+    if (cd "$1" && grep git RELEASE 2> /dev/null >/dev/null) ; then
+        revision=$(cd "$1" && git describe --tags --match N 2> /dev/null)
+    else
+        revision=$(cd "$1" && git describe --tags --always 2> /dev/null)
+    fi
+fi
+
+# Shallow Git clones (--depth) do not have the N tag:
+# use 'git-YYYY-MM-DD-hhhhhhh'.
+test "$revision" || revision=$(cd "$1" &&
+  git log -1 --pretty=format:"git-%cd-%h" --date=short 2> /dev/null)
+
+# Snapshots from gitweb are in a directory called ffmpeg-hhhhhhh or
+# ffmpeg-HEAD-hhhhhhh.
+if [ -z "$revision" ]; then
+  srcdir=$(cd "$1" && pwd)
+  case "$srcdir" in
+    */ffmpeg-[0-9a-f][0-9a-f][0-9a-f][0-9a-f][0-9a-f][0-9a-f][0-9a-f])
+      git_hash="${srcdir##*-}";;
+    */ffmpeg-HEAD-[0-9a-f][0-9a-f][0-9a-f][0-9a-f][0-9a-f][0-9a-f][0-9a-f])
+      git_hash="${srcdir##*-}";;
+  esac
+fi
+
+# no revision number found
+test "$revision" || revision=$(cd "$1" && cat RELEASE 2> /dev/null)
+
+# Append the Git hash if we have one
+test "$revision" && test "$git_hash" && revision="$revision-$git_hash"
+
+# releases extract the version number from the VERSION file
+version=$(cd "$1" && cat VERSION 2> /dev/null)
+test "$version" || version=$revision
+
+test -n "$3" && version=$version-$3
+
+if [ -z "$2" ]; then
+    echo "$version"
+    exit
+fi
+
+NEW_REVISION="#define FFMPEG_VERSION \"$version\""
+OLD_REVISION=$(cat "$2" 2> /dev/null | head -4 | tail -1)
+
+# String used for preprocessor guard
+GUARD=$(echo "$2" | sed 's/\//_/' | sed 's/\./_/' | tr '[:lower:]' '[:upper:]' | sed 's/LIB//')
+
+# Update version header only on revision changes to avoid spurious rebuilds
+if test "$NEW_REVISION" != "$OLD_REVISION"; then
+    cat << EOF > "$2"
+/* Automatically generated by version.sh, do not manually edit! */
+#ifndef $GUARD
+#define $GUARD
+$NEW_REVISION
+#endif /* $GUARD */
+EOF
+fi

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aac.h

@@ -0,0 +1,377 @@
+/*
+ * AAC definitions and structures
+ * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
+ * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC definitions and structures
+ * @author Oded Shimon  ( ods15 ods15 dyndns org )
+ * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
+ */
+
+#ifndef AVCODEC_AAC_H
+#define AVCODEC_AAC_H
+
+
+#include "aac_defines.h"
+#include "libavutil/float_dsp.h"
+#include "libavutil/fixed_dsp.h"
+#include "avcodec.h"
+#if !USE_FIXED
+#include "mdct15.h"
+#endif
+#include "fft.h"
+#include "mpeg4audio.h"
+#include "sbr.h"
+
+#include <stdint.h>
+
+#define MAX_CHANNELS 64
+#define MAX_ELEM_ID 16
+
+#define TNS_MAX_ORDER 20
+#define MAX_LTP_LONG_SFB 40
+
+#define CLIP_AVOIDANCE_FACTOR 0.95f
+
+enum RawDataBlockType {
+    TYPE_SCE,
+    TYPE_CPE,
+    TYPE_CCE,
+    TYPE_LFE,
+    TYPE_DSE,
+    TYPE_PCE,
+    TYPE_FIL,
+    TYPE_END,
+};
+
+enum ExtensionPayloadID {
+    EXT_FILL,
+    EXT_FILL_DATA,
+    EXT_DATA_ELEMENT,
+    EXT_DYNAMIC_RANGE = 0xb,
+    EXT_SBR_DATA      = 0xd,
+    EXT_SBR_DATA_CRC  = 0xe,
+};
+
+enum WindowSequence {
+    ONLY_LONG_SEQUENCE,
+    LONG_START_SEQUENCE,
+    EIGHT_SHORT_SEQUENCE,
+    LONG_STOP_SEQUENCE,
+};
+
+enum BandType {
+    ZERO_BT        = 0,     ///< Scalefactors and spectral data are all zero.
+    FIRST_PAIR_BT  = 5,     ///< This and later band types encode two values (rather than four) with one code word.
+    ESC_BT         = 11,    ///< Spectral data are coded with an escape sequence.
+    RESERVED_BT    = 12,    ///< Band types following are encoded differently from others.
+    NOISE_BT       = 13,    ///< Spectral data are scaled white noise not coded in the bitstream.
+    INTENSITY_BT2  = 14,    ///< Scalefactor data are intensity stereo positions (out of phase).
+    INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions (in phase).
+};
+
+#define IS_CODEBOOK_UNSIGNED(x) (((x) - 1) & 10)
+
+enum ChannelPosition {
+    AAC_CHANNEL_OFF   = 0,
+    AAC_CHANNEL_FRONT = 1,
+    AAC_CHANNEL_SIDE  = 2,
+    AAC_CHANNEL_BACK  = 3,
+    AAC_CHANNEL_LFE   = 4,
+    AAC_CHANNEL_CC    = 5,
+};
+
+/**
+ * The point during decoding at which channel coupling is applied.
+ */
+enum CouplingPoint {
+    BEFORE_TNS,
+    BETWEEN_TNS_AND_IMDCT,
+    AFTER_IMDCT = 3,
+};
+
+/**
+ * Output configuration status
+ */
+enum OCStatus {
+    OC_NONE,        ///< Output unconfigured
+    OC_TRIAL_PCE,   ///< Output configuration under trial specified by an inband PCE
+    OC_TRIAL_FRAME, ///< Output configuration under trial specified by a frame header
+    OC_GLOBAL_HDR,  ///< Output configuration set in a global header but not yet locked
+    OC_LOCKED,      ///< Output configuration locked in place
+};
+
+typedef struct OutputConfiguration {
+    MPEG4AudioConfig m4ac;
+    uint8_t layout_map[MAX_ELEM_ID*4][3];
+    int layout_map_tags;
+    int channels;
+    uint64_t channel_layout;
+    enum OCStatus status;
+} OutputConfiguration;
+
+/**
+ * Predictor State
+ */
+typedef struct PredictorState {
+    AAC_FLOAT cor0;
+    AAC_FLOAT cor1;
+    AAC_FLOAT var0;
+    AAC_FLOAT var1;
+    AAC_FLOAT r0;
+    AAC_FLOAT r1;
+    AAC_FLOAT k1;
+    AAC_FLOAT x_est;
+} PredictorState;
+
+#define MAX_PREDICTORS 672
+
+#define SCALE_DIV_512    36    ///< scalefactor difference that corresponds to scale difference in 512 times
+#define SCALE_ONE_POS   140    ///< scalefactor index that corresponds to scale=1.0
+#define SCALE_MAX_POS   255    ///< scalefactor index maximum value
+#define SCALE_MAX_DIFF   60    ///< maximum scalefactor difference allowed by standard
+#define SCALE_DIFF_ZERO  60    ///< codebook index corresponding to zero scalefactor indices difference
+
+#define POW_SF2_ZERO    200    ///< ff_aac_pow2sf_tab index corresponding to pow(2, 0);
+
+#define NOISE_PRE       256    ///< preamble for NOISE_BT, put in bitstream with the first noise band
+#define NOISE_PRE_BITS    9    ///< length of preamble
+#define NOISE_OFFSET     90    ///< subtracted from global gain, used as offset for the preamble
+
+/**
+ * Long Term Prediction
+ */
+typedef struct LongTermPrediction {
+    int8_t present;
+    int16_t lag;
+    int coef_idx;
+    INTFLOAT coef;
+    int8_t used[MAX_LTP_LONG_SFB];
+} LongTermPrediction;
+
+/**
+ * Individual Channel Stream
+ */
+typedef struct IndividualChannelStream {
+    uint8_t max_sfb;            ///< number of scalefactor bands per group
+    enum WindowSequence window_sequence[2];
+    uint8_t use_kb_window[2];   ///< If set, use Kaiser-Bessel window, otherwise use a sine window.
+    int num_window_groups;
+    uint8_t group_len[8];
+    LongTermPrediction ltp;
+    const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
+    const uint8_t *swb_sizes;   ///< table of scalefactor band sizes for a particular window
+    int num_swb;                ///< number of scalefactor window bands
+    int num_windows;
+    int tns_max_bands;
+    int predictor_present;
+    int predictor_initialized;
+    int predictor_reset_group;
+    int predictor_reset_count[31];  ///< used by encoder to count prediction resets
+    uint8_t prediction_used[41];
+    uint8_t window_clipping[8]; ///< set if a certain window is near clipping
+    float clip_avoidance_factor; ///< set if any window is near clipping to the necessary atennuation factor to avoid it
+} IndividualChannelStream;
+
+/**
+ * Temporal Noise Shaping
+ */
+typedef struct TemporalNoiseShaping {
+    int present;
+    int n_filt[8];
+    int length[8][4];
+    int direction[8][4];
+    int order[8][4];
+    int coef_idx[8][4][TNS_MAX_ORDER];
+    INTFLOAT coef[8][4][TNS_MAX_ORDER];
+} TemporalNoiseShaping;
+
+/**
+ * Dynamic Range Control - decoded from the bitstream but not processed further.
+ */
+typedef struct DynamicRangeControl {
+    int pce_instance_tag;                           ///< Indicates with which program the DRC info is associated.
+    int dyn_rng_sgn[17];                            ///< DRC sign information; 0 - positive, 1 - negative
+    int dyn_rng_ctl[17];                            ///< DRC magnitude information
+    int exclude_mask[MAX_CHANNELS];                 ///< Channels to be excluded from DRC processing.
+    int band_incr;                                  ///< Number of DRC bands greater than 1 having DRC info.
+    int interpolation_scheme;                       ///< Indicates the interpolation scheme used in the SBR QMF domain.
+    int band_top[17];                               ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
+    int prog_ref_level;                             /**< A reference level for the long-term program audio level for all
+                                                     *   channels combined.
+                                                     */
+} DynamicRangeControl;
+
+typedef struct Pulse {
+    int num_pulse;
+    int start;
+    int pos[4];
+    int amp[4];
+} Pulse;
+
+/**
+ * coupling parameters
+ */
+typedef struct ChannelCoupling {
+    enum CouplingPoint coupling_point;  ///< The point during decoding at which coupling is applied.
+    int num_coupled;       ///< number of target elements
+    enum RawDataBlockType type[8];   ///< Type of channel element to be coupled - SCE or CPE.
+    int id_select[8];      ///< element id
+    int ch_select[8];      /**< [0] shared list of gains; [1] list of gains for right channel;
+                            *   [2] list of gains for left channel; [3] lists of gains for both channels
+                            */
+    INTFLOAT gain[16][120];
+} ChannelCoupling;
+
+/**
+ * Single Channel Element - used for both SCE and LFE elements.
+ */
+typedef struct SingleChannelElement {
+    IndividualChannelStream ics;
+    TemporalNoiseShaping tns;
+    Pulse pulse;
+    enum BandType band_type[128];                   ///< band types
+    enum BandType band_alt[128];                    ///< alternative band type (used by encoder)
+    int band_type_run_end[120];                     ///< band type run end points
+    INTFLOAT sf[120];                               ///< scalefactors
+    int sf_idx[128];                                ///< scalefactor indices (used by encoder)
+    uint8_t zeroes[128];                            ///< band is not coded (used by encoder)
+    uint8_t can_pns[128];                           ///< band is allowed to PNS (informative)
+    float  is_ener[128];                            ///< Intensity stereo pos (used by encoder)
+    float pns_ener[128];                            ///< Noise energy values (used by encoder)
+    DECLARE_ALIGNED(32, INTFLOAT, pcoeffs)[1024];   ///< coefficients for IMDCT, pristine
+    DECLARE_ALIGNED(32, INTFLOAT, coeffs)[1024];    ///< coefficients for IMDCT, maybe processed
+    DECLARE_ALIGNED(32, INTFLOAT, saved)[1536];     ///< overlap
+    DECLARE_ALIGNED(32, INTFLOAT, ret_buf)[2048];   ///< PCM output buffer
+    DECLARE_ALIGNED(16, INTFLOAT, ltp_state)[3072]; ///< time signal for LTP
+    DECLARE_ALIGNED(32, AAC_FLOAT, lcoeffs)[1024];  ///< MDCT of LTP coefficients (used by encoder)
+    DECLARE_ALIGNED(32, AAC_FLOAT, prcoeffs)[1024]; ///< Main prediction coefs (used by encoder)
+    PredictorState predictor_state[MAX_PREDICTORS];
+    INTFLOAT *ret;                                  ///< PCM output
+} SingleChannelElement;
+
+/**
+ * channel element - generic struct for SCE/CPE/CCE/LFE
+ */
+typedef struct ChannelElement {
+    int present;
+    // CPE specific
+    int common_window;        ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
+    int     ms_mode;          ///< Signals mid/side stereo flags coding mode (used by encoder)
+    uint8_t is_mode;          ///< Set if any bands have been encoded using intensity stereo (used by encoder)
+    uint8_t ms_mask[128];     ///< Set if mid/side stereo is used for each scalefactor window band
+    uint8_t is_mask[128];     ///< Set if intensity stereo is used (used by encoder)
+    // shared
+    SingleChannelElement ch[2];
+    // CCE specific
+    ChannelCoupling coup;
+    SpectralBandReplication sbr;
+} ChannelElement;
+
+/**
+ * main AAC context
+ */
+struct AACContext {
+    AVClass        *class;
+    AVCodecContext *avctx;
+    AVFrame *frame;
+
+    int is_saved;                 ///< Set if elements have stored overlap from previous frame.
+    DynamicRangeControl che_drc;
+
+    /**
+     * @name Channel element related data
+     * @{
+     */
+    ChannelElement          *che[4][MAX_ELEM_ID];
+    ChannelElement  *tag_che_map[4][MAX_ELEM_ID];
+    int tags_mapped;
+    int warned_remapping_once;
+    /** @} */
+
+    /**
+     * @name temporary aligned temporary buffers
+     * (We do not want to have these on the stack.)
+     * @{
+     */
+    DECLARE_ALIGNED(32, INTFLOAT, buf_mdct)[1024];
+    /** @} */
+
+    /**
+     * @name Computed / set up during initialization
+     * @{
+     */
+    FFTContext mdct;
+    FFTContext mdct_small;
+    FFTContext mdct_ld;
+    FFTContext mdct_ltp;
+#if USE_FIXED
+    AVFixedDSPContext *fdsp;
+#else
+    MDCT15Context *mdct120;
+    MDCT15Context *mdct480;
+    MDCT15Context *mdct960;
+    AVFloatDSPContext *fdsp;
+#endif /* USE_FIXED */
+    int random_state;
+    /** @} */
+
+    /**
+     * @name Members used for output
+     * @{
+     */
+    SingleChannelElement *output_element[MAX_CHANNELS]; ///< Points to each SingleChannelElement
+    /** @} */
+
+
+    /**
+     * @name Japanese DTV specific extension
+     * @{
+     */
+    int force_dmono_mode;///< 0->not dmono, 1->use first channel, 2->use second channel
+    int dmono_mode;      ///< 0->not dmono, 1->use first channel, 2->use second channel
+    /** @} */
+
+    DECLARE_ALIGNED(32, INTFLOAT, temp)[128];
+
+    OutputConfiguration oc[2];
+    int warned_num_aac_frames;
+    int warned_960_sbr;
+
+    int warned_gain_control;
+
+    /* aacdec functions pointers */
+    void (*imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce);
+    void (*apply_ltp)(AACContext *ac, SingleChannelElement *sce);
+    void (*apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
+                      IndividualChannelStream *ics, int decode);
+    void (*windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out,
+                                   INTFLOAT *in, IndividualChannelStream *ics);
+    void (*update_ltp)(AACContext *ac, SingleChannelElement *sce);
+    void (*vector_pow43)(int *coefs, int len);
+    void (*subband_scale)(int *dst, int *src, int scale, int offset, int len);
+
+};
+
+void ff_aacdec_init_mips(AACContext *c);
+
+#endif /* AVCODEC_AAC_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aac_ac3_parser.c

@@ -0,0 +1,105 @@
+/*
+ * Common AAC and AC-3 parser
+ * Copyright (c) 2003 Fabrice Bellard
+ * Copyright (c) 2003 Michael Niedermayer
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/common.h"
+#include "parser.h"
+#include "aac_ac3_parser.h"
+
+int ff_aac_ac3_parse(AVCodecParserContext *s1,
+                     AVCodecContext *avctx,
+                     const uint8_t **poutbuf, int *poutbuf_size,
+                     const uint8_t *buf, int buf_size)
+{
+    AACAC3ParseContext *s = s1->priv_data;
+    ParseContext *pc = &s->pc;
+    int len, i;
+    int new_frame_start;
+    int got_frame = 0;
+
+get_next:
+    i=END_NOT_FOUND;
+    if(s->remaining_size <= buf_size){
+        if(s->remaining_size && !s->need_next_header){
+            i= s->remaining_size;
+            s->remaining_size = 0;
+        }else{ //we need a header first
+            len=0;
+            for(i=s->remaining_size; i<buf_size; i++){
+                s->state = (s->state<<8) + buf[i];
+                if((len=s->sync(s->state, s, &s->need_next_header, &new_frame_start)))
+                    break;
+            }
+            if(len<=0){
+                i=END_NOT_FOUND;
+            }else{
+                got_frame = 1;
+                s->state=0;
+                i-= s->header_size -1;
+                s->remaining_size = len;
+                if(!new_frame_start || pc->index+i<=0){
+                    s->remaining_size += i;
+                    goto get_next;
+                }
+                else if (i < 0) {
+                    s->remaining_size += i;
+                }
+            }
+        }
+    }
+
+    if(ff_combine_frame(pc, i, &buf, &buf_size)<0){
+        s->remaining_size -= FFMIN(s->remaining_size, buf_size);
+        *poutbuf = NULL;
+        *poutbuf_size = 0;
+        return buf_size;
+    }
+
+    *poutbuf = buf;
+    *poutbuf_size = buf_size;
+
+    /* update codec info */
+    if(s->codec_id)
+        avctx->codec_id = s->codec_id;
+
+    if (got_frame) {
+        /* Due to backwards compatible HE-AAC the sample rate, channel count,
+           and total number of samples found in an AAC ADTS header are not
+           reliable. Bit rate is still accurate because the total frame
+           duration in seconds is still correct (as is the number of bits in
+           the frame). */
+        if (avctx->codec_id != AV_CODEC_ID_AAC) {
+            avctx->sample_rate = s->sample_rate;
+            if (avctx->codec_id != AV_CODEC_ID_EAC3) {
+                avctx->channels = s->channels;
+                avctx->channel_layout = s->channel_layout;
+            }
+            s1->duration = s->samples;
+            avctx->audio_service_type = s->service_type;
+        }
+
+        if (avctx->codec_id != AV_CODEC_ID_EAC3)
+            avctx->bit_rate = s->bit_rate;
+    }
+
+    return i;
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aac_ac3_parser.h

@@ -0,0 +1,66 @@
+/*
+ * Common AAC and AC-3 parser prototypes
+ * Copyright (c) 2003 Fabrice Bellard
+ * Copyright (c) 2003 Michael Niedermayer
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AAC_AC3_PARSER_H
+#define AVCODEC_AAC_AC3_PARSER_H
+
+#include <stdint.h>
+#include "avcodec.h"
+#include "parser.h"
+
+typedef enum {
+    AAC_AC3_PARSE_ERROR_SYNC        = -0x1030c0a,
+    AAC_AC3_PARSE_ERROR_BSID        = -0x2030c0a,
+    AAC_AC3_PARSE_ERROR_SAMPLE_RATE = -0x3030c0a,
+    AAC_AC3_PARSE_ERROR_FRAME_SIZE  = -0x4030c0a,
+    AAC_AC3_PARSE_ERROR_FRAME_TYPE  = -0x5030c0a,
+    AAC_AC3_PARSE_ERROR_CRC         = -0x6030c0a,
+    AAC_AC3_PARSE_ERROR_CHANNEL_CFG = -0x7030c0a,
+} AACAC3ParseError;
+
+typedef struct AACAC3ParseContext {
+    ParseContext pc;
+    int frame_size;
+    int header_size;
+    int (*sync)(uint64_t state, struct AACAC3ParseContext *hdr_info,
+            int *need_next_header, int *new_frame_start);
+
+    int channels;
+    int sample_rate;
+    int bit_rate;
+    int samples;
+    uint64_t channel_layout;
+    int service_type;
+
+    int remaining_size;
+    uint64_t state;
+
+    int need_next_header;
+    enum AVCodecID codec_id;
+} AACAC3ParseContext;
+
+int ff_aac_ac3_parse(AVCodecParserContext *s1,
+                     AVCodecContext *avctx,
+                     const uint8_t **poutbuf, int *poutbuf_size,
+                     const uint8_t *buf, int buf_size);
+
+#endif /* AVCODEC_AAC_AC3_PARSER_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aac_adtstoasc_bsf.c

@@ -0,0 +1,158 @@
+/*
+ * MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration bitstream filter
+ * Copyright (c) 2009 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "adts_header.h"
+#include "adts_parser.h"
+#include "avcodec.h"
+#include "bsf.h"
+#include "put_bits.h"
+#include "get_bits.h"
+#include "mpeg4audio.h"
+#include "internal.h"
+
+typedef struct AACBSFContext {
+    int first_frame_done;
+} AACBSFContext;
+
+/**
+ * This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
+ * ADTS header and removes the ADTS header.
+ */
+static int aac_adtstoasc_filter(AVBSFContext *bsfc, AVPacket *pkt)
+{
+    AACBSFContext *ctx = bsfc->priv_data;
+
+    GetBitContext gb;
+    PutBitContext pb;
+    AACADTSHeaderInfo hdr;
+    int ret;
+
+    ret = ff_bsf_get_packet_ref(bsfc, pkt);
+    if (ret < 0)
+        return ret;
+
+    if (bsfc->par_in->extradata && pkt->size >= 2 && (AV_RB16(pkt->data) >> 4) != 0xfff)
+        return 0;
+
+    if (pkt->size < AV_AAC_ADTS_HEADER_SIZE)
+        goto packet_too_small;
+
+    init_get_bits(&gb, pkt->data, AV_AAC_ADTS_HEADER_SIZE * 8);
+
+    if (ff_adts_header_parse(&gb, &hdr) < 0) {
+        av_log(bsfc, AV_LOG_ERROR, "Error parsing ADTS frame header!\n");
+        ret = AVERROR_INVALIDDATA;
+        goto fail;
+    }
+
+    if (!hdr.crc_absent && hdr.num_aac_frames > 1) {
+        avpriv_report_missing_feature(bsfc,
+                                      "Multiple RDBs per frame with CRC");
+        ret = AVERROR_PATCHWELCOME;
+        goto fail;
+    }
+
+    pkt->size -= AV_AAC_ADTS_HEADER_SIZE + 2 * !hdr.crc_absent;
+    if (pkt->size <= 0)
+        goto packet_too_small;
+    pkt->data += AV_AAC_ADTS_HEADER_SIZE + 2 * !hdr.crc_absent;
+
+    if (!ctx->first_frame_done) {
+        int            pce_size = 0;
+        uint8_t        pce_data[MAX_PCE_SIZE];
+        uint8_t       *extradata;
+
+        if (!hdr.chan_config) {
+            init_get_bits(&gb, pkt->data, pkt->size * 8);
+            if (get_bits(&gb, 3) != 5) {
+                avpriv_report_missing_feature(bsfc,
+                                              "PCE-based channel configuration "
+                                              "without PCE as first syntax "
+                                              "element");
+                ret = AVERROR_PATCHWELCOME;
+                goto fail;
+            }
+            init_put_bits(&pb, pce_data, MAX_PCE_SIZE);
+            pce_size = ff_copy_pce_data(&pb, &gb) / 8;
+            flush_put_bits(&pb);
+            pkt->size -= get_bits_count(&gb)/8;
+            pkt->data += get_bits_count(&gb)/8;
+        }
+
+        extradata = av_packet_new_side_data(pkt, AV_PKT_DATA_NEW_EXTRADATA,
+                                            2 + pce_size);
+        if (!extradata) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
+
+        init_put_bits(&pb, extradata, 2 + pce_size);
+        put_bits(&pb, 5, hdr.object_type);
+        put_bits(&pb, 4, hdr.sampling_index);
+        put_bits(&pb, 4, hdr.chan_config);
+        put_bits(&pb, 1, 0); //frame length - 1024 samples
+        put_bits(&pb, 1, 0); //does not depend on core coder
+        put_bits(&pb, 1, 0); //is not extension
+        flush_put_bits(&pb);
+        if (pce_size) {
+            memcpy(extradata + 2, pce_data, pce_size);
+        }
+
+        ctx->first_frame_done = 1;
+    }
+
+    return 0;
+
+packet_too_small:
+    av_log(bsfc, AV_LOG_ERROR, "Input packet too small\n");
+    ret = AVERROR_INVALIDDATA;
+fail:
+    av_packet_unref(pkt);
+    return ret;
+}
+
+static int aac_adtstoasc_init(AVBSFContext *ctx)
+{
+    /* Validate the extradata if the stream is already MPEG-4 AudioSpecificConfig */
+    if (ctx->par_in->extradata) {
+        MPEG4AudioConfig mp4ac;
+        int ret = avpriv_mpeg4audio_get_config(&mp4ac, ctx->par_in->extradata,
+                                               ctx->par_in->extradata_size * 8, 1);
+        if (ret < 0) {
+            av_log(ctx, AV_LOG_ERROR, "Error parsing AudioSpecificConfig extradata!\n");
+            return ret;
+        }
+    }
+
+    return 0;
+}
+
+static const enum AVCodecID codec_ids[] = {
+    AV_CODEC_ID_AAC, AV_CODEC_ID_NONE,
+};
+
+const AVBitStreamFilter ff_aac_adtstoasc_bsf = {
+    .name           = "aac_adtstoasc",
+    .priv_data_size = sizeof(AACBSFContext),
+    .init           = aac_adtstoasc_init,
+    .filter         = aac_adtstoasc_filter,
+    .codec_ids      = codec_ids,
+};

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aac_defines.h

@@ -0,0 +1,116 @@
+/*
+ * AAC defines
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AAC_DEFINES_H
+#define AVCODEC_AAC_DEFINES_H
+
+#ifndef USE_FIXED
+#define USE_FIXED 0
+#endif
+
+#if USE_FIXED
+
+#include "libavutil/softfloat.h"
+
+#define FFT_FLOAT    0
+#define FFT_FIXED_32 1
+
+#define AAC_RENAME(x)       x ## _fixed
+#define AAC_RENAME_32(x)    x ## _fixed_32
+typedef int                 INTFLOAT;
+typedef unsigned            UINTFLOAT;  ///< Equivalent to INTFLOAT, Used as temporal cast to avoid undefined sign overflow operations.
+typedef int64_t             INT64FLOAT;
+typedef int16_t             SHORTFLOAT;
+typedef SoftFloat           AAC_FLOAT;
+typedef int                 AAC_SIGNE;
+#define FIXR(a)             ((int)((a) * 1 + 0.5))
+#define FIXR10(a)           ((int)((a) * 1024.0 + 0.5))
+#define Q23(a)              (int)((a) * 8388608.0 + 0.5)
+#define Q30(x)              (int)((x)*1073741824.0 + 0.5)
+#define Q31(x)              (int)((x)*2147483648.0 + 0.5)
+#define RANGE15(x)          x
+#define GET_GAIN(x, y)      (-(y) * (1 << (x))) + 1024
+#define AAC_MUL16(x, y)     (int)(((int64_t)(x) * (y) + 0x8000) >> 16)
+#define AAC_MUL26(x, y)     (int)(((int64_t)(x) * (y) + 0x2000000) >> 26)
+#define AAC_MUL30(x, y)     (int)(((int64_t)(x) * (y) + 0x20000000) >> 30)
+#define AAC_MUL31(x, y)     (int)(((int64_t)(x) * (y) + 0x40000000) >> 31)
+#define AAC_MADD28(x, y, a, b) (int)((((int64_t)(x) * (y)) + \
+                                      ((int64_t)(a) * (b)) + \
+                                        0x8000000) >> 28)
+#define AAC_MADD30(x, y, a, b) (int)((((int64_t)(x) * (y)) + \
+                                      ((int64_t)(a) * (b)) + \
+                                        0x20000000) >> 30)
+#define AAC_MADD30_V8(x, y, a, b, c, d, e, f) (int)((((int64_t)(x) * (y)) + \
+                                                     ((int64_t)(a) * (b)) + \
+                                                     ((int64_t)(c) * (d)) + \
+                                                     ((int64_t)(e) * (f)) + \
+                                                       0x20000000) >> 30)
+#define AAC_MSUB30(x, y, a, b) (int)((((int64_t)(x) * (y)) - \
+                                      ((int64_t)(a) * (b)) + \
+                                        0x20000000) >> 30)
+#define AAC_MSUB30_V8(x, y, a, b, c, d, e, f) (int)((((int64_t)(x) * (y)) + \
+                                                     ((int64_t)(a) * (b)) - \
+                                                     ((int64_t)(c) * (d)) - \
+                                                     ((int64_t)(e) * (f)) + \
+                                                       0x20000000) >> 30)
+#define AAC_MSUB31_V3(x, y, z)    (int)((((int64_t)(x) * (z)) - \
+                                      ((int64_t)(y) * (z)) + \
+                                        0x40000000) >> 31)
+#define AAC_HALF_SUM(x, y)  (((x) >> 1) + ((y) >> 1))
+#define AAC_SRA_R(x, y)     (int)(((x) + (1 << ((y) - 1))) >> (y))
+
+#else
+
+#define FFT_FLOAT    1
+#define FFT_FIXED_32 0
+
+#define AAC_RENAME(x)       x
+#define AAC_RENAME_32(x)    x
+typedef float               INTFLOAT;
+typedef float               UINTFLOAT;
+typedef float               INT64FLOAT;
+typedef float               SHORTFLOAT;
+typedef float               AAC_FLOAT;
+typedef unsigned            AAC_SIGNE;
+#define FIXR(x)             ((float)(x))
+#define FIXR10(x)           ((float)(x))
+#define Q23(x)              ((float)(x))
+#define Q30(x)              ((float)(x))
+#define Q31(x)              ((float)(x))
+#define RANGE15(x)          (32768.0 * (x))
+#define GET_GAIN(x, y)      powf((x), -(y))
+#define AAC_MUL16(x, y)     ((x) * (y))
+#define AAC_MUL26(x, y)     ((x) * (y))
+#define AAC_MUL30(x, y)     ((x) * (y))
+#define AAC_MUL31(x, y)     ((x) * (y))
+#define AAC_MADD28(x, y, a, b) ((x) * (y) + (a) * (b))
+#define AAC_MADD30(x, y, a, b) ((x) * (y) + (a) * (b))
+#define AAC_MADD30_V8(x, y, a, b, c, d, e, f) ((x) * (y) + (a) * (b) + \
+                                               (c) * (d) + (e) * (f))
+#define AAC_MSUB30(x, y, a, b) ((x) * (y) - (a) * (b))
+#define AAC_MSUB30_V8(x, y, a, b, c, d, e, f) ((x) * (y) + (a) * (b) - \
+                                               (c) * (d) - (e) * (f))
+#define AAC_MSUB31_V3(x, y, z)    ((x) - (y)) * (z)
+#define AAC_HALF_SUM(x, y)  ((x) + (y)) * 0.5f
+#define AAC_SRA_R(x, y)     (x)
+
+#endif /* USE_FIXED */
+
+#endif /* AVCODEC_AAC_DEFINES_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aac_parser.c

@@ -0,0 +1,71 @@
+/*
+ * Audio and Video frame extraction
+ * Copyright (c) 2003 Fabrice Bellard
+ * Copyright (c) 2003 Michael Niedermayer
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "parser.h"
+#include "aac_ac3_parser.h"
+#include "adts_header.h"
+#include "adts_parser.h"
+#include "get_bits.h"
+#include "mpeg4audio.h"
+
+static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
+        int *need_next_header, int *new_frame_start)
+{
+    GetBitContext bits;
+    AACADTSHeaderInfo hdr;
+    int size;
+    union {
+        uint64_t u64;
+        uint8_t  u8[8 + AV_INPUT_BUFFER_PADDING_SIZE];
+    } tmp;
+
+    tmp.u64 = av_be2ne64(state);
+    init_get_bits(&bits, tmp.u8 + 8 - AV_AAC_ADTS_HEADER_SIZE,
+                  AV_AAC_ADTS_HEADER_SIZE * 8);
+
+    if ((size = ff_adts_header_parse(&bits, &hdr)) < 0)
+        return 0;
+    *need_next_header = 0;
+    *new_frame_start  = 1;
+    hdr_info->sample_rate = hdr.sample_rate;
+    hdr_info->channels    = ff_mpeg4audio_channels[hdr.chan_config];
+    hdr_info->samples     = hdr.samples;
+    hdr_info->bit_rate    = hdr.bit_rate;
+    return size;
+}
+
+static av_cold int aac_parse_init(AVCodecParserContext *s1)
+{
+    AACAC3ParseContext *s = s1->priv_data;
+    s->header_size = AV_AAC_ADTS_HEADER_SIZE;
+    s->sync = aac_sync;
+    return 0;
+}
+
+
+AVCodecParser ff_aac_parser = {
+    .codec_ids      = { AV_CODEC_ID_AAC },
+    .priv_data_size = sizeof(AACAC3ParseContext),
+    .parser_init    = aac_parse_init,
+    .parser_parse   = ff_aac_ac3_parse,
+    .parser_close   = ff_parse_close,
+};

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aaccoder.c

@@ -0,0 +1,964 @@
+/*
+ * AAC coefficients encoder
+ * Copyright (C) 2008-2009 Konstantin Shishkov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC coefficients encoder
+ */
+
+/***********************************
+ *              TODOs:
+ * speedup quantizer selection
+ * add sane pulse detection
+ ***********************************/
+
+#include "libavutil/libm.h" // brought forward to work around cygwin header breakage
+
+#include <float.h>
+
+#include "libavutil/mathematics.h"
+#include "mathops.h"
+#include "avcodec.h"
+#include "put_bits.h"
+#include "aac.h"
+#include "aacenc.h"
+#include "aactab.h"
+#include "aacenctab.h"
+#include "aacenc_utils.h"
+#include "aacenc_quantization.h"
+
+#include "aacenc_is.h"
+#include "aacenc_tns.h"
+#include "aacenc_ltp.h"
+#include "aacenc_pred.h"
+
+#include "libavcodec/aaccoder_twoloop.h"
+
+/* Parameter of f(x) = a*(lambda/100), defines the maximum fourier spread
+ * beyond which no PNS is used (since the SFBs contain tone rather than noise) */
+#define NOISE_SPREAD_THRESHOLD 0.9f
+
+/* Parameter of f(x) = a*(100/lambda), defines how much PNS is allowed to
+ * replace low energy non zero bands */
+#define NOISE_LAMBDA_REPLACE 1.948f
+
+#include "libavcodec/aaccoder_trellis.h"
+
+/**
+ * structure used in optimal codebook search
+ */
+typedef struct BandCodingPath {
+    int prev_idx; ///< pointer to the previous path point
+    float cost;   ///< path cost
+    int run;
+} BandCodingPath;
+
+/**
+ * Encode band info for single window group bands.
+ */
+static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce,
+                                     int win, int group_len, const float lambda)
+{
+    BandCodingPath path[120][CB_TOT_ALL];
+    int w, swb, cb, start, size;
+    int i, j;
+    const int max_sfb  = sce->ics.max_sfb;
+    const int run_bits = sce->ics.num_windows == 1 ? 5 : 3;
+    const int run_esc  = (1 << run_bits) - 1;
+    int idx, ppos, count;
+    int stackrun[120], stackcb[120], stack_len;
+    float next_minrd = INFINITY;
+    int next_mincb = 0;
+
+    s->abs_pow34(s->scoefs, sce->coeffs, 1024);
+    start = win*128;
+    for (cb = 0; cb < CB_TOT_ALL; cb++) {
+        path[0][cb].cost     = 0.0f;
+        path[0][cb].prev_idx = -1;
+        path[0][cb].run      = 0;
+    }
+    for (swb = 0; swb < max_sfb; swb++) {
+        size = sce->ics.swb_sizes[swb];
+        if (sce->zeroes[win*16 + swb]) {
+            for (cb = 0; cb < CB_TOT_ALL; cb++) {
+                path[swb+1][cb].prev_idx = cb;
+                path[swb+1][cb].cost     = path[swb][cb].cost;
+                path[swb+1][cb].run      = path[swb][cb].run + 1;
+            }
+        } else {
+            float minrd = next_minrd;
+            int mincb = next_mincb;
+            next_minrd = INFINITY;
+            next_mincb = 0;
+            for (cb = 0; cb < CB_TOT_ALL; cb++) {
+                float cost_stay_here, cost_get_here;
+                float rd = 0.0f;
+                if (cb >= 12 && sce->band_type[win*16+swb] < aac_cb_out_map[cb] ||
+                    cb  < aac_cb_in_map[sce->band_type[win*16+swb]] && sce->band_type[win*16+swb] > aac_cb_out_map[cb]) {
+                    path[swb+1][cb].prev_idx = -1;
+                    path[swb+1][cb].cost     = INFINITY;
+                    path[swb+1][cb].run      = path[swb][cb].run + 1;
+                    continue;
+                }
+                for (w = 0; w < group_len; w++) {
+                    FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(win+w)*16+swb];
+                    rd += quantize_band_cost(s, &sce->coeffs[start + w*128],
+                                             &s->scoefs[start + w*128], size,
+                                             sce->sf_idx[(win+w)*16+swb], aac_cb_out_map[cb],
+                                             lambda / band->threshold, INFINITY, NULL, NULL, 0);
+                }
+                cost_stay_here = path[swb][cb].cost + rd;
+                cost_get_here  = minrd              + rd + run_bits + 4;
+                if (   run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run]
+                    != run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run+1])
+                    cost_stay_here += run_bits;
+                if (cost_get_here < cost_stay_here) {
+                    path[swb+1][cb].prev_idx = mincb;
+                    path[swb+1][cb].cost     = cost_get_here;
+                    path[swb+1][cb].run      = 1;
+                } else {
+                    path[swb+1][cb].prev_idx = cb;
+                    path[swb+1][cb].cost     = cost_stay_here;
+                    path[swb+1][cb].run      = path[swb][cb].run + 1;
+                }
+                if (path[swb+1][cb].cost < next_minrd) {
+                    next_minrd = path[swb+1][cb].cost;
+                    next_mincb = cb;
+                }
+            }
+        }
+        start += sce->ics.swb_sizes[swb];
+    }
+
+    //convert resulting path from backward-linked list
+    stack_len = 0;
+    idx       = 0;
+    for (cb = 1; cb < CB_TOT_ALL; cb++)
+        if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
+            idx = cb;
+    ppos = max_sfb;
+    while (ppos > 0) {
+        av_assert1(idx >= 0);
+        cb = idx;
+        stackrun[stack_len] = path[ppos][cb].run;
+        stackcb [stack_len] = cb;
+        idx = path[ppos-path[ppos][cb].run+1][cb].prev_idx;
+        ppos -= path[ppos][cb].run;
+        stack_len++;
+    }
+    //perform actual band info encoding
+    start = 0;
+    for (i = stack_len - 1; i >= 0; i--) {
+        cb = aac_cb_out_map[stackcb[i]];
+        put_bits(&s->pb, 4, cb);
+        count = stackrun[i];
+        memset(sce->zeroes + win*16 + start, !cb, count);
+        //XXX: memset when band_type is also uint8_t
+        for (j = 0; j < count; j++) {
+            sce->band_type[win*16 + start] = cb;
+            start++;
+        }
+        while (count >= run_esc) {
+            put_bits(&s->pb, run_bits, run_esc);
+            count -= run_esc;
+        }
+        put_bits(&s->pb, run_bits, count);
+    }
+}
+
+
+typedef struct TrellisPath {
+    float cost;
+    int prev;
+} TrellisPath;
+
+#define TRELLIS_STAGES 121
+#define TRELLIS_STATES (SCALE_MAX_DIFF+1)
+
+static void set_special_band_scalefactors(AACEncContext *s, SingleChannelElement *sce)
+{
+    int w, g;
+    int prevscaler_n = -255, prevscaler_i = 0;
+    int bands = 0;
+
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        for (g = 0; g < sce->ics.num_swb; g++) {
+            if (sce->zeroes[w*16+g])
+                continue;
+            if (sce->band_type[w*16+g] == INTENSITY_BT || sce->band_type[w*16+g] == INTENSITY_BT2) {
+                sce->sf_idx[w*16+g] = av_clip(roundf(log2f(sce->is_ener[w*16+g])*2), -155, 100);
+                bands++;
+            } else if (sce->band_type[w*16+g] == NOISE_BT) {
+                sce->sf_idx[w*16+g] = av_clip(3+ceilf(log2f(sce->pns_ener[w*16+g])*2), -100, 155);
+                if (prevscaler_n == -255)
+                    prevscaler_n = sce->sf_idx[w*16+g];
+                bands++;
+            }
+        }
+    }
+
+    if (!bands)
+        return;
+
+    /* Clip the scalefactor indices */
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        for (g = 0; g < sce->ics.num_swb; g++) {
+            if (sce->zeroes[w*16+g])
+                continue;
+            if (sce->band_type[w*16+g] == INTENSITY_BT || sce->band_type[w*16+g] == INTENSITY_BT2) {
+                sce->sf_idx[w*16+g] = prevscaler_i = av_clip(sce->sf_idx[w*16+g], prevscaler_i - SCALE_MAX_DIFF, prevscaler_i + SCALE_MAX_DIFF);
+            } else if (sce->band_type[w*16+g] == NOISE_BT) {
+                sce->sf_idx[w*16+g] = prevscaler_n = av_clip(sce->sf_idx[w*16+g], prevscaler_n - SCALE_MAX_DIFF, prevscaler_n + SCALE_MAX_DIFF);
+            }
+        }
+    }
+}
+
+static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
+                                       SingleChannelElement *sce,
+                                       const float lambda)
+{
+    int q, w, w2, g, start = 0;
+    int i, j;
+    int idx;
+    TrellisPath paths[TRELLIS_STAGES][TRELLIS_STATES];
+    int bandaddr[TRELLIS_STAGES];
+    int minq;
+    float mincost;
+    float q0f = FLT_MAX, q1f = 0.0f, qnrgf = 0.0f;
+    int q0, q1, qcnt = 0;
+
+    for (i = 0; i < 1024; i++) {
+        float t = fabsf(sce->coeffs[i]);
+        if (t > 0.0f) {
+            q0f = FFMIN(q0f, t);
+            q1f = FFMAX(q1f, t);
+            qnrgf += t*t;
+            qcnt++;
+        }
+    }
+
+    if (!qcnt) {
+        memset(sce->sf_idx, 0, sizeof(sce->sf_idx));
+        memset(sce->zeroes, 1, sizeof(sce->zeroes));
+        return;
+    }
+
+    //minimum scalefactor index is when minimum nonzero coefficient after quantizing is not clipped
+    q0 = av_clip(coef2minsf(q0f), 0, SCALE_MAX_POS-1);
+    //maximum scalefactor index is when maximum coefficient after quantizing is still not zero
+    q1 = av_clip(coef2maxsf(q1f), 1, SCALE_MAX_POS);
+    if (q1 - q0 > 60) {
+        int q0low  = q0;
+        int q1high = q1;
+        //minimum scalefactor index is when maximum nonzero coefficient after quantizing is not clipped
+        int qnrg = av_clip_uint8(log2f(sqrtf(qnrgf/qcnt))*4 - 31 + SCALE_ONE_POS - SCALE_DIV_512);
+        q1 = qnrg + 30;
+        q0 = qnrg - 30;
+        if (q0 < q0low) {
+            q1 += q0low - q0;
+            q0  = q0low;
+        } else if (q1 > q1high) {
+            q0 -= q1 - q1high;
+            q1  = q1high;
+        }
+    }
+    // q0 == q1 isn't really a legal situation
+    if (q0 == q1) {
+        // the following is indirect but guarantees q1 != q0 && q1 near q0
+        q1 = av_clip(q0+1, 1, SCALE_MAX_POS);
+        q0 = av_clip(q1-1, 0, SCALE_MAX_POS - 1);
+    }
+
+    for (i = 0; i < TRELLIS_STATES; i++) {
+        paths[0][i].cost    = 0.0f;
+        paths[0][i].prev    = -1;
+    }
+    for (j = 1; j < TRELLIS_STAGES; j++) {
+        for (i = 0; i < TRELLIS_STATES; i++) {
+            paths[j][i].cost    = INFINITY;
+            paths[j][i].prev    = -2;
+        }
+    }
+    idx = 1;
+    s->abs_pow34(s->scoefs, sce->coeffs, 1024);
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        start = w*128;
+        for (g = 0; g < sce->ics.num_swb; g++) {
+            const float *coefs = &sce->coeffs[start];
+            float qmin, qmax;
+            int nz = 0;
+
+            bandaddr[idx] = w * 16 + g;
+            qmin = INT_MAX;
+            qmax = 0.0f;
+            for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
+                if (band->energy <= band->threshold || band->threshold == 0.0f) {
+                    sce->zeroes[(w+w2)*16+g] = 1;
+                    continue;
+                }
+                sce->zeroes[(w+w2)*16+g] = 0;
+                nz = 1;
+                for (i = 0; i < sce->ics.swb_sizes[g]; i++) {
+                    float t = fabsf(coefs[w2*128+i]);
+                    if (t > 0.0f)
+                        qmin = FFMIN(qmin, t);
+                    qmax = FFMAX(qmax, t);
+                }
+            }
+            if (nz) {
+                int minscale, maxscale;
+                float minrd = INFINITY;
+                float maxval;
+                //minimum scalefactor index is when minimum nonzero coefficient after quantizing is not clipped
+                minscale = coef2minsf(qmin);
+                //maximum scalefactor index is when maximum coefficient after quantizing is still not zero
+                maxscale = coef2maxsf(qmax);
+                minscale = av_clip(minscale - q0, 0, TRELLIS_STATES - 1);
+                maxscale = av_clip(maxscale - q0, 0, TRELLIS_STATES);
+                if (minscale == maxscale) {
+                    maxscale = av_clip(minscale+1, 1, TRELLIS_STATES);
+                    minscale = av_clip(maxscale-1, 0, TRELLIS_STATES - 1);
+                }
+                maxval = find_max_val(sce->ics.group_len[w], sce->ics.swb_sizes[g], s->scoefs+start);
+                for (q = minscale; q < maxscale; q++) {
+                    float dist = 0;
+                    int cb = find_min_book(maxval, sce->sf_idx[w*16+g]);
+                    for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                        FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
+                        dist += quantize_band_cost(s, coefs + w2*128, s->scoefs + start + w2*128, sce->ics.swb_sizes[g],
+                                                   q + q0, cb, lambda / band->threshold, INFINITY, NULL, NULL, 0);
+                    }
+                    minrd = FFMIN(minrd, dist);
+
+                    for (i = 0; i < q1 - q0; i++) {
+                        float cost;
+                        cost = paths[idx - 1][i].cost + dist
+                               + ff_aac_scalefactor_bits[q - i + SCALE_DIFF_ZERO];
+                        if (cost < paths[idx][q].cost) {
+                            paths[idx][q].cost    = cost;
+                            paths[idx][q].prev    = i;
+                        }
+                    }
+                }
+            } else {
+                for (q = 0; q < q1 - q0; q++) {
+                    paths[idx][q].cost = paths[idx - 1][q].cost + 1;
+                    paths[idx][q].prev = q;
+                }
+            }
+            sce->zeroes[w*16+g] = !nz;
+            start += sce->ics.swb_sizes[g];
+            idx++;
+        }
+    }
+    idx--;
+    mincost = paths[idx][0].cost;
+    minq    = 0;
+    for (i = 1; i < TRELLIS_STATES; i++) {
+        if (paths[idx][i].cost < mincost) {
+            mincost = paths[idx][i].cost;
+            minq = i;
+        }
+    }
+    while (idx) {
+        sce->sf_idx[bandaddr[idx]] = minq + q0;
+        minq = FFMAX(paths[idx][minq].prev, 0);
+        idx--;
+    }
+    //set the same quantizers inside window groups
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
+        for (g = 0; g < sce->ics.num_swb; g++)
+            for (w2 = 1; w2 < sce->ics.group_len[w]; w2++)
+                sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g];
+}
+
+static void search_for_quantizers_fast(AVCodecContext *avctx, AACEncContext *s,
+                                       SingleChannelElement *sce,
+                                       const float lambda)
+{
+    int start = 0, i, w, w2, g;
+    int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels * (lambda / 120.f);
+    float dists[128] = { 0 }, uplims[128] = { 0 };
+    float maxvals[128];
+    int fflag, minscaler;
+    int its  = 0;
+    int allz = 0;
+    float minthr = INFINITY;
+
+    // for values above this the decoder might end up in an endless loop
+    // due to always having more bits than what can be encoded.
+    destbits = FFMIN(destbits, 5800);
+    //some heuristic to determine initial quantizers will reduce search time
+    //determine zero bands and upper limits
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        start = 0;
+        for (g = 0; g < sce->ics.num_swb; g++) {
+            int nz = 0;
+            float uplim = 0.0f, energy = 0.0f;
+            for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
+                uplim += band->threshold;
+                energy += band->energy;
+                if (band->energy <= band->threshold || band->threshold == 0.0f) {
+                    sce->zeroes[(w+w2)*16+g] = 1;
+                    continue;
+                }
+                nz = 1;
+            }
+            uplims[w*16+g] = uplim *512;
+            sce->band_type[w*16+g] = 0;
+            sce->zeroes[w*16+g] = !nz;
+            if (nz)
+                minthr = FFMIN(minthr, uplim);
+            allz |= nz;
+            start += sce->ics.swb_sizes[g];
+        }
+    }
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        for (g = 0; g < sce->ics.num_swb; g++) {
+            if (sce->zeroes[w*16+g]) {
+                sce->sf_idx[w*16+g] = SCALE_ONE_POS;
+                continue;
+            }
+            sce->sf_idx[w*16+g] = SCALE_ONE_POS + FFMIN(log2f(uplims[w*16+g]/minthr)*4,59);
+        }
+    }
+
+    if (!allz)
+        return;
+    s->abs_pow34(s->scoefs, sce->coeffs, 1024);
+    ff_quantize_band_cost_cache_init(s);
+
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        start = w*128;
+        for (g = 0; g < sce->ics.num_swb; g++) {
+            const float *scaled = s->scoefs + start;
+            maxvals[w*16+g] = find_max_val(sce->ics.group_len[w], sce->ics.swb_sizes[g], scaled);
+            start += sce->ics.swb_sizes[g];
+        }
+    }
+
+    //perform two-loop search
+    //outer loop - improve quality
+    do {
+        int tbits, qstep;
+        minscaler = sce->sf_idx[0];
+        //inner loop - quantize spectrum to fit into given number of bits
+        qstep = its ? 1 : 32;
+        do {
+            int prev = -1;
+            tbits = 0;
+            for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+                start = w*128;
+                for (g = 0; g < sce->ics.num_swb; g++) {
+                    const float *coefs = sce->coeffs + start;
+                    const float *scaled = s->scoefs + start;
+                    int bits = 0;
+                    int cb;
+                    float dist = 0.0f;
+
+                    if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
+                        start += sce->ics.swb_sizes[g];
+                        continue;
+                    }
+                    minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]);
+                    cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
+                    for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                        int b;
+                        dist += quantize_band_cost_cached(s, w + w2, g,
+                                                          coefs + w2*128,
+                                                          scaled + w2*128,
+                                                          sce->ics.swb_sizes[g],
+                                                          sce->sf_idx[w*16+g],
+                                                          cb, 1.0f, INFINITY,
+                                                          &b, NULL, 0);
+                        bits += b;
+                    }
+                    dists[w*16+g] = dist - bits;
+                    if (prev != -1) {
+                        bits += ff_aac_scalefactor_bits[sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO];
+                    }
+                    tbits += bits;
+                    start += sce->ics.swb_sizes[g];
+                    prev = sce->sf_idx[w*16+g];
+                }
+            }
+            if (tbits > destbits) {
+                for (i = 0; i < 128; i++)
+                    if (sce->sf_idx[i] < 218 - qstep)
+                        sce->sf_idx[i] += qstep;
+            } else {
+                for (i = 0; i < 128; i++)
+                    if (sce->sf_idx[i] > 60 - qstep)
+                        sce->sf_idx[i] -= qstep;
+            }
+            qstep >>= 1;
+            if (!qstep && tbits > destbits*1.02 && sce->sf_idx[0] < 217)
+                qstep = 1;
+        } while (qstep);
+
+        fflag = 0;
+        minscaler = av_clip(minscaler, 60, 255 - SCALE_MAX_DIFF);
+
+        for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+            for (g = 0; g < sce->ics.num_swb; g++) {
+                int prevsc = sce->sf_idx[w*16+g];
+                if (dists[w*16+g] > uplims[w*16+g] && sce->sf_idx[w*16+g] > 60) {
+                    if (find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]-1))
+                        sce->sf_idx[w*16+g]--;
+                    else //Try to make sure there is some energy in every band
+                        sce->sf_idx[w*16+g]-=2;
+                }
+                sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler, minscaler + SCALE_MAX_DIFF);
+                sce->sf_idx[w*16+g] = FFMIN(sce->sf_idx[w*16+g], 219);
+                if (sce->sf_idx[w*16+g] != prevsc)
+                    fflag = 1;
+                sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
+            }
+        }
+        its++;
+    } while (fflag && its < 10);
+}
+
+static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce)
+{
+    FFPsyBand *band;
+    int w, g, w2, i;
+    int wlen = 1024 / sce->ics.num_windows;
+    int bandwidth, cutoff;
+    float *PNS = &s->scoefs[0*128], *PNS34 = &s->scoefs[1*128];
+    float *NOR34 = &s->scoefs[3*128];
+    uint8_t nextband[128];
+    const float lambda = s->lambda;
+    const float freq_mult = avctx->sample_rate*0.5f/wlen;
+    const float thr_mult = NOISE_LAMBDA_REPLACE*(100.0f/lambda);
+    const float spread_threshold = FFMIN(0.75f, NOISE_SPREAD_THRESHOLD*FFMAX(0.5f, lambda/100.f));
+    const float dist_bias = av_clipf(4.f * 120 / lambda, 0.25f, 4.0f);
+    const float pns_transient_energy_r = FFMIN(0.7f, lambda / 140.f);
+
+    int refbits = avctx->bit_rate * 1024.0 / avctx->sample_rate
+        / ((avctx->flags & AV_CODEC_FLAG_QSCALE) ? 2.0f : avctx->channels)
+        * (lambda / 120.f);
+
+    /** Keep this in sync with twoloop's cutoff selection */
+    float rate_bandwidth_multiplier = 1.5f;
+    int prev = -1000, prev_sf = -1;
+    int frame_bit_rate = (avctx->flags & AV_CODEC_FLAG_QSCALE)
+        ? (refbits * rate_bandwidth_multiplier * avctx->sample_rate / 1024)
+        : (avctx->bit_rate / avctx->channels);
+
+    frame_bit_rate *= 1.15f;
+
+    if (avctx->cutoff > 0) {
+        bandwidth = avctx->cutoff;
+    } else {
+        bandwidth = FFMAX(3000, AAC_CUTOFF_FROM_BITRATE(frame_bit_rate, 1, avctx->sample_rate));
+    }
+
+    cutoff = bandwidth * 2 * wlen / avctx->sample_rate;
+
+    memcpy(sce->band_alt, sce->band_type, sizeof(sce->band_type));
+    ff_init_nextband_map(sce, nextband);
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        int wstart = w*128;
+        for (g = 0; g < sce->ics.num_swb; g++) {
+            int noise_sfi;
+            float dist1 = 0.0f, dist2 = 0.0f, noise_amp;
+            float pns_energy = 0.0f, pns_tgt_energy, energy_ratio, dist_thresh;
+            float sfb_energy = 0.0f, threshold = 0.0f, spread = 2.0f;
+            float min_energy = -1.0f, max_energy = 0.0f;
+            const int start = wstart+sce->ics.swb_offset[g];
+            const float freq = (start-wstart)*freq_mult;
+            const float freq_boost = FFMAX(0.88f*freq/NOISE_LOW_LIMIT, 1.0f);
+            if (freq < NOISE_LOW_LIMIT || (start-wstart) >= cutoff) {
+                if (!sce->zeroes[w*16+g])
+                    prev_sf = sce->sf_idx[w*16+g];
+                continue;
+            }
+            for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
+                sfb_energy += band->energy;
+                spread     = FFMIN(spread, band->spread);
+                threshold  += band->threshold;
+                if (!w2) {
+                    min_energy = max_energy = band->energy;
+                } else {
+                    min_energy = FFMIN(min_energy, band->energy);
+                    max_energy = FFMAX(max_energy, band->energy);
+                }
+            }
+
+            /* Ramps down at ~8000Hz and loosens the dist threshold */
+            dist_thresh = av_clipf(2.5f*NOISE_LOW_LIMIT/freq, 0.5f, 2.5f) * dist_bias;
+
+            /* PNS is acceptable when all of these are true:
+             * 1. high spread energy (noise-like band)
+             * 2. near-threshold energy (high PE means the random nature of PNS content will be noticed)
+             * 3. on short window groups, all windows have similar energy (variations in energy would be destroyed by PNS)
+             *
+             * At this stage, point 2 is relaxed for zeroed bands near the noise threshold (hole avoidance is more important)
+             */
+            if ((!sce->zeroes[w*16+g] && !ff_sfdelta_can_remove_band(sce, nextband, prev_sf, w*16+g)) ||
+                ((sce->zeroes[w*16+g] || !sce->band_alt[w*16+g]) && sfb_energy < threshold*sqrtf(1.0f/freq_boost)) || spread < spread_threshold ||
+                (!sce->zeroes[w*16+g] && sce->band_alt[w*16+g] && sfb_energy > threshold*thr_mult*freq_boost) ||
+                min_energy < pns_transient_energy_r * max_energy ) {
+                sce->pns_ener[w*16+g] = sfb_energy;
+                if (!sce->zeroes[w*16+g])
+                    prev_sf = sce->sf_idx[w*16+g];
+                continue;
+            }
+
+            pns_tgt_energy = sfb_energy*FFMIN(1.0f, spread*spread);
+            noise_sfi = av_clip(roundf(log2f(pns_tgt_energy)*2), -100, 155); /* Quantize */
+            noise_amp = -ff_aac_pow2sf_tab[noise_sfi + POW_SF2_ZERO];    /* Dequantize */
+            if (prev != -1000) {
+                int noise_sfdiff = noise_sfi - prev + SCALE_DIFF_ZERO;
+                if (noise_sfdiff < 0 || noise_sfdiff > 2*SCALE_MAX_DIFF) {
+                    if (!sce->zeroes[w*16+g])
+                        prev_sf = sce->sf_idx[w*16+g];
+                    continue;
+                }
+            }
+            for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                float band_energy, scale, pns_senergy;
+                const int start_c = (w+w2)*128+sce->ics.swb_offset[g];
+                band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
+                for (i = 0; i < sce->ics.swb_sizes[g]; i++) {
+                    s->random_state  = lcg_random(s->random_state);
+                    PNS[i] = s->random_state;
+                }
+                band_energy = s->fdsp->scalarproduct_float(PNS, PNS, sce->ics.swb_sizes[g]);
+                scale = noise_amp/sqrtf(band_energy);
+                s->fdsp->vector_fmul_scalar(PNS, PNS, scale, sce->ics.swb_sizes[g]);
+                pns_senergy = s->fdsp->scalarproduct_float(PNS, PNS, sce->ics.swb_sizes[g]);
+                pns_energy += pns_senergy;
+                s->abs_pow34(NOR34, &sce->coeffs[start_c], sce->ics.swb_sizes[g]);
+                s->abs_pow34(PNS34, PNS, sce->ics.swb_sizes[g]);
+                dist1 += quantize_band_cost(s, &sce->coeffs[start_c],
+                                            NOR34,
+                                            sce->ics.swb_sizes[g],
+                                            sce->sf_idx[(w+w2)*16+g],
+                                            sce->band_alt[(w+w2)*16+g],
+                                            lambda/band->threshold, INFINITY, NULL, NULL, 0);
+                /* Estimate rd on average as 5 bits for SF, 4 for the CB, plus spread energy * lambda/thr */
+                dist2 += band->energy/(band->spread*band->spread)*lambda*dist_thresh/band->threshold;
+            }
+            if (g && sce->band_type[w*16+g-1] == NOISE_BT) {
+                dist2 += 5;
+            } else {
+                dist2 += 9;
+            }
+            energy_ratio = pns_tgt_energy/pns_energy; /* Compensates for quantization error */
+            sce->pns_ener[w*16+g] = energy_ratio*pns_tgt_energy;
+            if (sce->zeroes[w*16+g] || !sce->band_alt[w*16+g] || (energy_ratio > 0.85f && energy_ratio < 1.25f && dist2 < dist1)) {
+                sce->band_type[w*16+g] = NOISE_BT;
+                sce->zeroes[w*16+g] = 0;
+                prev = noise_sfi;
+            } else {
+                if (!sce->zeroes[w*16+g])
+                    prev_sf = sce->sf_idx[w*16+g];
+            }
+        }
+    }
+}
+
+static void mark_pns(AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce)
+{
+    FFPsyBand *band;
+    int w, g, w2;
+    int wlen = 1024 / sce->ics.num_windows;
+    int bandwidth, cutoff;
+    const float lambda = s->lambda;
+    const float freq_mult = avctx->sample_rate*0.5f/wlen;
+    const float spread_threshold = FFMIN(0.75f, NOISE_SPREAD_THRESHOLD*FFMAX(0.5f, lambda/100.f));
+    const float pns_transient_energy_r = FFMIN(0.7f, lambda / 140.f);
+
+    int refbits = avctx->bit_rate * 1024.0 / avctx->sample_rate
+        / ((avctx->flags & AV_CODEC_FLAG_QSCALE) ? 2.0f : avctx->channels)
+        * (lambda / 120.f);
+
+    /** Keep this in sync with twoloop's cutoff selection */
+    float rate_bandwidth_multiplier = 1.5f;
+    int frame_bit_rate = (avctx->flags & AV_CODEC_FLAG_QSCALE)
+        ? (refbits * rate_bandwidth_multiplier * avctx->sample_rate / 1024)
+        : (avctx->bit_rate / avctx->channels);
+
+    frame_bit_rate *= 1.15f;
+
+    if (avctx->cutoff > 0) {
+        bandwidth = avctx->cutoff;
+    } else {
+        bandwidth = FFMAX(3000, AAC_CUTOFF_FROM_BITRATE(frame_bit_rate, 1, avctx->sample_rate));
+    }
+
+    cutoff = bandwidth * 2 * wlen / avctx->sample_rate;
+
+    memcpy(sce->band_alt, sce->band_type, sizeof(sce->band_type));
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        for (g = 0; g < sce->ics.num_swb; g++) {
+            float sfb_energy = 0.0f, threshold = 0.0f, spread = 2.0f;
+            float min_energy = -1.0f, max_energy = 0.0f;
+            const int start = sce->ics.swb_offset[g];
+            const float freq = start*freq_mult;
+            const float freq_boost = FFMAX(0.88f*freq/NOISE_LOW_LIMIT, 1.0f);
+            if (freq < NOISE_LOW_LIMIT || start >= cutoff) {
+                sce->can_pns[w*16+g] = 0;
+                continue;
+            }
+            for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
+                sfb_energy += band->energy;
+                spread     = FFMIN(spread, band->spread);
+                threshold  += band->threshold;
+                if (!w2) {
+                    min_energy = max_energy = band->energy;
+                } else {
+                    min_energy = FFMIN(min_energy, band->energy);
+                    max_energy = FFMAX(max_energy, band->energy);
+                }
+            }
+
+            /* PNS is acceptable when all of these are true:
+             * 1. high spread energy (noise-like band)
+             * 2. near-threshold energy (high PE means the random nature of PNS content will be noticed)
+             * 3. on short window groups, all windows have similar energy (variations in energy would be destroyed by PNS)
+             */
+            sce->pns_ener[w*16+g] = sfb_energy;
+            if (sfb_energy < threshold*sqrtf(1.5f/freq_boost) || spread < spread_threshold || min_energy < pns_transient_energy_r * max_energy) {
+                sce->can_pns[w*16+g] = 0;
+            } else {
+                sce->can_pns[w*16+g] = 1;
+            }
+        }
+    }
+}
+
+static void search_for_ms(AACEncContext *s, ChannelElement *cpe)
+{
+    int start = 0, i, w, w2, g, sid_sf_boost, prev_mid, prev_side;
+    uint8_t nextband0[128], nextband1[128];
+    float *M   = s->scoefs + 128*0, *S   = s->scoefs + 128*1;
+    float *L34 = s->scoefs + 128*2, *R34 = s->scoefs + 128*3;
+    float *M34 = s->scoefs + 128*4, *S34 = s->scoefs + 128*5;
+    const float lambda = s->lambda;
+    const float mslambda = FFMIN(1.0f, lambda / 120.f);
+    SingleChannelElement *sce0 = &cpe->ch[0];
+    SingleChannelElement *sce1 = &cpe->ch[1];
+    if (!cpe->common_window)
+        return;
+
+    /** Scout out next nonzero bands */
+    ff_init_nextband_map(sce0, nextband0);
+    ff_init_nextband_map(sce1, nextband1);
+
+    prev_mid = sce0->sf_idx[0];
+    prev_side = sce1->sf_idx[0];
+    for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
+        start = 0;
+        for (g = 0; g < sce0->ics.num_swb; g++) {
+            float bmax = bval2bmax(g * 17.0f / sce0->ics.num_swb) / 0.0045f;
+            if (!cpe->is_mask[w*16+g])
+                cpe->ms_mask[w*16+g] = 0;
+            if (!sce0->zeroes[w*16+g] && !sce1->zeroes[w*16+g] && !cpe->is_mask[w*16+g]) {
+                float Mmax = 0.0f, Smax = 0.0f;
+
+                /* Must compute mid/side SF and book for the whole window group */
+                for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
+                    for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
+                        M[i] = (sce0->coeffs[start+(w+w2)*128+i]
+                              + sce1->coeffs[start+(w+w2)*128+i]) * 0.5;
+                        S[i] =  M[i]
+                              - sce1->coeffs[start+(w+w2)*128+i];
+                    }
+                    s->abs_pow34(M34, M, sce0->ics.swb_sizes[g]);
+                    s->abs_pow34(S34, S, sce0->ics.swb_sizes[g]);
+                    for (i = 0; i < sce0->ics.swb_sizes[g]; i++ ) {
+                        Mmax = FFMAX(Mmax, M34[i]);
+                        Smax = FFMAX(Smax, S34[i]);
+                    }
+                }
+
+                for (sid_sf_boost = 0; sid_sf_boost < 4; sid_sf_boost++) {
+                    float dist1 = 0.0f, dist2 = 0.0f;
+                    int B0 = 0, B1 = 0;
+                    int minidx;
+                    int mididx, sididx;
+                    int midcb, sidcb;
+
+                    minidx = FFMIN(sce0->sf_idx[w*16+g], sce1->sf_idx[w*16+g]);
+                    mididx = av_clip(minidx, 0, SCALE_MAX_POS - SCALE_DIV_512);
+                    sididx = av_clip(minidx - sid_sf_boost * 3, 0, SCALE_MAX_POS - SCALE_DIV_512);
+                    if (sce0->band_type[w*16+g] != NOISE_BT && sce1->band_type[w*16+g] != NOISE_BT
+                        && (   !ff_sfdelta_can_replace(sce0, nextband0, prev_mid, mididx, w*16+g)
+                            || !ff_sfdelta_can_replace(sce1, nextband1, prev_side, sididx, w*16+g))) {
+                        /* scalefactor range violation, bad stuff, will decrease quality unacceptably */
+                        continue;
+                    }
+
+                    midcb = find_min_book(Mmax, mididx);
+                    sidcb = find_min_book(Smax, sididx);
+
+                    /* No CB can be zero */
+                    midcb = FFMAX(1,midcb);
+                    sidcb = FFMAX(1,sidcb);
+
+                    for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
+                        FFPsyBand *band0 = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
+                        FFPsyBand *band1 = &s->psy.ch[s->cur_channel+1].psy_bands[(w+w2)*16+g];
+                        float minthr = FFMIN(band0->threshold, band1->threshold);
+                        int b1,b2,b3,b4;
+                        for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
+                            M[i] = (sce0->coeffs[start+(w+w2)*128+i]
+                                  + sce1->coeffs[start+(w+w2)*128+i]) * 0.5;
+                            S[i] =  M[i]
+                                  - sce1->coeffs[start+(w+w2)*128+i];
+                        }
+
+                        s->abs_pow34(L34, sce0->coeffs+start+(w+w2)*128, sce0->ics.swb_sizes[g]);
+                        s->abs_pow34(R34, sce1->coeffs+start+(w+w2)*128, sce0->ics.swb_sizes[g]);
+                        s->abs_pow34(M34, M,                         sce0->ics.swb_sizes[g]);
+                        s->abs_pow34(S34, S,                         sce0->ics.swb_sizes[g]);
+                        dist1 += quantize_band_cost(s, &sce0->coeffs[start + (w+w2)*128],
+                                                    L34,
+                                                    sce0->ics.swb_sizes[g],
+                                                    sce0->sf_idx[w*16+g],
+                                                    sce0->band_type[w*16+g],
+                                                    lambda / band0->threshold, INFINITY, &b1, NULL, 0);
+                        dist1 += quantize_band_cost(s, &sce1->coeffs[start + (w+w2)*128],
+                                                    R34,
+                                                    sce1->ics.swb_sizes[g],
+                                                    sce1->sf_idx[w*16+g],
+                                                    sce1->band_type[w*16+g],
+                                                    lambda / band1->threshold, INFINITY, &b2, NULL, 0);
+                        dist2 += quantize_band_cost(s, M,
+                                                    M34,
+                                                    sce0->ics.swb_sizes[g],
+                                                    mididx,
+                                                    midcb,
+                                                    lambda / minthr, INFINITY, &b3, NULL, 0);
+                        dist2 += quantize_band_cost(s, S,
+                                                    S34,
+                                                    sce1->ics.swb_sizes[g],
+                                                    sididx,
+                                                    sidcb,
+                                                    mslambda / (minthr * bmax), INFINITY, &b4, NULL, 0);
+                        B0 += b1+b2;
+                        B1 += b3+b4;
+                        dist1 -= b1+b2;
+                        dist2 -= b3+b4;
+                    }
+                    cpe->ms_mask[w*16+g] = dist2 <= dist1 && B1 < B0;
+                    if (cpe->ms_mask[w*16+g]) {
+                        if (sce0->band_type[w*16+g] != NOISE_BT && sce1->band_type[w*16+g] != NOISE_BT) {
+                            sce0->sf_idx[w*16+g] = mididx;
+                            sce1->sf_idx[w*16+g] = sididx;
+                            sce0->band_type[w*16+g] = midcb;
+                            sce1->band_type[w*16+g] = sidcb;
+                        } else if ((sce0->band_type[w*16+g] != NOISE_BT) ^ (sce1->band_type[w*16+g] != NOISE_BT)) {
+                            /* ms_mask unneeded, and it confuses some decoders */
+                            cpe->ms_mask[w*16+g] = 0;
+                        }
+                        break;
+                    } else if (B1 > B0) {
+                        /* More boost won't fix this */
+                        break;
+                    }
+                }
+            }
+            if (!sce0->zeroes[w*16+g] && sce0->band_type[w*16+g] < RESERVED_BT)
+                prev_mid = sce0->sf_idx[w*16+g];
+            if (!sce1->zeroes[w*16+g] && !cpe->is_mask[w*16+g] && sce1->band_type[w*16+g] < RESERVED_BT)
+                prev_side = sce1->sf_idx[w*16+g];
+            start += sce0->ics.swb_sizes[g];
+        }
+    }
+}
+
+const AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB] = {
+    [AAC_CODER_ANMR] = {
+        search_for_quantizers_anmr,
+        encode_window_bands_info,
+        quantize_and_encode_band,
+        ff_aac_encode_tns_info,
+        ff_aac_encode_ltp_info,
+        ff_aac_encode_main_pred,
+        ff_aac_adjust_common_pred,
+        ff_aac_adjust_common_ltp,
+        ff_aac_apply_main_pred,
+        ff_aac_apply_tns,
+        ff_aac_update_ltp,
+        ff_aac_ltp_insert_new_frame,
+        set_special_band_scalefactors,
+        search_for_pns,
+        mark_pns,
+        ff_aac_search_for_tns,
+        ff_aac_search_for_ltp,
+        search_for_ms,
+        ff_aac_search_for_is,
+        ff_aac_search_for_pred,
+    },
+    [AAC_CODER_TWOLOOP] = {
+        search_for_quantizers_twoloop,
+        codebook_trellis_rate,
+        quantize_and_encode_band,
+        ff_aac_encode_tns_info,
+        ff_aac_encode_ltp_info,
+        ff_aac_encode_main_pred,
+        ff_aac_adjust_common_pred,
+        ff_aac_adjust_common_ltp,
+        ff_aac_apply_main_pred,
+        ff_aac_apply_tns,
+        ff_aac_update_ltp,
+        ff_aac_ltp_insert_new_frame,
+        set_special_band_scalefactors,
+        search_for_pns,
+        mark_pns,
+        ff_aac_search_for_tns,
+        ff_aac_search_for_ltp,
+        search_for_ms,
+        ff_aac_search_for_is,
+        ff_aac_search_for_pred,
+    },
+    [AAC_CODER_FAST] = {
+        search_for_quantizers_fast,
+        codebook_trellis_rate,
+        quantize_and_encode_band,
+        ff_aac_encode_tns_info,
+        ff_aac_encode_ltp_info,
+        ff_aac_encode_main_pred,
+        ff_aac_adjust_common_pred,
+        ff_aac_adjust_common_ltp,
+        ff_aac_apply_main_pred,
+        ff_aac_apply_tns,
+        ff_aac_update_ltp,
+        ff_aac_ltp_insert_new_frame,
+        set_special_band_scalefactors,
+        search_for_pns,
+        mark_pns,
+        ff_aac_search_for_tns,
+        ff_aac_search_for_ltp,
+        search_for_ms,
+        ff_aac_search_for_is,
+        ff_aac_search_for_pred,
+    },
+};

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aaccoder_trellis.h

@@ -0,0 +1,192 @@
+/*
+ * AAC encoder trellis codebook selector
+ * Copyright (C) 2008-2009 Konstantin Shishkov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder trellis codebook selector
+ * @author Konstantin Shishkov
+ */
+
+/**
+ * This file contains a template for the codebook_trellis_rate selector function.
+ * It needs to be provided, externally, as an already included declaration,
+ * the following functions from aacenc_quantization/util.h. They're not included
+ * explicitly here to make it possible to provide alternative implementations:
+ *  - quantize_band_cost_bits
+ *  - abs_pow34_v
+ */
+
+#ifndef AVCODEC_AACCODER_TRELLIS_H
+#define AVCODEC_AACCODER_TRELLIS_H
+
+#include <float.h>
+#include "libavutil/mathematics.h"
+#include "avcodec.h"
+#include "put_bits.h"
+#include "aac.h"
+#include "aacenc.h"
+#include "aactab.h"
+#include "aacenctab.h"
+
+/**
+ * structure used in optimal codebook search
+ */
+typedef struct TrellisBandCodingPath {
+    int prev_idx; ///< pointer to the previous path point
+    float cost;   ///< path cost
+    int run;
+} TrellisBandCodingPath;
+
+
+static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
+                                  int win, int group_len, const float lambda)
+{
+    TrellisBandCodingPath path[120][CB_TOT_ALL];
+    int w, swb, cb, start, size;
+    int i, j;
+    const int max_sfb  = sce->ics.max_sfb;
+    const int run_bits = sce->ics.num_windows == 1 ? 5 : 3;
+    const int run_esc  = (1 << run_bits) - 1;
+    int idx, ppos, count;
+    int stackrun[120], stackcb[120], stack_len;
+    float next_minbits = INFINITY;
+    int next_mincb = 0;
+
+    s->abs_pow34(s->scoefs, sce->coeffs, 1024);
+    start = win*128;
+    for (cb = 0; cb < CB_TOT_ALL; cb++) {
+        path[0][cb].cost     = run_bits+4;
+        path[0][cb].prev_idx = -1;
+        path[0][cb].run      = 0;
+    }
+    for (swb = 0; swb < max_sfb; swb++) {
+        size = sce->ics.swb_sizes[swb];
+        if (sce->zeroes[win*16 + swb]) {
+            float cost_stay_here = path[swb][0].cost;
+            float cost_get_here  = next_minbits + run_bits + 4;
+            if (   run_value_bits[sce->ics.num_windows == 8][path[swb][0].run]
+                != run_value_bits[sce->ics.num_windows == 8][path[swb][0].run+1])
+                cost_stay_here += run_bits;
+            if (cost_get_here < cost_stay_here) {
+                path[swb+1][0].prev_idx = next_mincb;
+                path[swb+1][0].cost     = cost_get_here;
+                path[swb+1][0].run      = 1;
+            } else {
+                path[swb+1][0].prev_idx = 0;
+                path[swb+1][0].cost     = cost_stay_here;
+                path[swb+1][0].run      = path[swb][0].run + 1;
+            }
+            next_minbits = path[swb+1][0].cost;
+            next_mincb = 0;
+            for (cb = 1; cb < CB_TOT_ALL; cb++) {
+                path[swb+1][cb].cost = 61450;
+                path[swb+1][cb].prev_idx = -1;
+                path[swb+1][cb].run = 0;
+            }
+        } else {
+            float minbits = next_minbits;
+            int mincb = next_mincb;
+            int startcb = sce->band_type[win*16+swb];
+            startcb = aac_cb_in_map[startcb];
+            next_minbits = INFINITY;
+            next_mincb = 0;
+            for (cb = 0; cb < startcb; cb++) {
+                path[swb+1][cb].cost = 61450;
+                path[swb+1][cb].prev_idx = -1;
+                path[swb+1][cb].run = 0;
+            }
+            for (cb = startcb; cb < CB_TOT_ALL; cb++) {
+                float cost_stay_here, cost_get_here;
+                float bits = 0.0f;
+                if (cb >= 12 && sce->band_type[win*16+swb] != aac_cb_out_map[cb]) {
+                    path[swb+1][cb].cost = 61450;
+                    path[swb+1][cb].prev_idx = -1;
+                    path[swb+1][cb].run = 0;
+                    continue;
+                }
+                for (w = 0; w < group_len; w++) {
+                    bits += quantize_band_cost_bits(s, &sce->coeffs[start + w*128],
+                                               &s->scoefs[start + w*128], size,
+                                               sce->sf_idx[win*16+swb],
+                                               aac_cb_out_map[cb],
+                                               0, INFINITY, NULL, NULL, 0);
+                }
+                cost_stay_here = path[swb][cb].cost + bits;
+                cost_get_here  = minbits            + bits + run_bits + 4;
+                if (   run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run]
+                    != run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run+1])
+                    cost_stay_here += run_bits;
+                if (cost_get_here < cost_stay_here) {
+                    path[swb+1][cb].prev_idx = mincb;
+                    path[swb+1][cb].cost     = cost_get_here;
+                    path[swb+1][cb].run      = 1;
+                } else {
+                    path[swb+1][cb].prev_idx = cb;
+                    path[swb+1][cb].cost     = cost_stay_here;
+                    path[swb+1][cb].run      = path[swb][cb].run + 1;
+                }
+                if (path[swb+1][cb].cost < next_minbits) {
+                    next_minbits = path[swb+1][cb].cost;
+                    next_mincb = cb;
+                }
+            }
+        }
+        start += sce->ics.swb_sizes[swb];
+    }
+
+    //convert resulting path from backward-linked list
+    stack_len = 0;
+    idx       = 0;
+    for (cb = 1; cb < CB_TOT_ALL; cb++)
+        if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
+            idx = cb;
+    ppos = max_sfb;
+    while (ppos > 0) {
+        av_assert1(idx >= 0);
+        cb = idx;
+        stackrun[stack_len] = path[ppos][cb].run;
+        stackcb [stack_len] = cb;
+        idx = path[ppos-path[ppos][cb].run+1][cb].prev_idx;
+        ppos -= path[ppos][cb].run;
+        stack_len++;
+    }
+    //perform actual band info encoding
+    start = 0;
+    for (i = stack_len - 1; i >= 0; i--) {
+        cb = aac_cb_out_map[stackcb[i]];
+        put_bits(&s->pb, 4, cb);
+        count = stackrun[i];
+        memset(sce->zeroes + win*16 + start, !cb, count);
+        //XXX: memset when band_type is also uint8_t
+        for (j = 0; j < count; j++) {
+            sce->band_type[win*16 + start] = cb;
+            start++;
+        }
+        while (count >= run_esc) {
+            put_bits(&s->pb, run_bits, run_esc);
+            count -= run_esc;
+        }
+        put_bits(&s->pb, run_bits, count);
+    }
+}
+
+
+#endif /* AVCODEC_AACCODER_TRELLIS_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aaccoder_twoloop.h

@@ -0,0 +1,763 @@
+/*
+ * AAC encoder twoloop coder
+ * Copyright (C) 2008-2009 Konstantin Shishkov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder twoloop coder
+ * @author Konstantin Shishkov, Claudio Freire
+ */
+
+/**
+ * This file contains a template for the twoloop coder function.
+ * It needs to be provided, externally, as an already included declaration,
+ * the following functions from aacenc_quantization/util.h. They're not included
+ * explicitly here to make it possible to provide alternative implementations:
+ *  - quantize_band_cost
+ *  - abs_pow34_v
+ *  - find_max_val
+ *  - find_min_book
+ *  - find_form_factor
+ */
+
+#ifndef AVCODEC_AACCODER_TWOLOOP_H
+#define AVCODEC_AACCODER_TWOLOOP_H
+
+#include <float.h>
+#include "libavutil/mathematics.h"
+#include "mathops.h"
+#include "avcodec.h"
+#include "put_bits.h"
+#include "aac.h"
+#include "aacenc.h"
+#include "aactab.h"
+#include "aacenctab.h"
+
+/** Frequency in Hz for lower limit of noise substitution **/
+#define NOISE_LOW_LIMIT 4000
+
+#define sclip(x) av_clip(x,60,218)
+
+/* Reflects the cost to change codebooks */
+static inline int ff_pns_bits(SingleChannelElement *sce, int w, int g)
+{
+    return (!g || !sce->zeroes[w*16+g-1] || !sce->can_pns[w*16+g-1]) ? 9 : 5;
+}
+
+/**
+ * two-loop quantizers search taken from ISO 13818-7 Appendix C
+ */
+static void search_for_quantizers_twoloop(AVCodecContext *avctx,
+                                          AACEncContext *s,
+                                          SingleChannelElement *sce,
+                                          const float lambda)
+{
+    int start = 0, i, w, w2, g, recomprd;
+    int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate
+        / ((avctx->flags & AV_CODEC_FLAG_QSCALE) ? 2.0f : avctx->channels)
+        * (lambda / 120.f);
+    int refbits = destbits;
+    int toomanybits, toofewbits;
+    char nzs[128];
+    uint8_t nextband[128];
+    int maxsf[128], minsf[128];
+    float dists[128] = { 0 }, qenergies[128] = { 0 }, uplims[128], euplims[128], energies[128];
+    float maxvals[128], spread_thr_r[128];
+    float min_spread_thr_r, max_spread_thr_r;
+
+    /**
+     * rdlambda controls the maximum tolerated distortion. Twoloop
+     * will keep iterating until it fails to lower it or it reaches
+     * ulimit * rdlambda. Keeping it low increases quality on difficult
+     * signals, but lower it too much, and bits will be taken from weak
+     * signals, creating "holes". A balance is necessary.
+     * rdmax and rdmin specify the relative deviation from rdlambda
+     * allowed for tonality compensation
+     */
+    float rdlambda = av_clipf(2.0f * 120.f / lambda, 0.0625f, 16.0f);
+    const float nzslope = 1.5f;
+    float rdmin = 0.03125f;
+    float rdmax = 1.0f;
+
+    /**
+     * sfoffs controls an offset of optmium allocation that will be
+     * applied based on lambda. Keep it real and modest, the loop
+     * will take care of the rest, this just accelerates convergence
+     */
+    float sfoffs = av_clipf(log2f(120.0f / lambda) * 4.0f, -5, 10);
+
+    int fflag, minscaler, maxscaler, nminscaler;
+    int its  = 0;
+    int maxits = 30;
+    int allz = 0;
+    int tbits;
+    int cutoff = 1024;
+    int pns_start_pos;
+    int prev;
+
+    /**
+     * zeroscale controls a multiplier of the threshold, if band energy
+     * is below this, a zero is forced. Keep it lower than 1, unless
+     * low lambda is used, because energy < threshold doesn't mean there's
+     * no audible signal outright, it's just energy. Also make it rise
+     * slower than rdlambda, as rdscale has due compensation with
+     * noisy band depriorization below, whereas zeroing logic is rather dumb
+     */
+    float zeroscale;
+    if (lambda > 120.f) {
+        zeroscale = av_clipf(powf(120.f / lambda, 0.25f), 0.0625f, 1.0f);
+    } else {
+        zeroscale = 1.f;
+    }
+
+    if (s->psy.bitres.alloc >= 0) {
+        /**
+         * Psy granted us extra bits to use, from the reservoire
+         * adjust for lambda except what psy already did
+         */
+        destbits = s->psy.bitres.alloc
+            * (lambda / (avctx->global_quality ? avctx->global_quality : 120));
+    }
+
+    if (avctx->flags & AV_CODEC_FLAG_QSCALE) {
+        /**
+         * Constant Q-scale doesn't compensate MS coding on its own
+         * No need to be overly precise, this only controls RD
+         * adjustment CB limits when going overboard
+         */
+        if (s->options.mid_side && s->cur_type == TYPE_CPE)
+            destbits *= 2;
+
+        /**
+         * When using a constant Q-scale, don't adjust bits, just use RD
+         * Don't let it go overboard, though... 8x psy target is enough
+         */
+        toomanybits = 5800;
+        toofewbits = destbits / 16;
+
+        /** Don't offset scalers, just RD */
+        sfoffs = sce->ics.num_windows - 1;
+        rdlambda = sqrtf(rdlambda);
+
+        /** search further */
+        maxits *= 2;
+    } else {
+        /* When using ABR, be strict, but a reasonable leeway is
+         * critical to allow RC to smoothly track desired bitrate
+         * without sudden quality drops that cause audible artifacts.
+         * Symmetry is also desirable, to avoid systematic bias.
+         */
+        toomanybits = destbits + destbits/8;
+        toofewbits = destbits - destbits/8;
+
+        sfoffs = 0;
+        rdlambda = sqrtf(rdlambda);
+    }
+
+    /** and zero out above cutoff frequency */
+    {
+        int wlen = 1024 / sce->ics.num_windows;
+        int bandwidth;
+
+        /**
+         * Scale, psy gives us constant quality, this LP only scales
+         * bitrate by lambda, so we save bits on subjectively unimportant HF
+         * rather than increase quantization noise. Adjust nominal bitrate
+         * to effective bitrate according to encoding parameters,
+         * AAC_CUTOFF_FROM_BITRATE is calibrated for effective bitrate.
+         */
+        float rate_bandwidth_multiplier = 1.5f;
+        int frame_bit_rate = (avctx->flags & AV_CODEC_FLAG_QSCALE)
+            ? (refbits * rate_bandwidth_multiplier * avctx->sample_rate / 1024)
+            : (avctx->bit_rate / avctx->channels);
+
+        /** Compensate for extensions that increase efficiency */
+        if (s->options.pns || s->options.intensity_stereo)
+            frame_bit_rate *= 1.15f;
+
+        if (avctx->cutoff > 0) {
+            bandwidth = avctx->cutoff;
+        } else {
+            bandwidth = FFMAX(3000, AAC_CUTOFF_FROM_BITRATE(frame_bit_rate, 1, avctx->sample_rate));
+            s->psy.cutoff = bandwidth;
+        }
+
+        cutoff = bandwidth * 2 * wlen / avctx->sample_rate;
+        pns_start_pos = NOISE_LOW_LIMIT * 2 * wlen / avctx->sample_rate;
+    }
+
+    /**
+     * for values above this the decoder might end up in an endless loop
+     * due to always having more bits than what can be encoded.
+     */
+    destbits = FFMIN(destbits, 5800);
+    toomanybits = FFMIN(toomanybits, 5800);
+    toofewbits = FFMIN(toofewbits, 5800);
+    /**
+     * XXX: some heuristic to determine initial quantizers will reduce search time
+     * determine zero bands and upper distortion limits
+     */
+    min_spread_thr_r = -1;
+    max_spread_thr_r = -1;
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        for (g = start = 0;  g < sce->ics.num_swb; start += sce->ics.swb_sizes[g++]) {
+            int nz = 0;
+            float uplim = 0.0f, energy = 0.0f, spread = 0.0f;
+            for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
+                if (start >= cutoff || band->energy <= (band->threshold * zeroscale) || band->threshold == 0.0f) {
+                    sce->zeroes[(w+w2)*16+g] = 1;
+                    continue;
+                }
+                nz = 1;
+            }
+            if (!nz) {
+                uplim = 0.0f;
+            } else {
+                nz = 0;
+                for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                    FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
+                    if (band->energy <= (band->threshold * zeroscale) || band->threshold == 0.0f)
+                        continue;
+                    uplim += band->threshold;
+                    energy += band->energy;
+                    spread += band->spread;
+                    nz++;
+                }
+            }
+            uplims[w*16+g] = uplim;
+            energies[w*16+g] = energy;
+            nzs[w*16+g] = nz;
+            sce->zeroes[w*16+g] = !nz;
+            allz |= nz;
+            if (nz && sce->can_pns[w*16+g]) {
+                spread_thr_r[w*16+g] = energy * nz / (uplim * spread);
+                if (min_spread_thr_r < 0) {
+                    min_spread_thr_r = max_spread_thr_r = spread_thr_r[w*16+g];
+                } else {
+                    min_spread_thr_r = FFMIN(min_spread_thr_r, spread_thr_r[w*16+g]);
+                    max_spread_thr_r = FFMAX(max_spread_thr_r, spread_thr_r[w*16+g]);
+                }
+            }
+        }
+    }
+
+    /** Compute initial scalers */
+    minscaler = 65535;
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        for (g = 0;  g < sce->ics.num_swb; g++) {
+            if (sce->zeroes[w*16+g]) {
+                sce->sf_idx[w*16+g] = SCALE_ONE_POS;
+                continue;
+            }
+            /**
+             * log2f-to-distortion ratio is, technically, 2 (1.5db = 4, but it's power vs level so it's 2).
+             * But, as offsets are applied, low-frequency signals are too sensitive to the induced distortion,
+             * so we make scaling more conservative by choosing a lower log2f-to-distortion ratio, and thus
+             * more robust.
+             */
+            sce->sf_idx[w*16+g] = av_clip(
+                SCALE_ONE_POS
+                    + 1.75*log2f(FFMAX(0.00125f,uplims[w*16+g]) / sce->ics.swb_sizes[g])
+                    + sfoffs,
+                60, SCALE_MAX_POS);
+            minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]);
+        }
+    }
+
+    /** Clip */
+    minscaler = av_clip(minscaler, SCALE_ONE_POS - SCALE_DIV_512, SCALE_MAX_POS - SCALE_DIV_512);
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
+        for (g = 0;  g < sce->ics.num_swb; g++)
+            if (!sce->zeroes[w*16+g])
+                sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler, minscaler + SCALE_MAX_DIFF - 1);
+
+    if (!allz)
+        return;
+    s->abs_pow34(s->scoefs, sce->coeffs, 1024);
+    ff_quantize_band_cost_cache_init(s);
+
+    for (i = 0; i < sizeof(minsf) / sizeof(minsf[0]); ++i)
+        minsf[i] = 0;
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        start = w*128;
+        for (g = 0;  g < sce->ics.num_swb; g++) {
+            const float *scaled = s->scoefs + start;
+            int minsfidx;
+            maxvals[w*16+g] = find_max_val(sce->ics.group_len[w], sce->ics.swb_sizes[g], scaled);
+            if (maxvals[w*16+g] > 0) {
+                minsfidx = coef2minsf(maxvals[w*16+g]);
+                for (w2 = 0; w2 < sce->ics.group_len[w]; w2++)
+                    minsf[(w+w2)*16+g] = minsfidx;
+            }
+            start += sce->ics.swb_sizes[g];
+        }
+    }
+
+    /**
+     * Scale uplims to match rate distortion to quality
+     * bu applying noisy band depriorization and tonal band priorization.
+     * Maxval-energy ratio gives us an idea of how noisy/tonal the band is.
+     * If maxval^2 ~ energy, then that band is mostly noise, and we can relax
+     * rate distortion requirements.
+     */
+    memcpy(euplims, uplims, sizeof(euplims));
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        /** psy already priorizes transients to some extent */
+        float de_psy_factor = (sce->ics.num_windows > 1) ? 8.0f / sce->ics.group_len[w] : 1.0f;
+        start = w*128;
+        for (g = 0;  g < sce->ics.num_swb; g++) {
+            if (nzs[g] > 0) {
+                float cleanup_factor = ff_sqrf(av_clipf(start / (cutoff * 0.75f), 1.0f, 2.0f));
+                float energy2uplim = find_form_factor(
+                    sce->ics.group_len[w], sce->ics.swb_sizes[g],
+                    uplims[w*16+g] / (nzs[g] * sce->ics.swb_sizes[w]),
+                    sce->coeffs + start,
+                    nzslope * cleanup_factor);
+                energy2uplim *= de_psy_factor;
+                if (!(avctx->flags & AV_CODEC_FLAG_QSCALE)) {
+                    /** In ABR, we need to priorize less and let rate control do its thing */
+                    energy2uplim = sqrtf(energy2uplim);
+                }
+                energy2uplim = FFMAX(0.015625f, FFMIN(1.0f, energy2uplim));
+                uplims[w*16+g] *= av_clipf(rdlambda * energy2uplim, rdmin, rdmax)
+                                  * sce->ics.group_len[w];
+
+                energy2uplim = find_form_factor(
+                    sce->ics.group_len[w], sce->ics.swb_sizes[g],
+                    uplims[w*16+g] / (nzs[g] * sce->ics.swb_sizes[w]),
+                    sce->coeffs + start,
+                    2.0f);
+                energy2uplim *= de_psy_factor;
+                if (!(avctx->flags & AV_CODEC_FLAG_QSCALE)) {
+                    /** In ABR, we need to priorize less and let rate control do its thing */
+                    energy2uplim = sqrtf(energy2uplim);
+                }
+                energy2uplim = FFMAX(0.015625f, FFMIN(1.0f, energy2uplim));
+                euplims[w*16+g] *= av_clipf(rdlambda * energy2uplim * sce->ics.group_len[w],
+                    0.5f, 1.0f);
+            }
+            start += sce->ics.swb_sizes[g];
+        }
+    }
+
+    for (i = 0; i < sizeof(maxsf) / sizeof(maxsf[0]); ++i)
+        maxsf[i] = SCALE_MAX_POS;
+
+    //perform two-loop search
+    //outer loop - improve quality
+    do {
+        //inner loop - quantize spectrum to fit into given number of bits
+        int overdist;
+        int qstep = its ? 1 : 32;
+        do {
+            int changed = 0;
+            prev = -1;
+            recomprd = 0;
+            tbits = 0;
+            for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+                start = w*128;
+                for (g = 0;  g < sce->ics.num_swb; g++) {
+                    const float *coefs = &sce->coeffs[start];
+                    const float *scaled = &s->scoefs[start];
+                    int bits = 0;
+                    int cb;
+                    float dist = 0.0f;
+                    float qenergy = 0.0f;
+
+                    if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
+                        start += sce->ics.swb_sizes[g];
+                        if (sce->can_pns[w*16+g]) {
+                            /** PNS isn't free */
+                            tbits += ff_pns_bits(sce, w, g);
+                        }
+                        continue;
+                    }
+                    cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
+                    for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                        int b;
+                        float sqenergy;
+                        dist += quantize_band_cost_cached(s, w + w2, g, coefs + w2*128,
+                                                   scaled + w2*128,
+                                                   sce->ics.swb_sizes[g],
+                                                   sce->sf_idx[w*16+g],
+                                                   cb,
+                                                   1.0f,
+                                                   INFINITY,
+                                                   &b, &sqenergy,
+                                                   0);
+                        bits += b;
+                        qenergy += sqenergy;
+                    }
+                    dists[w*16+g] = dist - bits;
+                    qenergies[w*16+g] = qenergy;
+                    if (prev != -1) {
+                        int sfdiff = av_clip(sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO, 0, 2*SCALE_MAX_DIFF);
+                        bits += ff_aac_scalefactor_bits[sfdiff];
+                    }
+                    tbits += bits;
+                    start += sce->ics.swb_sizes[g];
+                    prev = sce->sf_idx[w*16+g];
+                }
+            }
+            if (tbits > toomanybits) {
+                recomprd = 1;
+                for (i = 0; i < 128; i++) {
+                    if (sce->sf_idx[i] < (SCALE_MAX_POS - SCALE_DIV_512)) {
+                        int maxsf_i = (tbits > 5800) ? SCALE_MAX_POS : maxsf[i];
+                        int new_sf = FFMIN(maxsf_i, sce->sf_idx[i] + qstep);
+                        if (new_sf != sce->sf_idx[i]) {
+                            sce->sf_idx[i] = new_sf;
+                            changed = 1;
+                        }
+                    }
+                }
+            } else if (tbits < toofewbits) {
+                recomprd = 1;
+                for (i = 0; i < 128; i++) {
+                    if (sce->sf_idx[i] > SCALE_ONE_POS) {
+                        int new_sf = FFMAX3(minsf[i], SCALE_ONE_POS, sce->sf_idx[i] - qstep);
+                        if (new_sf != sce->sf_idx[i]) {
+                            sce->sf_idx[i] = new_sf;
+                            changed = 1;
+                        }
+                    }
+                }
+            }
+            qstep >>= 1;
+            if (!qstep && tbits > toomanybits && sce->sf_idx[0] < 217 && changed)
+                qstep = 1;
+        } while (qstep);
+
+        overdist = 1;
+        fflag = tbits < toofewbits;
+        for (i = 0; i < 2 && (overdist || recomprd); ++i) {
+            if (recomprd) {
+                /** Must recompute distortion */
+                prev = -1;
+                tbits = 0;
+                for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+                    start = w*128;
+                    for (g = 0;  g < sce->ics.num_swb; g++) {
+                        const float *coefs = sce->coeffs + start;
+                        const float *scaled = s->scoefs + start;
+                        int bits = 0;
+                        int cb;
+                        float dist = 0.0f;
+                        float qenergy = 0.0f;
+
+                        if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
+                            start += sce->ics.swb_sizes[g];
+                            if (sce->can_pns[w*16+g]) {
+                                /** PNS isn't free */
+                                tbits += ff_pns_bits(sce, w, g);
+                            }
+                            continue;
+                        }
+                        cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
+                        for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                            int b;
+                            float sqenergy;
+                            dist += quantize_band_cost_cached(s, w + w2, g, coefs + w2*128,
+                                                    scaled + w2*128,
+                                                    sce->ics.swb_sizes[g],
+                                                    sce->sf_idx[w*16+g],
+                                                    cb,
+                                                    1.0f,
+                                                    INFINITY,
+                                                    &b, &sqenergy,
+                                                    0);
+                            bits += b;
+                            qenergy += sqenergy;
+                        }
+                        dists[w*16+g] = dist - bits;
+                        qenergies[w*16+g] = qenergy;
+                        if (prev != -1) {
+                            int sfdiff = av_clip(sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO, 0, 2*SCALE_MAX_DIFF);
+                            bits += ff_aac_scalefactor_bits[sfdiff];
+                        }
+                        tbits += bits;
+                        start += sce->ics.swb_sizes[g];
+                        prev = sce->sf_idx[w*16+g];
+                    }
+                }
+            }
+            if (!i && s->options.pns && its > maxits/2 && tbits > toofewbits) {
+                float maxoverdist = 0.0f;
+                float ovrfactor = 1.f+(maxits-its)*16.f/maxits;
+                overdist = recomprd = 0;
+                for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+                    for (g = start = 0;  g < sce->ics.num_swb; start += sce->ics.swb_sizes[g++]) {
+                        if (!sce->zeroes[w*16+g] && sce->sf_idx[w*16+g] > SCALE_ONE_POS && dists[w*16+g] > uplims[w*16+g]*ovrfactor) {
+                            float ovrdist = dists[w*16+g] / FFMAX(uplims[w*16+g],euplims[w*16+g]);
+                            maxoverdist = FFMAX(maxoverdist, ovrdist);
+                            overdist++;
+                        }
+                    }
+                }
+                if (overdist) {
+                    /* We have overdistorted bands, trade for zeroes (that can be noise)
+                     * Zero the bands in the lowest 1.25% spread-energy-threshold ranking
+                     */
+                    float minspread = max_spread_thr_r;
+                    float maxspread = min_spread_thr_r;
+                    float zspread;
+                    int zeroable = 0;
+                    int zeroed = 0;
+                    int maxzeroed, zloop;
+                    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+                        for (g = start = 0;  g < sce->ics.num_swb; start += sce->ics.swb_sizes[g++]) {
+                            if (start >= pns_start_pos && !sce->zeroes[w*16+g] && sce->can_pns[w*16+g]) {
+                                minspread = FFMIN(minspread, spread_thr_r[w*16+g]);
+                                maxspread = FFMAX(maxspread, spread_thr_r[w*16+g]);
+                                zeroable++;
+                            }
+                        }
+                    }
+                    zspread = (maxspread-minspread) * 0.0125f + minspread;
+                    /* Don't PNS everything even if allowed. It suppresses bit starvation signals from RC,
+                     * and forced the hand of the later search_for_pns step.
+                     * Instead, PNS a fraction of the spread_thr_r range depending on how starved for bits we are,
+                     * and leave further PNSing to search_for_pns if worthwhile.
+                     */
+                    zspread = FFMIN3(min_spread_thr_r * 8.f, zspread,
+                        ((toomanybits - tbits) * min_spread_thr_r + (tbits - toofewbits) * max_spread_thr_r) / (toomanybits - toofewbits + 1));
+                    maxzeroed = FFMIN(zeroable, FFMAX(1, (zeroable * its + maxits - 1) / (2 * maxits)));
+                    for (zloop = 0; zloop < 2; zloop++) {
+                        /* Two passes: first distorted stuff - two birds in one shot and all that,
+                         * then anything viable. Viable means not zero, but either CB=zero-able
+                         * (too high SF), not SF <= 1 (that means we'd be operating at very high
+                         * quality, we don't want PNS when doing VHQ), PNS allowed, and within
+                         * the lowest ranking percentile.
+                         */
+                        float loopovrfactor = (zloop) ? 1.0f : ovrfactor;
+                        int loopminsf = (zloop) ? (SCALE_ONE_POS - SCALE_DIV_512) : SCALE_ONE_POS;
+                        int mcb;
+                        for (g = sce->ics.num_swb-1; g > 0 && zeroed < maxzeroed; g--) {
+                            if (sce->ics.swb_offset[g] < pns_start_pos)
+                                continue;
+                            for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+                                if (!sce->zeroes[w*16+g] && sce->can_pns[w*16+g] && spread_thr_r[w*16+g] <= zspread
+                                    && sce->sf_idx[w*16+g] > loopminsf
+                                    && (dists[w*16+g] > loopovrfactor*uplims[w*16+g] || !(mcb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]))
+                                        || (mcb <= 1 && dists[w*16+g] > FFMIN(uplims[w*16+g], euplims[w*16+g]))) ) {
+                                    sce->zeroes[w*16+g] = 1;
+                                    sce->band_type[w*16+g] = 0;
+                                    zeroed++;
+                                }
+                            }
+                        }
+                    }
+                    if (zeroed)
+                        recomprd = fflag = 1;
+                } else {
+                    overdist = 0;
+                }
+            }
+        }
+
+        minscaler = SCALE_MAX_POS;
+        maxscaler = 0;
+        for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+            for (g = 0;  g < sce->ics.num_swb; g++) {
+                if (!sce->zeroes[w*16+g]) {
+                    minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]);
+                    maxscaler = FFMAX(maxscaler, sce->sf_idx[w*16+g]);
+                }
+            }
+        }
+
+        minscaler = nminscaler = av_clip(minscaler, SCALE_ONE_POS - SCALE_DIV_512, SCALE_MAX_POS - SCALE_DIV_512);
+        prev = -1;
+        for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+            /** Start with big steps, end up fine-tunning */
+            int depth = (its > maxits/2) ? ((its > maxits*2/3) ? 1 : 3) : 10;
+            int edepth = depth+2;
+            float uplmax = its / (maxits*0.25f) + 1.0f;
+            uplmax *= (tbits > destbits) ? FFMIN(2.0f, tbits / (float)FFMAX(1,destbits)) : 1.0f;
+            start = w * 128;
+            for (g = 0; g < sce->ics.num_swb; g++) {
+                int prevsc = sce->sf_idx[w*16+g];
+                if (prev < 0 && !sce->zeroes[w*16+g])
+                    prev = sce->sf_idx[0];
+                if (!sce->zeroes[w*16+g]) {
+                    const float *coefs = sce->coeffs + start;
+                    const float *scaled = s->scoefs + start;
+                    int cmb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
+                    int mindeltasf = FFMAX(0, prev - SCALE_MAX_DIFF);
+                    int maxdeltasf = FFMIN(SCALE_MAX_POS - SCALE_DIV_512, prev + SCALE_MAX_DIFF);
+                    if ((!cmb || dists[w*16+g] > uplims[w*16+g]) && sce->sf_idx[w*16+g] > FFMAX(mindeltasf, minsf[w*16+g])) {
+                        /* Try to make sure there is some energy in every nonzero band
+                         * NOTE: This algorithm must be forcibly imbalanced, pushing harder
+                         *  on holes or more distorted bands at first, otherwise there's
+                         *  no net gain (since the next iteration will offset all bands
+                         *  on the opposite direction to compensate for extra bits)
+                         */
+                        for (i = 0; i < edepth && sce->sf_idx[w*16+g] > mindeltasf; ++i) {
+                            int cb, bits;
+                            float dist, qenergy;
+                            int mb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]-1);
+                            cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
+                            dist = qenergy = 0.f;
+                            bits = 0;
+                            if (!cb) {
+                                maxsf[w*16+g] = FFMIN(sce->sf_idx[w*16+g]-1, maxsf[w*16+g]);
+                            } else if (i >= depth && dists[w*16+g] < euplims[w*16+g]) {
+                                break;
+                            }
+                            /* !g is the DC band, it's important, since quantization error here
+                             * applies to less than a cycle, it creates horrible intermodulation
+                             * distortion if it doesn't stick to what psy requests
+                             */
+                            if (!g && sce->ics.num_windows > 1 && dists[w*16+g] >= euplims[w*16+g])
+                                maxsf[w*16+g] = FFMIN(sce->sf_idx[w*16+g], maxsf[w*16+g]);
+                            for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                                int b;
+                                float sqenergy;
+                                dist += quantize_band_cost_cached(s, w + w2, g, coefs + w2*128,
+                                                        scaled + w2*128,
+                                                        sce->ics.swb_sizes[g],
+                                                        sce->sf_idx[w*16+g]-1,
+                                                        cb,
+                                                        1.0f,
+                                                        INFINITY,
+                                                        &b, &sqenergy,
+                                                        0);
+                                bits += b;
+                                qenergy += sqenergy;
+                            }
+                            sce->sf_idx[w*16+g]--;
+                            dists[w*16+g] = dist - bits;
+                            qenergies[w*16+g] = qenergy;
+                            if (mb && (sce->sf_idx[w*16+g] < mindeltasf || (
+                                    (dists[w*16+g] < FFMIN(uplmax*uplims[w*16+g], euplims[w*16+g]))
+                                    && (fabsf(qenergies[w*16+g]-energies[w*16+g]) < euplims[w*16+g])
+                                ) )) {
+                                break;
+                            }
+                        }
+                    } else if (tbits > toofewbits && sce->sf_idx[w*16+g] < FFMIN(maxdeltasf, maxsf[w*16+g])
+                            && (dists[w*16+g] < FFMIN(euplims[w*16+g], uplims[w*16+g]))
+                            && (fabsf(qenergies[w*16+g]-energies[w*16+g]) < euplims[w*16+g])
+                        ) {
+                        /** Um... over target. Save bits for more important stuff. */
+                        for (i = 0; i < depth && sce->sf_idx[w*16+g] < maxdeltasf; ++i) {
+                            int cb, bits;
+                            float dist, qenergy;
+                            cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]+1);
+                            if (cb > 0) {
+                                dist = qenergy = 0.f;
+                                bits = 0;
+                                for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                                    int b;
+                                    float sqenergy;
+                                    dist += quantize_band_cost_cached(s, w + w2, g, coefs + w2*128,
+                                                            scaled + w2*128,
+                                                            sce->ics.swb_sizes[g],
+                                                            sce->sf_idx[w*16+g]+1,
+                                                            cb,
+                                                            1.0f,
+                                                            INFINITY,
+                                                            &b, &sqenergy,
+                                                            0);
+                                    bits += b;
+                                    qenergy += sqenergy;
+                                }
+                                dist -= bits;
+                                if (dist < FFMIN(euplims[w*16+g], uplims[w*16+g])) {
+                                    sce->sf_idx[w*16+g]++;
+                                    dists[w*16+g] = dist;
+                                    qenergies[w*16+g] = qenergy;
+                                } else {
+                                    break;
+                                }
+                            } else {
+                                maxsf[w*16+g] = FFMIN(sce->sf_idx[w*16+g], maxsf[w*16+g]);
+                                break;
+                            }
+                        }
+                    }
+                    prev = sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], mindeltasf, maxdeltasf);
+                    if (sce->sf_idx[w*16+g] != prevsc)
+                        fflag = 1;
+                    nminscaler = FFMIN(nminscaler, sce->sf_idx[w*16+g]);
+                    sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
+                }
+                start += sce->ics.swb_sizes[g];
+            }
+        }
+
+        /** SF difference limit violation risk. Must re-clamp. */
+        prev = -1;
+        for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+            for (g = 0; g < sce->ics.num_swb; g++) {
+                if (!sce->zeroes[w*16+g]) {
+                    int prevsf = sce->sf_idx[w*16+g];
+                    if (prev < 0)
+                        prev = prevsf;
+                    sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], prev - SCALE_MAX_DIFF, prev + SCALE_MAX_DIFF);
+                    sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
+                    prev = sce->sf_idx[w*16+g];
+                    if (!fflag && prevsf != sce->sf_idx[w*16+g])
+                        fflag = 1;
+                }
+            }
+        }
+
+        its++;
+    } while (fflag && its < maxits);
+
+    /** Scout out next nonzero bands */
+    ff_init_nextband_map(sce, nextband);
+
+    prev = -1;
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        /** Make sure proper codebooks are set */
+        for (g = 0; g < sce->ics.num_swb; g++) {
+            if (!sce->zeroes[w*16+g]) {
+                sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
+                if (sce->band_type[w*16+g] <= 0) {
+                    if (!ff_sfdelta_can_remove_band(sce, nextband, prev, w*16+g)) {
+                        /** Cannot zero out, make sure it's not attempted */
+                        sce->band_type[w*16+g] = 1;
+                    } else {
+                        sce->zeroes[w*16+g] = 1;
+                        sce->band_type[w*16+g] = 0;
+                    }
+                }
+            } else {
+                sce->band_type[w*16+g] = 0;
+            }
+            /** Check that there's no SF delta range violations */
+            if (!sce->zeroes[w*16+g]) {
+                if (prev != -1) {
+                    av_unused int sfdiff = sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO;
+                    av_assert1(sfdiff >= 0 && sfdiff <= 2*SCALE_MAX_DIFF);
+                } else if (sce->zeroes[0]) {
+                    /** Set global gain to something useful */
+                    sce->sf_idx[0] = sce->sf_idx[w*16+g];
+                }
+                prev = sce->sf_idx[w*16+g];
+            }
+        }
+    }
+}
+
+#endif /* AVCODEC_AACCODER_TWOLOOP_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacdec.c

@@ -0,0 +1,595 @@
+/*
+ * AAC decoder
+ * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
+ * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
+ *
+ * AAC LATM decoder
+ * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
+ * Copyright (c) 2010      Janne Grunau <janne-libav@jannau.net>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC decoder
+ * @author Oded Shimon  ( ods15 ods15 dyndns org )
+ * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
+ */
+
+#define FFT_FLOAT 1
+#define FFT_FIXED_32 0
+#define USE_FIXED 0
+
+#include "libavutil/float_dsp.h"
+#include "libavutil/opt.h"
+#include "avcodec.h"
+#include "internal.h"
+#include "get_bits.h"
+#include "fft.h"
+#include "mdct15.h"
+#include "lpc.h"
+#include "kbdwin.h"
+#include "sinewin.h"
+
+#include "aac.h"
+#include "aactab.h"
+#include "aacdectab.h"
+#include "adts_header.h"
+#include "cbrt_data.h"
+#include "sbr.h"
+#include "aacsbr.h"
+#include "mpeg4audio.h"
+#include "profiles.h"
+#include "libavutil/intfloat.h"
+
+#include <errno.h>
+#include <math.h>
+#include <stdint.h>
+#include <string.h>
+
+#if ARCH_ARM
+#   include "arm/aac.h"
+#elif ARCH_MIPS
+#   include "mips/aacdec_mips.h"
+#endif
+
+static av_always_inline void reset_predict_state(PredictorState *ps)
+{
+    ps->r0   = 0.0f;
+    ps->r1   = 0.0f;
+    ps->cor0 = 0.0f;
+    ps->cor1 = 0.0f;
+    ps->var0 = 1.0f;
+    ps->var1 = 1.0f;
+}
+
+#ifndef VMUL2
+static inline float *VMUL2(float *dst, const float *v, unsigned idx,
+                           const float *scale)
+{
+    float s = *scale;
+    *dst++ = v[idx    & 15] * s;
+    *dst++ = v[idx>>4 & 15] * s;
+    return dst;
+}
+#endif
+
+#ifndef VMUL4
+static inline float *VMUL4(float *dst, const float *v, unsigned idx,
+                           const float *scale)
+{
+    float s = *scale;
+    *dst++ = v[idx    & 3] * s;
+    *dst++ = v[idx>>2 & 3] * s;
+    *dst++ = v[idx>>4 & 3] * s;
+    *dst++ = v[idx>>6 & 3] * s;
+    return dst;
+}
+#endif
+
+#ifndef VMUL2S
+static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
+                            unsigned sign, const float *scale)
+{
+    union av_intfloat32 s0, s1;
+
+    s0.f = s1.f = *scale;
+    s0.i ^= sign >> 1 << 31;
+    s1.i ^= sign      << 31;
+
+    *dst++ = v[idx    & 15] * s0.f;
+    *dst++ = v[idx>>4 & 15] * s1.f;
+
+    return dst;
+}
+#endif
+
+#ifndef VMUL4S
+static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
+                            unsigned sign, const float *scale)
+{
+    unsigned nz = idx >> 12;
+    union av_intfloat32 s = { .f = *scale };
+    union av_intfloat32 t;
+
+    t.i = s.i ^ (sign & 1U<<31);
+    *dst++ = v[idx    & 3] * t.f;
+
+    sign <<= nz & 1; nz >>= 1;
+    t.i = s.i ^ (sign & 1U<<31);
+    *dst++ = v[idx>>2 & 3] * t.f;
+
+    sign <<= nz & 1; nz >>= 1;
+    t.i = s.i ^ (sign & 1U<<31);
+    *dst++ = v[idx>>4 & 3] * t.f;
+
+    sign <<= nz & 1;
+    t.i = s.i ^ (sign & 1U<<31);
+    *dst++ = v[idx>>6 & 3] * t.f;
+
+    return dst;
+}
+#endif
+
+static av_always_inline float flt16_round(float pf)
+{
+    union av_intfloat32 tmp;
+    tmp.f = pf;
+    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
+    return tmp.f;
+}
+
+static av_always_inline float flt16_even(float pf)
+{
+    union av_intfloat32 tmp;
+    tmp.f = pf;
+    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
+    return tmp.f;
+}
+
+static av_always_inline float flt16_trunc(float pf)
+{
+    union av_intfloat32 pun;
+    pun.f = pf;
+    pun.i &= 0xFFFF0000U;
+    return pun.f;
+}
+
+static av_always_inline void predict(PredictorState *ps, float *coef,
+                                     int output_enable)
+{
+    const float a     = 0.953125; // 61.0 / 64
+    const float alpha = 0.90625;  // 29.0 / 32
+    float e0, e1;
+    float pv;
+    float k1, k2;
+    float   r0 = ps->r0,     r1 = ps->r1;
+    float cor0 = ps->cor0, cor1 = ps->cor1;
+    float var0 = ps->var0, var1 = ps->var1;
+
+    k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
+    k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
+
+    pv = flt16_round(k1 * r0 + k2 * r1);
+    if (output_enable)
+        *coef += pv;
+
+    e0 = *coef;
+    e1 = e0 - k1 * r0;
+
+    ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
+    ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
+    ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
+    ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
+
+    ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
+    ps->r0 = flt16_trunc(a * e0);
+}
+
+/**
+ * Apply dependent channel coupling (applied before IMDCT).
+ *
+ * @param   index   index into coupling gain array
+ */
+static void apply_dependent_coupling(AACContext *ac,
+                                     SingleChannelElement *target,
+                                     ChannelElement *cce, int index)
+{
+    IndividualChannelStream *ics = &cce->ch[0].ics;
+    const uint16_t *offsets = ics->swb_offset;
+    float *dest = target->coeffs;
+    const float *src = cce->ch[0].coeffs;
+    int g, i, group, k, idx = 0;
+    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
+        av_log(ac->avctx, AV_LOG_ERROR,
+               "Dependent coupling is not supported together with LTP\n");
+        return;
+    }
+    for (g = 0; g < ics->num_window_groups; g++) {
+        for (i = 0; i < ics->max_sfb; i++, idx++) {
+            if (cce->ch[0].band_type[idx] != ZERO_BT) {
+                const float gain = cce->coup.gain[index][idx];
+                for (group = 0; group < ics->group_len[g]; group++) {
+                    for (k = offsets[i]; k < offsets[i + 1]; k++) {
+                        // FIXME: SIMDify
+                        dest[group * 128 + k] += gain * src[group * 128 + k];
+                    }
+                }
+            }
+        }
+        dest += ics->group_len[g] * 128;
+        src  += ics->group_len[g] * 128;
+    }
+}
+
+/**
+ * Apply independent channel coupling (applied after IMDCT).
+ *
+ * @param   index   index into coupling gain array
+ */
+static void apply_independent_coupling(AACContext *ac,
+                                       SingleChannelElement *target,
+                                       ChannelElement *cce, int index)
+{
+    int i;
+    const float gain = cce->coup.gain[index][0];
+    const float *src = cce->ch[0].ret;
+    float *dest = target->ret;
+    const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
+
+    for (i = 0; i < len; i++)
+        dest[i] += gain * src[i];
+}
+
+#include "aacdec_template.c"
+
+#define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
+
+struct LATMContext {
+    AACContext aac_ctx;     ///< containing AACContext
+    int initialized;        ///< initialized after a valid extradata was seen
+
+    // parser data
+    int audio_mux_version_A; ///< LATM syntax version
+    int frame_length_type;   ///< 0/1 variable/fixed frame length
+    int frame_length;        ///< frame length for fixed frame length
+};
+
+static inline uint32_t latm_get_value(GetBitContext *b)
+{
+    int length = get_bits(b, 2);
+
+    return get_bits_long(b, (length+1)*8);
+}
+
+static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
+                                             GetBitContext *gb, int asclen)
+{
+    AACContext *ac        = &latmctx->aac_ctx;
+    AVCodecContext *avctx = ac->avctx;
+    MPEG4AudioConfig m4ac = { 0 };
+    GetBitContext gbc;
+    int config_start_bit  = get_bits_count(gb);
+    int sync_extension    = 0;
+    int bits_consumed, esize, i;
+
+    if (asclen > 0) {
+        sync_extension = 1;
+        asclen         = FFMIN(asclen, get_bits_left(gb));
+        init_get_bits(&gbc, gb->buffer, config_start_bit + asclen);
+        skip_bits_long(&gbc, config_start_bit);
+    } else if (asclen == 0) {
+        gbc = *gb;
+    } else {
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (get_bits_left(gb) <= 0)
+        return AVERROR_INVALIDDATA;
+
+    bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac,
+                                                    &gbc, config_start_bit,
+                                                    sync_extension);
+
+    if (bits_consumed < config_start_bit)
+        return AVERROR_INVALIDDATA;
+    bits_consumed -= config_start_bit;
+
+    if (asclen == 0)
+      asclen = bits_consumed;
+
+    if (!latmctx->initialized ||
+        ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
+        ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
+
+        if (latmctx->initialized) {
+            av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config);
+        } else {
+            av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
+        }
+        latmctx->initialized = 0;
+
+        esize = (asclen + 7) / 8;
+
+        if (avctx->extradata_size < esize) {
+            av_free(avctx->extradata);
+            avctx->extradata = av_malloc(esize + AV_INPUT_BUFFER_PADDING_SIZE);
+            if (!avctx->extradata)
+                return AVERROR(ENOMEM);
+        }
+
+        avctx->extradata_size = esize;
+        gbc = *gb;
+        for (i = 0; i < esize; i++) {
+          avctx->extradata[i] = get_bits(&gbc, 8);
+        }
+        memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE);
+    }
+    skip_bits_long(gb, asclen);
+
+    return 0;
+}
+
+static int read_stream_mux_config(struct LATMContext *latmctx,
+                                  GetBitContext *gb)
+{
+    int ret, audio_mux_version = get_bits(gb, 1);
+
+    latmctx->audio_mux_version_A = 0;
+    if (audio_mux_version)
+        latmctx->audio_mux_version_A = get_bits(gb, 1);
+
+    if (!latmctx->audio_mux_version_A) {
+
+        if (audio_mux_version)
+            latm_get_value(gb);                 // taraFullness
+
+        skip_bits(gb, 1);                       // allStreamSameTimeFraming
+        skip_bits(gb, 6);                       // numSubFrames
+        // numPrograms
+        if (get_bits(gb, 4)) {                  // numPrograms
+            avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
+            return AVERROR_PATCHWELCOME;
+        }
+
+        // for each program (which there is only one in DVB)
+
+        // for each layer (which there is only one in DVB)
+        if (get_bits(gb, 3)) {                   // numLayer
+            avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
+            return AVERROR_PATCHWELCOME;
+        }
+
+        // for all but first stream: use_same_config = get_bits(gb, 1);
+        if (!audio_mux_version) {
+            if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
+                return ret;
+        } else {
+            int ascLen = latm_get_value(gb);
+            if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
+                return ret;
+        }
+
+        latmctx->frame_length_type = get_bits(gb, 3);
+        switch (latmctx->frame_length_type) {
+        case 0:
+            skip_bits(gb, 8);       // latmBufferFullness
+            break;
+        case 1:
+            latmctx->frame_length = get_bits(gb, 9);
+            break;
+        case 3:
+        case 4:
+        case 5:
+            skip_bits(gb, 6);       // CELP frame length table index
+            break;
+        case 6:
+        case 7:
+            skip_bits(gb, 1);       // HVXC frame length table index
+            break;
+        }
+
+        if (get_bits(gb, 1)) {                  // other data
+            if (audio_mux_version) {
+                latm_get_value(gb);             // other_data_bits
+            } else {
+                int esc;
+                do {
+                    if (get_bits_left(gb) < 9)
+                        return AVERROR_INVALIDDATA;
+                    esc = get_bits(gb, 1);
+                    skip_bits(gb, 8);
+                } while (esc);
+            }
+        }
+
+        if (get_bits(gb, 1))                     // crc present
+            skip_bits(gb, 8);                    // config_crc
+    }
+
+    return 0;
+}
+
+static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
+{
+    uint8_t tmp;
+
+    if (ctx->frame_length_type == 0) {
+        int mux_slot_length = 0;
+        do {
+            if (get_bits_left(gb) < 8)
+                return AVERROR_INVALIDDATA;
+            tmp = get_bits(gb, 8);
+            mux_slot_length += tmp;
+        } while (tmp == 255);
+        return mux_slot_length;
+    } else if (ctx->frame_length_type == 1) {
+        return ctx->frame_length;
+    } else if (ctx->frame_length_type == 3 ||
+               ctx->frame_length_type == 5 ||
+               ctx->frame_length_type == 7) {
+        skip_bits(gb, 2);          // mux_slot_length_coded
+    }
+    return 0;
+}
+
+static int read_audio_mux_element(struct LATMContext *latmctx,
+                                  GetBitContext *gb)
+{
+    int err;
+    uint8_t use_same_mux = get_bits(gb, 1);
+    if (!use_same_mux) {
+        if ((err = read_stream_mux_config(latmctx, gb)) < 0)
+            return err;
+    } else if (!latmctx->aac_ctx.avctx->extradata) {
+        av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
+               "no decoder config found\n");
+        return 1;
+    }
+    if (latmctx->audio_mux_version_A == 0) {
+        int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
+        if (mux_slot_length_bytes < 0 || mux_slot_length_bytes * 8LL > get_bits_left(gb)) {
+            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
+            return AVERROR_INVALIDDATA;
+        } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
+            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
+                   "frame length mismatch %d << %d\n",
+                   mux_slot_length_bytes * 8, get_bits_left(gb));
+            return AVERROR_INVALIDDATA;
+        }
+    }
+    return 0;
+}
+
+
+static int latm_decode_frame(AVCodecContext *avctx, void *out,
+                             int *got_frame_ptr, AVPacket *avpkt)
+{
+    struct LATMContext *latmctx = avctx->priv_data;
+    int                 muxlength, err;
+    GetBitContext       gb;
+
+    if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
+        return err;
+
+    // check for LOAS sync word
+    if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
+        return AVERROR_INVALIDDATA;
+
+    muxlength = get_bits(&gb, 13) + 3;
+    // not enough data, the parser should have sorted this out
+    if (muxlength > avpkt->size)
+        return AVERROR_INVALIDDATA;
+
+    if ((err = read_audio_mux_element(latmctx, &gb)))
+        return (err < 0) ? err : avpkt->size;
+
+    if (!latmctx->initialized) {
+        if (!avctx->extradata) {
+            *got_frame_ptr = 0;
+            return avpkt->size;
+        } else {
+            push_output_configuration(&latmctx->aac_ctx);
+            if ((err = decode_audio_specific_config(
+                    &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
+                    avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) {
+                pop_output_configuration(&latmctx->aac_ctx);
+                return err;
+            }
+            latmctx->initialized = 1;
+        }
+    }
+
+    if (show_bits(&gb, 12) == 0xfff) {
+        av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
+               "ADTS header detected, probably as result of configuration "
+               "misparsing\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    switch (latmctx->aac_ctx.oc[1].m4ac.object_type) {
+    case AOT_ER_AAC_LC:
+    case AOT_ER_AAC_LTP:
+    case AOT_ER_AAC_LD:
+    case AOT_ER_AAC_ELD:
+        err = aac_decode_er_frame(avctx, out, got_frame_ptr, &gb);
+        break;
+    default:
+        err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt);
+    }
+    if (err < 0)
+        return err;
+
+    return muxlength;
+}
+
+static av_cold int latm_decode_init(AVCodecContext *avctx)
+{
+    struct LATMContext *latmctx = avctx->priv_data;
+    int ret = aac_decode_init(avctx);
+
+    if (avctx->extradata_size > 0)
+        latmctx->initialized = !ret;
+
+    return ret;
+}
+
+AVCodec ff_aac_decoder = {
+    .name            = "aac",
+    .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
+    .type            = AVMEDIA_TYPE_AUDIO,
+    .id              = AV_CODEC_ID_AAC,
+    .priv_data_size  = sizeof(AACContext),
+    .init            = aac_decode_init,
+    .close           = aac_decode_close,
+    .decode          = aac_decode_frame,
+    .sample_fmts     = (const enum AVSampleFormat[]) {
+        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
+    },
+    .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
+    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
+    .channel_layouts = aac_channel_layout,
+    .flush = flush,
+    .priv_class      = &aac_decoder_class,
+    .profiles        = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
+};
+
+/*
+    Note: This decoder filter is intended to decode LATM streams transferred
+    in MPEG transport streams which only contain one program.
+    To do a more complex LATM demuxing a separate LATM demuxer should be used.
+*/
+AVCodec ff_aac_latm_decoder = {
+    .name            = "aac_latm",
+    .long_name       = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
+    .type            = AVMEDIA_TYPE_AUDIO,
+    .id              = AV_CODEC_ID_AAC_LATM,
+    .priv_data_size  = sizeof(struct LATMContext),
+    .init            = latm_decode_init,
+    .close           = aac_decode_close,
+    .decode          = latm_decode_frame,
+    .sample_fmts     = (const enum AVSampleFormat[]) {
+        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
+    },
+    .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
+    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
+    .channel_layouts = aac_channel_layout,
+    .flush = flush,
+    .profiles        = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
+};

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacdec_fixed.c

@@ -0,0 +1,468 @@
+/*
+ * Copyright (c) 2013
+ *      MIPS Technologies, Inc., California.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
+ *    contributors may be used to endorse or promote products derived from
+ *    this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED.  IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
+ * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ * AAC decoder fixed-point implementation
+ *
+ * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
+ * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC decoder
+ * @author Oded Shimon  ( ods15 ods15 dyndns org )
+ * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * Fixed point implementation
+ * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
+ */
+
+#define FFT_FLOAT 0
+#define FFT_FIXED_32 1
+#define USE_FIXED 1
+
+#include "libavutil/fixed_dsp.h"
+#include "libavutil/opt.h"
+#include "avcodec.h"
+#include "internal.h"
+#include "get_bits.h"
+#include "fft.h"
+#include "lpc.h"
+#include "kbdwin.h"
+#include "sinewin.h"
+
+#include "aac.h"
+#include "aactab.h"
+#include "aacdectab.h"
+#include "adts_header.h"
+#include "cbrt_data.h"
+#include "sbr.h"
+#include "aacsbr.h"
+#include "mpeg4audio.h"
+#include "profiles.h"
+#include "libavutil/intfloat.h"
+
+#include <math.h>
+#include <string.h>
+
+static av_always_inline void reset_predict_state(PredictorState *ps)
+{
+    ps->r0.mant   = 0;
+    ps->r0.exp   = 0;
+    ps->r1.mant   = 0;
+    ps->r1.exp   = 0;
+    ps->cor0.mant = 0;
+    ps->cor0.exp = 0;
+    ps->cor1.mant = 0;
+    ps->cor1.exp = 0;
+    ps->var0.mant = 0x20000000;
+    ps->var0.exp = 1;
+    ps->var1.mant = 0x20000000;
+    ps->var1.exp = 1;
+}
+
+static const int exp2tab[4] = { Q31(1.0000000000/2), Q31(1.1892071150/2), Q31(1.4142135624/2), Q31(1.6817928305/2) };  // 2^0, 2^0.25, 2^0.5, 2^0.75
+
+static inline int *DEC_SPAIR(int *dst, unsigned idx)
+{
+    dst[0] = (idx & 15) - 4;
+    dst[1] = (idx >> 4 & 15) - 4;
+
+    return dst + 2;
+}
+
+static inline int *DEC_SQUAD(int *dst, unsigned idx)
+{
+    dst[0] = (idx & 3) - 1;
+    dst[1] = (idx >> 2 & 3) - 1;
+    dst[2] = (idx >> 4 & 3) - 1;
+    dst[3] = (idx >> 6 & 3) - 1;
+
+    return dst + 4;
+}
+
+static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
+{
+    dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
+    dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
+
+    return dst + 2;
+}
+
+static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
+{
+    unsigned nz = idx >> 12;
+
+    dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
+    sign <<= nz & 1;
+    nz >>= 1;
+    dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
+    sign <<= nz & 1;
+    nz >>= 1;
+    dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
+    sign <<= nz & 1;
+    nz >>= 1;
+    dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
+
+    return dst + 4;
+}
+
+static void vector_pow43(int *coefs, int len)
+{
+    int i, coef;
+
+    for (i=0; i<len; i++) {
+        coef = coefs[i];
+        if (coef < 0)
+            coef = -(int)ff_cbrt_tab_fixed[-coef];
+        else
+            coef = (int)ff_cbrt_tab_fixed[coef];
+        coefs[i] = coef;
+    }
+}
+
+static void subband_scale(int *dst, int *src, int scale, int offset, int len)
+{
+    int ssign = scale < 0 ? -1 : 1;
+    int s = FFABS(scale);
+    unsigned int round;
+    int i, out, c = exp2tab[s & 3];
+
+    s = offset - (s >> 2);
+
+    if (s > 31) {
+        for (i=0; i<len; i++) {
+            dst[i] = 0;
+        }
+    } else if (s > 0) {
+        round = 1 << (s-1);
+        for (i=0; i<len; i++) {
+            out = (int)(((int64_t)src[i] * c) >> 32);
+            dst[i] = ((int)(out+round) >> s) * ssign;
+        }
+    } else if (s > -32) {
+        s = s + 32;
+        round = 1U << (s-1);
+        for (i=0; i<len; i++) {
+            out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
+            dst[i] = out * (unsigned)ssign;
+        }
+    } else {
+        av_log(NULL, AV_LOG_ERROR, "Overflow in subband_scale()\n");
+    }
+}
+
+static void noise_scale(int *coefs, int scale, int band_energy, int len)
+{
+    int s = -scale;
+    unsigned int round;
+    int i, out, c = exp2tab[s & 3];
+    int nlz = 0;
+
+    av_assert0(s >= 0);
+    while (band_energy > 0x7fff) {
+        band_energy >>= 1;
+        nlz++;
+    }
+    c /= band_energy;
+    s = 21 + nlz - (s >> 2);
+
+    if (s > 31) {
+        for (i=0; i<len; i++) {
+            coefs[i] = 0;
+        }
+    } else if (s >= 0) {
+        round = s ? 1 << (s-1) : 0;
+        for (i=0; i<len; i++) {
+            out = (int)(((int64_t)coefs[i] * c) >> 32);
+            coefs[i] = -((int)(out+round) >> s);
+        }
+    }
+    else {
+        s = s + 32;
+        if (s > 0) {
+            round = 1 << (s-1);
+            for (i=0; i<len; i++) {
+                out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
+                coefs[i] = -out;
+            }
+        } else {
+            for (i=0; i<len; i++)
+                coefs[i] = -(int64_t)coefs[i] * c * (1 << -s);
+        }
+    }
+}
+
+static av_always_inline SoftFloat flt16_round(SoftFloat pf)
+{
+    SoftFloat tmp;
+    int s;
+
+    tmp.exp = pf.exp;
+    s = pf.mant >> 31;
+    tmp.mant = (pf.mant ^ s) - s;
+    tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U;
+    tmp.mant = (tmp.mant ^ s) - s;
+
+    return tmp;
+}
+
+static av_always_inline SoftFloat flt16_even(SoftFloat pf)
+{
+    SoftFloat tmp;
+    int s;
+
+    tmp.exp = pf.exp;
+    s = pf.mant >> 31;
+    tmp.mant = (pf.mant ^ s) - s;
+    tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U;
+    tmp.mant = (tmp.mant ^ s) - s;
+
+    return tmp;
+}
+
+static av_always_inline SoftFloat flt16_trunc(SoftFloat pf)
+{
+    SoftFloat pun;
+    int s;
+
+    pun.exp = pf.exp;
+    s = pf.mant >> 31;
+    pun.mant = (pf.mant ^ s) - s;
+    pun.mant = pun.mant & 0xFFC00000U;
+    pun.mant = (pun.mant ^ s) - s;
+
+    return pun;
+}
+
+static av_always_inline void predict(PredictorState *ps, int *coef,
+                                     int output_enable)
+{
+    const SoftFloat a     = { 1023410176, 0 };  // 61.0 / 64
+    const SoftFloat alpha = {  973078528, 0 };  // 29.0 / 32
+    SoftFloat e0, e1;
+    SoftFloat pv;
+    SoftFloat k1, k2;
+    SoftFloat   r0 = ps->r0,     r1 = ps->r1;
+    SoftFloat cor0 = ps->cor0, cor1 = ps->cor1;
+    SoftFloat var0 = ps->var0, var1 = ps->var1;
+    SoftFloat tmp;
+
+    if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) {
+        k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0)));
+    }
+    else {
+        k1.mant = 0;
+        k1.exp = 0;
+    }
+
+    if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
+        k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1)));
+    }
+    else {
+        k2.mant = 0;
+        k2.exp = 0;
+    }
+
+    tmp = av_mul_sf(k1, r0);
+    pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1)));
+    if (output_enable) {
+        int shift = 28 - pv.exp;
+
+        if (shift < 31) {
+            if (shift > 0) {
+                *coef += (unsigned)((pv.mant + (1 << (shift - 1))) >> shift);
+            } else
+                *coef += (unsigned)pv.mant << -shift;
+        }
+    }
+
+    e0 = av_int2sf(*coef, 2);
+    e1 = av_sub_sf(e0, tmp);
+
+    ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1)));
+    tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1));
+    tmp.exp--;
+    ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp));
+    ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0)));
+    tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0));
+    tmp.exp--;
+    ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp));
+
+    ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0))));
+    ps->r0 = flt16_trunc(av_mul_sf(a, e0));
+}
+
+
+static const int cce_scale_fixed[8] = {
+    Q30(1.0),          //2^(0/8)
+    Q30(1.0905077327), //2^(1/8)
+    Q30(1.1892071150), //2^(2/8)
+    Q30(1.2968395547), //2^(3/8)
+    Q30(1.4142135624), //2^(4/8)
+    Q30(1.5422108254), //2^(5/8)
+    Q30(1.6817928305), //2^(6/8)
+    Q30(1.8340080864), //2^(7/8)
+};
+
+/**
+ * Apply dependent channel coupling (applied before IMDCT).
+ *
+ * @param   index   index into coupling gain array
+ */
+static void apply_dependent_coupling_fixed(AACContext *ac,
+                                     SingleChannelElement *target,
+                                     ChannelElement *cce, int index)
+{
+    IndividualChannelStream *ics = &cce->ch[0].ics;
+    const uint16_t *offsets = ics->swb_offset;
+    int *dest = target->coeffs;
+    const int *src = cce->ch[0].coeffs;
+    int g, i, group, k, idx = 0;
+    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
+        av_log(ac->avctx, AV_LOG_ERROR,
+               "Dependent coupling is not supported together with LTP\n");
+        return;
+    }
+    for (g = 0; g < ics->num_window_groups; g++) {
+        for (i = 0; i < ics->max_sfb; i++, idx++) {
+            if (cce->ch[0].band_type[idx] != ZERO_BT) {
+                const int gain = cce->coup.gain[index][idx];
+                int shift, round, c, tmp;
+
+                if (gain < 0) {
+                    c = -cce_scale_fixed[-gain & 7];
+                    shift = (-gain-1024) >> 3;
+                }
+                else {
+                    c = cce_scale_fixed[gain & 7];
+                    shift = (gain-1024) >> 3;
+                }
+
+                if (shift < -31) {
+                    // Nothing to do
+                } else if (shift < 0) {
+                    shift = -shift;
+                    round = 1 << (shift - 1);
+
+                    for (group = 0; group < ics->group_len[g]; group++) {
+                        for (k = offsets[i]; k < offsets[i + 1]; k++) {
+                            tmp = (int)(((int64_t)src[group * 128 + k] * c + \
+                                       (int64_t)0x1000000000) >> 37);
+                            dest[group * 128 + k] += (tmp + (int64_t)round) >> shift;
+                        }
+                    }
+                }
+                else {
+                    for (group = 0; group < ics->group_len[g]; group++) {
+                        for (k = offsets[i]; k < offsets[i + 1]; k++) {
+                            tmp = (int)(((int64_t)src[group * 128 + k] * c + \
+                                        (int64_t)0x1000000000) >> 37);
+                            dest[group * 128 + k] += tmp * (1U << shift);
+                        }
+                    }
+                }
+            }
+        }
+        dest += ics->group_len[g] * 128;
+        src  += ics->group_len[g] * 128;
+    }
+}
+
+/**
+ * Apply independent channel coupling (applied after IMDCT).
+ *
+ * @param   index   index into coupling gain array
+ */
+static void apply_independent_coupling_fixed(AACContext *ac,
+                                       SingleChannelElement *target,
+                                       ChannelElement *cce, int index)
+{
+    int i, c, shift, round, tmp;
+    const int gain = cce->coup.gain[index][0];
+    const int *src = cce->ch[0].ret;
+    unsigned int *dest = target->ret;
+    const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
+
+    c = cce_scale_fixed[gain & 7];
+    shift = (gain-1024) >> 3;
+    if (shift < -31) {
+        return;
+    } else if (shift < 0) {
+        shift = -shift;
+        round = 1 << (shift - 1);
+
+        for (i = 0; i < len; i++) {
+            tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
+            dest[i] += (tmp + round) >> shift;
+        }
+    }
+    else {
+      for (i = 0; i < len; i++) {
+          tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
+          dest[i] += tmp * (1U << shift);
+      }
+    }
+}
+
+#include "aacdec_template.c"
+
+AVCodec ff_aac_fixed_decoder = {
+    .name            = "aac_fixed",
+    .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
+    .type            = AVMEDIA_TYPE_AUDIO,
+    .id              = AV_CODEC_ID_AAC,
+    .priv_data_size  = sizeof(AACContext),
+    .init            = aac_decode_init,
+    .close           = aac_decode_close,
+    .decode          = aac_decode_frame,
+    .sample_fmts     = (const enum AVSampleFormat[]) {
+        AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
+    },
+    .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
+    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE,
+    .channel_layouts = aac_channel_layout,
+    .profiles        = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
+    .flush = flush,
+};

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacdectab.h

@@ -0,0 +1,74 @@
+/*
+ * AAC decoder data
+ * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
+ * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC decoder data
+ * @author Oded Shimon  ( ods15 ods15 dyndns org )
+ * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
+ */
+
+#ifndef AVCODEC_AACDECTAB_H
+#define AVCODEC_AACDECTAB_H
+
+#include "libavutil/channel_layout.h"
+#include "aac.h"
+
+#include <stdint.h>
+
+static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 4, 5, 0, 5, 0 };
+
+static const uint8_t aac_channel_layout_map[16][5][3] = {
+    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, },
+    { { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, },
+    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, },
+    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_SCE, 1, AAC_CHANNEL_BACK }, },
+    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, },
+    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
+    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_FRONT }, { TYPE_CPE, 2, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
+    { { 0, } },
+    { { 0, } },
+    { { 0, } },
+    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, { TYPE_SCE, 1, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
+    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_SIDE }, { TYPE_CPE, 2, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
+    { { 0, } },
+    /* TODO: Add 7+1 TOP configuration */
+};
+
+static const uint64_t aac_channel_layout[16] = {
+    AV_CH_LAYOUT_MONO,
+    AV_CH_LAYOUT_STEREO,
+    AV_CH_LAYOUT_SURROUND,
+    AV_CH_LAYOUT_4POINT0,
+    AV_CH_LAYOUT_5POINT0_BACK,
+    AV_CH_LAYOUT_5POINT1_BACK,
+    AV_CH_LAYOUT_7POINT1_WIDE_BACK,
+    0,
+    0,
+    0,
+    AV_CH_LAYOUT_6POINT1,
+    AV_CH_LAYOUT_7POINT1,
+    0,
+    /* AV_CH_LAYOUT_7POINT1_TOP, */
+};
+
+#endif /* AVCODEC_AACDECTAB_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacenc.h

@@ -0,0 +1,428 @@
+/*
+ * AAC encoder
+ * Copyright (C) 2008 Konstantin Shishkov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AACENC_H
+#define AVCODEC_AACENC_H
+
+#include "libavutil/float_dsp.h"
+#include "avcodec.h"
+#include "put_bits.h"
+
+#include "aac.h"
+#include "audio_frame_queue.h"
+#include "psymodel.h"
+
+#include "lpc.h"
+
+typedef enum AACCoder {
+    AAC_CODER_ANMR = 0,
+    AAC_CODER_TWOLOOP,
+    AAC_CODER_FAST,
+
+    AAC_CODER_NB,
+}AACCoder;
+
+typedef struct AACEncOptions {
+    int coder;
+    int pns;
+    int tns;
+    int ltp;
+    int pce;
+    int pred;
+    int mid_side;
+    int intensity_stereo;
+} AACEncOptions;
+
+struct AACEncContext;
+
+typedef struct AACCoefficientsEncoder {
+    void (*search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s,
+                                  SingleChannelElement *sce, const float lambda);
+    void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce,
+                                     int win, int group_len, const float lambda);
+    void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, float *out, int size,
+                                     int scale_idx, int cb, const float lambda, int rtz);
+    void (*encode_tns_info)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*encode_ltp_info)(struct AACEncContext *s, SingleChannelElement *sce, int common_window);
+    void (*encode_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*adjust_common_pred)(struct AACEncContext *s, ChannelElement *cpe);
+    void (*adjust_common_ltp)(struct AACEncContext *s, ChannelElement *cpe);
+    void (*apply_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*apply_tns_filt)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*update_ltp)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*ltp_insert_new_frame)(struct AACEncContext *s);
+    void (*set_special_band_scalefactors)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*search_for_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce);
+    void (*mark_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce);
+    void (*search_for_tns)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*search_for_ltp)(struct AACEncContext *s, SingleChannelElement *sce, int common_window);
+    void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe);
+    void (*search_for_is)(struct AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe);
+    void (*search_for_pred)(struct AACEncContext *s, SingleChannelElement *sce);
+} AACCoefficientsEncoder;
+
+extern const AACCoefficientsEncoder ff_aac_coders[];
+
+typedef struct AACQuantizeBandCostCacheEntry {
+    float rd;
+    float energy;
+    int bits;
+    char cb;
+    char rtz;
+    uint16_t generation;
+} AACQuantizeBandCostCacheEntry;
+
+typedef struct AACPCEInfo {
+    int64_t layout;
+    int num_ele[4];                              ///< front, side, back, lfe
+    int pairing[3][8];                           ///< front, side, back
+    int index[4][8];                             ///< front, side, back, lfe
+    uint8_t config_map[16];                      ///< configs the encoder's channel specific settings
+    uint8_t reorder_map[16];                     ///< maps channels from lavc to aac order
+} AACPCEInfo;
+
+/**
+ * List of PCE (Program Configuration Element) for the channel layouts listed
+ * in channel_layout.h
+ *
+ * For those wishing in the future to add other layouts:
+ *
+ * - num_ele: number of elements in each group of front, side, back, lfe channels
+ *            (an element is of type SCE (single channel), CPE (channel pair) for
+ *            the first 3 groups; and is LFE for LFE group).
+ *
+ * - pairing: 0 for an SCE element or 1 for a CPE; does not apply to LFE group
+ *
+ * - index: there are three independent indices for SCE, CPE and LFE;
+ *     they are incremented irrespective of the group to which the element belongs;
+ *     they are not reset when going from one group to another
+ *
+ *     Example: for 7.0 channel layout,
+ *        .pairing = { { 1, 0 }, { 1 }, { 1 }, }, (3 CPE and 1 SCE in front group)
+ *        .index = { { 0, 0 }, { 1 }, { 2 }, },
+ *               (index is 0 for the single SCE but goes from 0 to 2 for the CPEs)
+ *
+ *     The index order impacts the channel ordering. But is otherwise arbitrary
+ *     (the sequence could have been 2, 0, 1 instead of 0, 1, 2).
+ *
+ *     Spec allows for discontinuous indices, e.g. if one has a total of two SCE,
+ *     SCE.0 SCE.15 is OK per spec; BUT it won't be decoded by our AAC decoder
+ *     which at this time requires that indices fully cover some range starting
+ *     from 0 (SCE.1 SCE.0 is OK but not SCE.0 SCE.15).
+ *
+ * - config_map: total number of elements and their types. Beware, the way the
+ *               types are ordered impacts the final channel ordering.
+ *
+ * - reorder_map: reorders the channels.
+ *
+ */
+static const AACPCEInfo aac_pce_configs[] = {
+    {
+        .layout = AV_CH_LAYOUT_MONO,
+        .num_ele = { 1, 0, 0, 0 },
+        .pairing = { { 0 }, },
+        .index = { { 0 }, },
+        .config_map = { 1, TYPE_SCE, },
+        .reorder_map = { 0 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_STEREO,
+        .num_ele = { 1, 0, 0, 0 },
+        .pairing = { { 1 }, },
+        .index = { { 0 }, },
+        .config_map = { 1, TYPE_CPE, },
+        .reorder_map = { 0, 1 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_2POINT1,
+        .num_ele = { 1, 0, 0, 1 },
+        .pairing = { { 1 }, },
+        .index = { { 0 },{ 0 },{ 0 },{ 0 } },
+        .config_map = { 2, TYPE_CPE, TYPE_LFE },
+        .reorder_map = { 0, 1, 2 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_2_1,
+        .num_ele = { 1, 0, 1, 0 },
+        .pairing = { { 1 },{ 0 },{ 0 } },
+        .index = { { 0 },{ 0 },{ 0 }, },
+        .config_map = { 2, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_SURROUND,
+        .num_ele = { 2, 0, 0, 0 },
+        .pairing = { { 1, 0 }, },
+        .index = { { 0, 0 }, },
+        .config_map = { 2, TYPE_CPE, TYPE_SCE, },
+        .reorder_map = { 0, 1, 2 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_3POINT1,
+        .num_ele = { 2, 0, 0, 1 },
+        .pairing = { { 1, 0 }, },
+        .index = { { 0, 0 }, { 0 }, { 0 }, { 0 }, },
+        .config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_LFE },
+        .reorder_map = { 0, 1, 2, 3 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_4POINT0,
+        .num_ele = { 2, 0, 1, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 0 }, },
+        .index = { { 0, 0 }, { 0 }, { 1 } },
+        .config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_SCE },
+        .reorder_map = {  0, 1, 2, 3 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_4POINT1,
+        .num_ele = { 2, 1, 1, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 0 }, },
+        .index = { { 0, 0 }, { 1 }, { 2 }, { 0 } },
+        .config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_2_2,
+        .num_ele = { 1, 1, 0, 0 },
+        .pairing = { { 1 }, { 1 }, },
+        .index = { { 0 }, { 1 }, },
+        .config_map = { 2, TYPE_CPE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_QUAD,
+        .num_ele = { 1, 0, 1, 0 },
+        .pairing = { { 1 }, { 0 }, { 1 }, },
+        .index = { { 0 }, { 0 }, { 1 } },
+        .config_map = { 2, TYPE_CPE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_5POINT0,
+        .num_ele = { 2, 1, 0, 0 },
+        .pairing = { { 1, 0 }, { 1 }, },
+        .index = { { 0, 0 }, { 1 } },
+        .config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_5POINT1,
+        .num_ele = { 2, 1, 1, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 1 }, },
+        .index = { { 0, 0 }, { 1 }, { 1 } },
+        .config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_5POINT0_BACK,
+        .num_ele = { 2, 0, 1, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 1 } },
+        .index = { { 0, 0 }, { 0 }, { 1 } },
+        .config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_5POINT1_BACK,
+        .num_ele = { 2, 1, 1, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 1 }, },
+        .index = { { 0, 0 }, { 1 }, { 1 } },
+        .config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_6POINT0,
+        .num_ele = { 2, 1, 1, 0 },
+        .pairing = { { 1, 0 }, { 1 }, { 0 }, },
+        .index = { { 0, 0 }, { 1 }, { 1 } },
+        .config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_6POINT0_FRONT,
+        .num_ele = { 2, 1, 0, 0 },
+        .pairing = { { 1, 1 }, { 1 } },
+        .index = { { 1, 0 }, { 2 }, },
+        .config_map = { 3, TYPE_CPE, TYPE_CPE, TYPE_CPE, },
+        .reorder_map = { 0, 1, 2, 3, 4, 5 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_HEXAGONAL,
+        .num_ele = { 2, 0, 2, 0 },
+        .pairing = { { 1, 0 },{ 0 },{ 1, 0 }, },
+        .index = { { 0, 0 },{ 0 },{ 1, 1 } },
+        .config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE, },
+        .reorder_map = { 0, 1, 2, 3, 4, 5 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_6POINT1,
+        .num_ele = { 2, 1, 2, 0 },
+        .pairing = { { 1, 0 },{ 0 },{ 1, 0 }, },
+        .index = { { 0, 0 },{ 1 },{ 1, 2 } },
+        .config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_6POINT1_BACK,
+        .num_ele = { 2, 1, 2, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 1, 0 }, },
+        .index = { { 0, 0 }, { 1 }, { 1, 2 } },
+        .config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_6POINT1_FRONT,
+        .num_ele = { 2, 1, 2, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 1, 0 }, },
+        .index = { { 0, 0 }, { 1 }, { 1, 2 } },
+        .config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_7POINT0,
+        .num_ele = { 2, 1, 1, 0 },
+        .pairing = { { 1, 0 }, { 1 }, { 1 }, },
+        .index = { { 0, 0 }, { 1 }, { 2 }, },
+        .config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_7POINT0_FRONT,
+        .num_ele = { 2, 1, 1, 0 },
+        .pairing = { { 1, 0 }, { 1 }, { 1 }, },
+        .index = { { 0, 0 }, { 1 }, { 2 }, },
+        .config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_7POINT1,
+        .num_ele = { 2, 1, 2, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 1, 1 }, },
+        .index = { { 0, 0 }, { 1 }, { 1, 2 }, { 0 } },
+        .config_map = { 5, TYPE_CPE, TYPE_SCE,  TYPE_SCE, TYPE_CPE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_7POINT1_WIDE,
+        .num_ele = { 2, 1, 2, 0 },
+        .pairing = { { 1, 0 }, { 0 },{  1, 1 }, },
+        .index = { { 0, 0 }, { 1 }, { 1, 2 }, { 0 } },
+        .config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_7POINT1_WIDE_BACK,
+        .num_ele = { 2, 1, 2, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 1, 1 }, },
+        .index = { { 0, 0 }, { 1 }, { 1, 2 }, { 0 } },
+        .config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_OCTAGONAL,
+        .num_ele = { 2, 1, 2, 0 },
+        .pairing = { { 1, 0 }, { 1 }, { 1, 0 }, },
+        .index = { { 0, 0 }, { 1 }, { 2, 1 } },
+        .config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 },
+    },
+    {   /* Meant for order 2/mixed ambisonics */
+        .layout = AV_CH_LAYOUT_OCTAGONAL | AV_CH_TOP_CENTER,
+        .num_ele = { 2, 2, 2, 0 },
+        .pairing = { { 1, 0 }, { 1, 0 }, { 1, 0 }, },
+        .index = { { 0, 0 }, { 1, 1 }, { 2, 2 } },
+        .config_map = { 6, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8 },
+    },
+    {   /* Meant for order 2/mixed ambisonics */
+        .layout = AV_CH_LAYOUT_6POINT0_FRONT | AV_CH_BACK_CENTER |
+                  AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT | AV_CH_TOP_CENTER,
+        .num_ele = { 2, 2, 2, 0 },
+        .pairing = { { 1, 1 }, { 1, 0 }, { 1, 0 }, },
+        .index = { { 0, 1 }, { 2, 0 }, { 3, 1 } },
+        .config_map = { 6, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9 },
+    },
+    {
+        .layout = AV_CH_LAYOUT_HEXADECAGONAL,
+        .num_ele = { 4, 2, 4, 0 },
+        .pairing = { { 1, 0, 1, 0 }, { 1, 1 }, { 1, 0, 1, 0 }, },
+        .index = { { 0, 0, 1, 1 }, { 2, 3 }, { 4, 2, 5, 3 } },
+        .config_map = { 10, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15 },
+    },
+};
+
+/**
+ * AAC encoder context
+ */
+typedef struct AACEncContext {
+    AVClass *av_class;
+    AACEncOptions options;                       ///< encoding options
+    PutBitContext pb;
+    FFTContext mdct1024;                         ///< long (1024 samples) frame transform context
+    FFTContext mdct128;                          ///< short (128 samples) frame transform context
+    AVFloatDSPContext *fdsp;
+    AACPCEInfo pce;                              ///< PCE data, if needed
+    float *planar_samples[16];                   ///< saved preprocessed input
+
+    int profile;                                 ///< copied from avctx
+    int needs_pce;                               ///< flag for non-standard layout
+    LPCContext lpc;                              ///< used by TNS
+    int samplerate_index;                        ///< MPEG-4 samplerate index
+    int channels;                                ///< channel count
+    const uint8_t *reorder_map;                  ///< lavc to aac reorder map
+    const uint8_t *chan_map;                     ///< channel configuration map
+
+    ChannelElement *cpe;                         ///< channel elements
+    FFPsyContext psy;
+    struct FFPsyPreprocessContext* psypp;
+    const AACCoefficientsEncoder *coder;
+    int cur_channel;                             ///< current channel for coder context
+    int random_state;
+    float lambda;
+    int last_frame_pb_count;                     ///< number of bits for the previous frame
+    float lambda_sum;                            ///< sum(lambda), for Qvg reporting
+    int lambda_count;                            ///< count(lambda), for Qvg reporting
+    enum RawDataBlockType cur_type;              ///< channel group type cur_channel belongs to
+
+    AudioFrameQueue afq;
+    DECLARE_ALIGNED(16, int,   qcoefs)[96];      ///< quantized coefficients
+    DECLARE_ALIGNED(32, float, scoefs)[1024];    ///< scaled coefficients
+
+    uint16_t quantize_band_cost_cache_generation;
+    AACQuantizeBandCostCacheEntry quantize_band_cost_cache[256][128]; ///< memoization area for quantize_band_cost
+
+    void (*abs_pow34)(float *out, const float *in, const int size);
+    void (*quant_bands)(int *out, const float *in, const float *scaled,
+                        int size, int is_signed, int maxval, const float Q34,
+                        const float rounding);
+
+    struct {
+        float *samples;
+    } buffer;
+} AACEncContext;
+
+void ff_aac_dsp_init_x86(AACEncContext *s);
+void ff_aac_coder_init_mips(AACEncContext *c);
+void ff_quantize_band_cost_cache_init(struct AACEncContext *s);
+
+
+#endif /* AVCODEC_AACENC_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacenc_is.c

@@ -0,0 +1,158 @@
+/*
+ * AAC encoder intensity stereo
+ * Copyright (C) 2015 Rostislav Pehlivanov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder Intensity Stereo
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#include "aacenc.h"
+#include "aacenc_utils.h"
+#include "aacenc_is.h"
+#include "aacenc_quantization.h"
+
+struct AACISError ff_aac_is_encoding_err(AACEncContext *s, ChannelElement *cpe,
+                                         int start, int w, int g, float ener0,
+                                         float ener1, float ener01,
+                                         int use_pcoeffs, int phase)
+{
+    int i, w2;
+    SingleChannelElement *sce0 = &cpe->ch[0];
+    SingleChannelElement *sce1 = &cpe->ch[1];
+    float *L = use_pcoeffs ? sce0->pcoeffs : sce0->coeffs;
+    float *R = use_pcoeffs ? sce1->pcoeffs : sce1->coeffs;
+    float *L34 = &s->scoefs[256*0], *R34 = &s->scoefs[256*1];
+    float *IS  = &s->scoefs[256*2], *I34 = &s->scoefs[256*3];
+    float dist1 = 0.0f, dist2 = 0.0f;
+    struct AACISError is_error = {0};
+
+    if (ener01 <= 0 || ener0 <= 0) {
+        is_error.pass = 0;
+        return is_error;
+    }
+
+    for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
+        FFPsyBand *band0 = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
+        FFPsyBand *band1 = &s->psy.ch[s->cur_channel+1].psy_bands[(w+w2)*16+g];
+        int is_band_type, is_sf_idx = FFMAX(1, sce0->sf_idx[w*16+g]-4);
+        float e01_34 = phase*pos_pow34(ener1/ener0);
+        float maxval, dist_spec_err = 0.0f;
+        float minthr = FFMIN(band0->threshold, band1->threshold);
+        for (i = 0; i < sce0->ics.swb_sizes[g]; i++)
+            IS[i] = (L[start+(w+w2)*128+i] + phase*R[start+(w+w2)*128+i])*sqrt(ener0/ener01);
+        s->abs_pow34(L34, &L[start+(w+w2)*128], sce0->ics.swb_sizes[g]);
+        s->abs_pow34(R34, &R[start+(w+w2)*128], sce0->ics.swb_sizes[g]);
+        s->abs_pow34(I34, IS,                   sce0->ics.swb_sizes[g]);
+        maxval = find_max_val(1, sce0->ics.swb_sizes[g], I34);
+        is_band_type = find_min_book(maxval, is_sf_idx);
+        dist1 += quantize_band_cost(s, &L[start + (w+w2)*128], L34,
+                                    sce0->ics.swb_sizes[g],
+                                    sce0->sf_idx[w*16+g],
+                                    sce0->band_type[w*16+g],
+                                    s->lambda / band0->threshold, INFINITY, NULL, NULL, 0);
+        dist1 += quantize_band_cost(s, &R[start + (w+w2)*128], R34,
+                                    sce1->ics.swb_sizes[g],
+                                    sce1->sf_idx[w*16+g],
+                                    sce1->band_type[w*16+g],
+                                    s->lambda / band1->threshold, INFINITY, NULL, NULL, 0);
+        dist2 += quantize_band_cost(s, IS, I34, sce0->ics.swb_sizes[g],
+                                    is_sf_idx, is_band_type,
+                                    s->lambda / minthr, INFINITY, NULL, NULL, 0);
+        for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
+            dist_spec_err += (L34[i] - I34[i])*(L34[i] - I34[i]);
+            dist_spec_err += (R34[i] - I34[i]*e01_34)*(R34[i] - I34[i]*e01_34);
+        }
+        dist_spec_err *= s->lambda / minthr;
+        dist2 += dist_spec_err;
+    }
+
+    is_error.pass = dist2 <= dist1;
+    is_error.phase = phase;
+    is_error.error = dist2 - dist1;
+    is_error.dist1 = dist1;
+    is_error.dist2 = dist2;
+    is_error.ener01 = ener01;
+
+    return is_error;
+}
+
+void ff_aac_search_for_is(AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe)
+{
+    SingleChannelElement *sce0 = &cpe->ch[0];
+    SingleChannelElement *sce1 = &cpe->ch[1];
+    int start = 0, count = 0, w, w2, g, i, prev_sf1 = -1, prev_bt = -1, prev_is = 0;
+    const float freq_mult = avctx->sample_rate/(1024.0f/sce0->ics.num_windows)/2.0f;
+    uint8_t nextband1[128];
+
+    if (!cpe->common_window)
+        return;
+
+    /** Scout out next nonzero bands */
+    ff_init_nextband_map(sce1, nextband1);
+
+    for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
+        start = 0;
+        for (g = 0;  g < sce0->ics.num_swb; g++) {
+            if (start*freq_mult > INT_STEREO_LOW_LIMIT*(s->lambda/170.0f) &&
+                cpe->ch[0].band_type[w*16+g] != NOISE_BT && !cpe->ch[0].zeroes[w*16+g] &&
+                cpe->ch[1].band_type[w*16+g] != NOISE_BT && !cpe->ch[1].zeroes[w*16+g] &&
+                ff_sfdelta_can_remove_band(sce1, nextband1, prev_sf1, w*16+g)) {
+                float ener0 = 0.0f, ener1 = 0.0f, ener01 = 0.0f, ener01p = 0.0f;
+                struct AACISError ph_err1, ph_err2, *best;
+                for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
+                    for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
+                        float coef0 = sce0->coeffs[start+(w+w2)*128+i];
+                        float coef1 = sce1->coeffs[start+(w+w2)*128+i];
+                        ener0  += coef0*coef0;
+                        ener1  += coef1*coef1;
+                        ener01 += (coef0 + coef1)*(coef0 + coef1);
+                        ener01p += (coef0 - coef1)*(coef0 - coef1);
+                    }
+                }
+                ph_err1 = ff_aac_is_encoding_err(s, cpe, start, w, g,
+                                                 ener0, ener1, ener01p, 0, -1);
+                ph_err2 = ff_aac_is_encoding_err(s, cpe, start, w, g,
+                                                 ener0, ener1, ener01, 0, +1);
+                best = (ph_err1.pass && ph_err1.error < ph_err2.error) ? &ph_err1 : &ph_err2;
+                if (best->pass) {
+                    cpe->is_mask[w*16+g] = 1;
+                    cpe->ms_mask[w*16+g] = 0;
+                    cpe->ch[0].is_ener[w*16+g] = sqrt(ener0 / best->ener01);
+                    cpe->ch[1].is_ener[w*16+g] = ener0/ener1;
+                    cpe->ch[1].band_type[w*16+g] = (best->phase > 0) ? INTENSITY_BT : INTENSITY_BT2;
+                    if (prev_is && prev_bt != cpe->ch[1].band_type[w*16+g]) {
+                        /** Flip M/S mask and pick the other CB, since it encodes more efficiently */
+                        cpe->ms_mask[w*16+g] = 1;
+                        cpe->ch[1].band_type[w*16+g] = (best->phase > 0) ? INTENSITY_BT2 : INTENSITY_BT;
+                    }
+                    prev_bt = cpe->ch[1].band_type[w*16+g];
+                    count++;
+                }
+            }
+            if (!sce1->zeroes[w*16+g] && sce1->band_type[w*16+g] < RESERVED_BT)
+                prev_sf1 = sce1->sf_idx[w*16+g];
+            prev_is = cpe->is_mask[w*16+g];
+            start += sce0->ics.swb_sizes[g];
+        }
+    }
+    cpe->is_mode = !!count;
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacenc_is.h

@@ -0,0 +1,51 @@
+/*
+ * AAC encoder intensity stereo
+ * Copyright (C) 2015 Rostislav Pehlivanov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder Intensity Stereo
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#ifndef AVCODEC_AACENC_IS_H
+#define AVCODEC_AACENC_IS_H
+
+#include "aacenc.h"
+
+/** Frequency in Hz for lower limit of intensity stereo **/
+#define INT_STEREO_LOW_LIMIT 6100
+
+struct AACISError {
+    int pass;    /* 1 if dist2 <= dist1  */
+    int phase;   /* -1 or +1             */
+    float error; /* fabs(dist1 - dist2)  */
+    float dist1; /* From original coeffs */
+    float dist2; /* From IS'd coeffs     */
+    float ener01;
+};
+
+struct AACISError ff_aac_is_encoding_err(AACEncContext *s, ChannelElement *cpe,
+                                         int start, int w, int g, float ener0,
+                                         float ener1, float ener01,
+                                         int use_pcoeffs, int phase);
+void ff_aac_search_for_is(AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe);
+
+#endif /* AVCODEC_AACENC_IS_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacenc_ltp.c

@@ -0,0 +1,236 @@
+/*
+ * AAC encoder long term prediction extension
+ * Copyright (C) 2015 Rostislav Pehlivanov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder long term prediction extension
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#include "aacenc_ltp.h"
+#include "aacenc_quantization.h"
+#include "aacenc_utils.h"
+
+/**
+ * Encode LTP data.
+ */
+void ff_aac_encode_ltp_info(AACEncContext *s, SingleChannelElement *sce,
+                            int common_window)
+{
+    int i;
+    IndividualChannelStream *ics = &sce->ics;
+    if (s->profile != FF_PROFILE_AAC_LTP || !ics->predictor_present)
+        return;
+    if (common_window)
+        put_bits(&s->pb, 1, 0);
+    put_bits(&s->pb, 1, ics->ltp.present);
+    if (!ics->ltp.present)
+        return;
+    put_bits(&s->pb, 11, ics->ltp.lag);
+    put_bits(&s->pb, 3,  ics->ltp.coef_idx);
+    for (i = 0; i < FFMIN(ics->max_sfb, MAX_LTP_LONG_SFB); i++)
+        put_bits(&s->pb, 1, ics->ltp.used[i]);
+}
+
+void ff_aac_ltp_insert_new_frame(AACEncContext *s)
+{
+    int i, ch, tag, chans, cur_channel, start_ch = 0;
+    ChannelElement *cpe;
+    SingleChannelElement *sce;
+    for (i = 0; i < s->chan_map[0]; i++) {
+        cpe = &s->cpe[i];
+        tag      = s->chan_map[i+1];
+        chans    = tag == TYPE_CPE ? 2 : 1;
+        for (ch = 0; ch < chans; ch++) {
+            sce = &cpe->ch[ch];
+            cur_channel = start_ch + ch;
+            /* New sample + overlap */
+            memcpy(&sce->ltp_state[0],    &sce->ltp_state[1024], 1024*sizeof(sce->ltp_state[0]));
+            memcpy(&sce->ltp_state[1024], &s->planar_samples[cur_channel][2048], 1024*sizeof(sce->ltp_state[0]));
+            memcpy(&sce->ltp_state[2048], &sce->ret_buf[0], 1024*sizeof(sce->ltp_state[0]));
+            sce->ics.ltp.lag = 0;
+        }
+        start_ch += chans;
+    }
+}
+
+static void get_lag(float *buf, const float *new, LongTermPrediction *ltp)
+{
+    int i, j, lag = 0, max_corr = 0;
+    float max_ratio = 0.0f;
+    for (i = 0; i < 2048; i++) {
+        float corr, s0 = 0.0f, s1 = 0.0f;
+        const int start = FFMAX(0, i - 1024);
+        for (j = start; j < 2048; j++) {
+            const int idx = j - i + 1024;
+            s0 += new[j]*buf[idx];
+            s1 += buf[idx]*buf[idx];
+        }
+        corr = s1 > 0.0f ? s0/sqrt(s1) : 0.0f;
+        if (corr > max_corr) {
+            max_corr = corr;
+            lag = i;
+            max_ratio = corr/(2048-start);
+        }
+    }
+    ltp->lag = FFMAX(av_clip_uintp2(lag, 11), 0);
+    ltp->coef_idx = quant_array_idx(max_ratio, ltp_coef, 8);
+    ltp->coef = ltp_coef[ltp->coef_idx];
+}
+
+static void generate_samples(float *buf, LongTermPrediction *ltp)
+{
+    int i, samples_num = 2048;
+    if (!ltp->lag) {
+        ltp->present = 0;
+        return;
+    } else if (ltp->lag < 1024) {
+        samples_num = ltp->lag + 1024;
+    }
+    for (i = 0; i < samples_num; i++)
+        buf[i] = ltp->coef*buf[i + 2048 - ltp->lag];
+    memset(&buf[i], 0, (2048 - i)*sizeof(float));
+}
+
+/**
+ * Process LTP parameters
+ * @see Patent WO2006070265A1
+ */
+void ff_aac_update_ltp(AACEncContext *s, SingleChannelElement *sce)
+{
+    float *pred_signal = &sce->ltp_state[0];
+    const float *samples = &s->planar_samples[s->cur_channel][1024];
+
+    if (s->profile != FF_PROFILE_AAC_LTP)
+        return;
+
+    /* Calculate lag */
+    get_lag(pred_signal, samples, &sce->ics.ltp);
+    generate_samples(pred_signal, &sce->ics.ltp);
+}
+
+void ff_aac_adjust_common_ltp(AACEncContext *s, ChannelElement *cpe)
+{
+    int sfb, count = 0;
+    SingleChannelElement *sce0 = &cpe->ch[0];
+    SingleChannelElement *sce1 = &cpe->ch[1];
+
+    if (!cpe->common_window ||
+        sce0->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE ||
+        sce1->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+        sce0->ics.ltp.present = 0;
+        return;
+    }
+
+    for (sfb = 0; sfb < FFMIN(sce0->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++) {
+        int sum = sce0->ics.ltp.used[sfb] + sce1->ics.ltp.used[sfb];
+        if (sum != 2) {
+            sce0->ics.ltp.used[sfb] = 0;
+        } else if (sum == 2) {
+            count++;
+        }
+    }
+
+    sce0->ics.ltp.present = !!count;
+    sce0->ics.predictor_present = !!count;
+}
+
+/**
+ * Mark LTP sfb's
+ */
+void ff_aac_search_for_ltp(AACEncContext *s, SingleChannelElement *sce,
+                           int common_window)
+{
+    int w, g, w2, i, start = 0, count = 0;
+    int saved_bits = -(15 + FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB));
+    float *C34 = &s->scoefs[128*0], *PCD = &s->scoefs[128*1];
+    float *PCD34 = &s->scoefs[128*2];
+    const int max_ltp = FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB);
+
+    if (sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+        if (sce->ics.ltp.lag) {
+            memset(&sce->ltp_state[0], 0, 3072*sizeof(sce->ltp_state[0]));
+            memset(&sce->ics.ltp, 0, sizeof(LongTermPrediction));
+        }
+        return;
+    }
+
+    if (!sce->ics.ltp.lag || s->lambda > 120.0f)
+        return;
+
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        start = 0;
+        for (g = 0;  g < sce->ics.num_swb; g++) {
+            int bits1 = 0, bits2 = 0;
+            float dist1 = 0.0f, dist2 = 0.0f;
+            if (w*16+g > max_ltp) {
+                start += sce->ics.swb_sizes[g];
+                continue;
+            }
+            for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                int bits_tmp1, bits_tmp2;
+                FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
+                for (i = 0; i < sce->ics.swb_sizes[g]; i++)
+                    PCD[i] = sce->coeffs[start+(w+w2)*128+i] - sce->lcoeffs[start+(w+w2)*128+i];
+                s->abs_pow34(C34,  &sce->coeffs[start+(w+w2)*128],  sce->ics.swb_sizes[g]);
+                s->abs_pow34(PCD34, PCD, sce->ics.swb_sizes[g]);
+                dist1 += quantize_band_cost(s, &sce->coeffs[start+(w+w2)*128], C34, sce->ics.swb_sizes[g],
+                                            sce->sf_idx[(w+w2)*16+g], sce->band_type[(w+w2)*16+g],
+                                            s->lambda/band->threshold, INFINITY, &bits_tmp1, NULL, 0);
+                dist2 += quantize_band_cost(s, PCD, PCD34, sce->ics.swb_sizes[g],
+                                            sce->sf_idx[(w+w2)*16+g],
+                                            sce->band_type[(w+w2)*16+g],
+                                            s->lambda/band->threshold, INFINITY, &bits_tmp2, NULL, 0);
+                bits1 += bits_tmp1;
+                bits2 += bits_tmp2;
+            }
+            if (dist2 < dist1 && bits2 < bits1) {
+                for (w2 = 0; w2 < sce->ics.group_len[w]; w2++)
+                    for (i = 0; i < sce->ics.swb_sizes[g]; i++)
+                        sce->coeffs[start+(w+w2)*128+i] -= sce->lcoeffs[start+(w+w2)*128+i];
+                sce->ics.ltp.used[w*16+g] = 1;
+                saved_bits += bits1 - bits2;
+                count++;
+            }
+            start += sce->ics.swb_sizes[g];
+        }
+    }
+
+    sce->ics.ltp.present = !!count && (saved_bits >= 0);
+    sce->ics.predictor_present = !!sce->ics.ltp.present;
+
+    /* Reset any marked sfbs */
+    if (!sce->ics.ltp.present && !!count) {
+        for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+            start = 0;
+            for (g = 0;  g < sce->ics.num_swb; g++) {
+                if (sce->ics.ltp.used[w*16+g]) {
+                    for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                        for (i = 0; i < sce->ics.swb_sizes[g]; i++) {
+                            sce->coeffs[start+(w+w2)*128+i] += sce->lcoeffs[start+(w+w2)*128+i];
+                        }
+                    }
+                }
+                start += sce->ics.swb_sizes[g];
+            }
+        }
+    }
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacenc_ltp.h

@@ -0,0 +1,41 @@
+/*
+ * AAC encoder long term prediction extension
+ * Copyright (C) 2015 Rostislav Pehlivanov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder long term prediction extension
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#ifndef AVCODEC_AACENC_LTP_H
+#define AVCODEC_AACENC_LTP_H
+
+#include "aacenc.h"
+
+void ff_aac_encode_ltp_info(AACEncContext *s, SingleChannelElement *sce,
+                            int common_window);
+void ff_aac_update_ltp(AACEncContext *s, SingleChannelElement *sce);
+void ff_aac_adjust_common_ltp(AACEncContext *s, ChannelElement *cpe);
+void ff_aac_ltp_insert_new_frame(AACEncContext *s);
+void ff_aac_search_for_ltp(AACEncContext *s, SingleChannelElement *sce,
+                           int common_window);
+
+#endif /* AVCODEC_AACENC_LTP_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacenc_pred.c

@@ -0,0 +1,347 @@
+/*
+ * AAC encoder main-type prediction
+ * Copyright (C) 2015 Rostislav Pehlivanov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder main-type prediction
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#include "aactab.h"
+#include "aacenc_pred.h"
+#include "aacenc_utils.h"
+#include "aacenc_is.h"            /* <- Needed for common window distortions */
+#include "aacenc_quantization.h"
+
+#define RESTORE_PRED(sce, sfb) \
+        if (sce->ics.prediction_used[sfb]) {\
+            sce->ics.prediction_used[sfb] = 0;\
+            sce->band_type[sfb] = sce->band_alt[sfb];\
+        }
+
+static inline float flt16_round(float pf)
+{
+    union av_intfloat32 tmp;
+    tmp.f = pf;
+    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
+    return tmp.f;
+}
+
+static inline float flt16_even(float pf)
+{
+    union av_intfloat32 tmp;
+    tmp.f = pf;
+    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
+    return tmp.f;
+}
+
+static inline float flt16_trunc(float pf)
+{
+    union av_intfloat32 pun;
+    pun.f = pf;
+    pun.i &= 0xFFFF0000U;
+    return pun.f;
+}
+
+static inline void predict(PredictorState *ps, float *coef, float *rcoef, int set)
+{
+    float k2;
+    const float a     = 0.953125; // 61.0 / 64
+    const float alpha = 0.90625;  // 29.0 / 32
+    const float   k1 = ps->k1;
+    const float   r0 = ps->r0,     r1 = ps->r1;
+    const float cor0 = ps->cor0, cor1 = ps->cor1;
+    const float var0 = ps->var0, var1 = ps->var1;
+    const float e0 = *coef - ps->x_est;
+    const float e1 = e0 - k1 * r0;
+
+    if (set)
+        *coef = e0;
+
+    ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
+    ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
+    ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
+    ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
+    ps->r1   = flt16_trunc(a * (r0 - k1 * e0));
+    ps->r0   = flt16_trunc(a * e0);
+
+    /* Prediction for next frame */
+    ps->k1   = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
+    k2       = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
+    *rcoef   = ps->x_est = flt16_round(ps->k1*ps->r0 + k2*ps->r1);
+}
+
+static inline void reset_predict_state(PredictorState *ps)
+{
+    ps->r0    = 0.0f;
+    ps->r1    = 0.0f;
+    ps->k1    = 0.0f;
+    ps->cor0  = 0.0f;
+    ps->cor1  = 0.0f;
+    ps->var0  = 1.0f;
+    ps->var1  = 1.0f;
+    ps->x_est = 0.0f;
+}
+
+static inline void reset_all_predictors(PredictorState *ps)
+{
+    int i;
+    for (i = 0; i < MAX_PREDICTORS; i++)
+        reset_predict_state(&ps[i]);
+}
+
+static inline void reset_predictor_group(SingleChannelElement *sce, int group_num)
+{
+    int i;
+    PredictorState *ps = sce->predictor_state;
+    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
+        reset_predict_state(&ps[i]);
+}
+
+void ff_aac_apply_main_pred(AACEncContext *s, SingleChannelElement *sce)
+{
+    int sfb, k;
+    const int pmax = FFMIN(sce->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
+
+    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
+        for (sfb = 0; sfb < pmax; sfb++) {
+            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
+                predict(&sce->predictor_state[k], &sce->coeffs[k], &sce->prcoeffs[k],
+                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
+            }
+        }
+        if (sce->ics.predictor_reset_group) {
+            reset_predictor_group(sce, sce->ics.predictor_reset_group);
+        }
+    } else {
+        reset_all_predictors(sce->predictor_state);
+    }
+}
+
+/* If inc = 0 you can check if this returns 0 to see if you can reset freely */
+static inline int update_counters(IndividualChannelStream *ics, int inc)
+{
+    int i;
+    for (i = 1; i < 31; i++) {
+        ics->predictor_reset_count[i] += inc;
+        if (ics->predictor_reset_count[i] > PRED_RESET_FRAME_MIN)
+            return i; /* Reset this immediately */
+    }
+    return 0;
+}
+
+void ff_aac_adjust_common_pred(AACEncContext *s, ChannelElement *cpe)
+{
+    int start, w, w2, g, i, count = 0;
+    SingleChannelElement *sce0 = &cpe->ch[0];
+    SingleChannelElement *sce1 = &cpe->ch[1];
+    const int pmax0 = FFMIN(sce0->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
+    const int pmax1 = FFMIN(sce1->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
+    const int pmax  = FFMIN(pmax0, pmax1);
+
+    if (!cpe->common_window ||
+        sce0->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE ||
+        sce1->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE)
+        return;
+
+    for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
+        start = 0;
+        for (g = 0; g < sce0->ics.num_swb; g++) {
+            int sfb = w*16+g;
+            int sum = sce0->ics.prediction_used[sfb] + sce1->ics.prediction_used[sfb];
+            float ener0 = 0.0f, ener1 = 0.0f, ener01 = 0.0f;
+            struct AACISError ph_err1, ph_err2, *erf;
+            if (sfb < PRED_SFB_START || sfb > pmax || sum != 2) {
+                RESTORE_PRED(sce0, sfb);
+                RESTORE_PRED(sce1, sfb);
+                start += sce0->ics.swb_sizes[g];
+                continue;
+            }
+            for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
+                for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
+                    float coef0 = sce0->pcoeffs[start+(w+w2)*128+i];
+                    float coef1 = sce1->pcoeffs[start+(w+w2)*128+i];
+                    ener0  += coef0*coef0;
+                    ener1  += coef1*coef1;
+                    ener01 += (coef0 + coef1)*(coef0 + coef1);
+                }
+            }
+            ph_err1 = ff_aac_is_encoding_err(s, cpe, start, w, g,
+                                             ener0, ener1, ener01, 1, -1);
+            ph_err2 = ff_aac_is_encoding_err(s, cpe, start, w, g,
+                                             ener0, ener1, ener01, 1, +1);
+            erf = ph_err1.error < ph_err2.error ? &ph_err1 : &ph_err2;
+            if (erf->pass) {
+                sce0->ics.prediction_used[sfb] = 1;
+                sce1->ics.prediction_used[sfb] = 1;
+                count++;
+            } else {
+                RESTORE_PRED(sce0, sfb);
+                RESTORE_PRED(sce1, sfb);
+            }
+            start += sce0->ics.swb_sizes[g];
+        }
+    }
+
+    sce1->ics.predictor_present = sce0->ics.predictor_present = !!count;
+}
+
+static void update_pred_resets(SingleChannelElement *sce)
+{
+    int i, max_group_id_c, max_frame = 0;
+    float avg_frame = 0.0f;
+    IndividualChannelStream *ics = &sce->ics;
+
+    /* Update the counters and immediately update any frame behind schedule */
+    if ((ics->predictor_reset_group = update_counters(&sce->ics, 1)))
+        return;
+
+    for (i = 1; i < 31; i++) {
+        /* Count-based */
+        if (ics->predictor_reset_count[i] > max_frame) {
+            max_group_id_c = i;
+            max_frame = ics->predictor_reset_count[i];
+        }
+        avg_frame = (ics->predictor_reset_count[i] + avg_frame)/2;
+    }
+
+    if (max_frame > PRED_RESET_MIN) {
+        ics->predictor_reset_group = max_group_id_c;
+    } else {
+        ics->predictor_reset_group = 0;
+    }
+}
+
+void ff_aac_search_for_pred(AACEncContext *s, SingleChannelElement *sce)
+{
+    int sfb, i, count = 0, cost_coeffs = 0, cost_pred = 0;
+    const int pmax = FFMIN(sce->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
+    float *O34  = &s->scoefs[128*0], *P34 = &s->scoefs[128*1];
+    float *SENT = &s->scoefs[128*2], *S34 = &s->scoefs[128*3];
+    float *QERR = &s->scoefs[128*4];
+
+    if (sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+        sce->ics.predictor_present = 0;
+        return;
+    }
+
+    if (!sce->ics.predictor_initialized) {
+        reset_all_predictors(sce->predictor_state);
+        sce->ics.predictor_initialized = 1;
+        memcpy(sce->prcoeffs, sce->coeffs, 1024*sizeof(float));
+        for (i = 1; i < 31; i++)
+            sce->ics.predictor_reset_count[i] = i;
+    }
+
+    update_pred_resets(sce);
+    memcpy(sce->band_alt, sce->band_type, sizeof(sce->band_type));
+
+    for (sfb = PRED_SFB_START; sfb < pmax; sfb++) {
+        int cost1, cost2, cb_p;
+        float dist1, dist2, dist_spec_err = 0.0f;
+        const int cb_n = sce->zeroes[sfb] ? 0 : sce->band_type[sfb];
+        const int cb_min = sce->zeroes[sfb] ? 0 : 1;
+        const int cb_max = sce->zeroes[sfb] ? 0 : RESERVED_BT;
+        const int start_coef = sce->ics.swb_offset[sfb];
+        const int num_coeffs = sce->ics.swb_offset[sfb + 1] - start_coef;
+        const FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[sfb];
+
+        if (start_coef + num_coeffs > MAX_PREDICTORS ||
+            (s->cur_channel && sce->band_type[sfb] >= INTENSITY_BT2) ||
+            sce->band_type[sfb] == NOISE_BT)
+            continue;
+
+        /* Normal coefficients */
+        s->abs_pow34(O34, &sce->coeffs[start_coef], num_coeffs);
+        dist1 = quantize_and_encode_band_cost(s, NULL, &sce->coeffs[start_coef], NULL,
+                                              O34, num_coeffs, sce->sf_idx[sfb],
+                                              cb_n, s->lambda / band->threshold, INFINITY, &cost1, NULL, 0);
+        cost_coeffs += cost1;
+
+        /* Encoded coefficients - needed for #bits, band type and quant. error */
+        for (i = 0; i < num_coeffs; i++)
+            SENT[i] = sce->coeffs[start_coef + i] - sce->prcoeffs[start_coef + i];
+        s->abs_pow34(S34, SENT, num_coeffs);
+        if (cb_n < RESERVED_BT)
+            cb_p = av_clip(find_min_book(find_max_val(1, num_coeffs, S34), sce->sf_idx[sfb]), cb_min, cb_max);
+        else
+            cb_p = cb_n;
+        quantize_and_encode_band_cost(s, NULL, SENT, QERR, S34, num_coeffs,
+                                      sce->sf_idx[sfb], cb_p, s->lambda / band->threshold, INFINITY,
+                                      &cost2, NULL, 0);
+
+        /* Reconstructed coefficients - needed for distortion measurements */
+        for (i = 0; i < num_coeffs; i++)
+            sce->prcoeffs[start_coef + i] += QERR[i] != 0.0f ? (sce->prcoeffs[start_coef + i] - QERR[i]) : 0.0f;
+        s->abs_pow34(P34, &sce->prcoeffs[start_coef], num_coeffs);
+        if (cb_n < RESERVED_BT)
+            cb_p = av_clip(find_min_book(find_max_val(1, num_coeffs, P34), sce->sf_idx[sfb]), cb_min, cb_max);
+        else
+            cb_p = cb_n;
+        dist2 = quantize_and_encode_band_cost(s, NULL, &sce->prcoeffs[start_coef], NULL,
+                                              P34, num_coeffs, sce->sf_idx[sfb],
+                                              cb_p, s->lambda / band->threshold, INFINITY, NULL, NULL, 0);
+        for (i = 0; i < num_coeffs; i++)
+            dist_spec_err += (O34[i] - P34[i])*(O34[i] - P34[i]);
+        dist_spec_err *= s->lambda / band->threshold;
+        dist2 += dist_spec_err;
+
+        if (dist2 <= dist1 && cb_p <= cb_n) {
+            cost_pred += cost2;
+            sce->ics.prediction_used[sfb] = 1;
+            sce->band_alt[sfb]  = cb_n;
+            sce->band_type[sfb] = cb_p;
+            count++;
+        } else {
+            cost_pred += cost1;
+            sce->band_alt[sfb] = cb_p;
+        }
+    }
+
+    if (count && cost_coeffs < cost_pred) {
+        count = 0;
+        for (sfb = PRED_SFB_START; sfb < pmax; sfb++)
+            RESTORE_PRED(sce, sfb);
+        memset(&sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
+    }
+
+    sce->ics.predictor_present = !!count;
+}
+
+/**
+ * Encoder predictors data.
+ */
+void ff_aac_encode_main_pred(AACEncContext *s, SingleChannelElement *sce)
+{
+    int sfb;
+    IndividualChannelStream *ics = &sce->ics;
+    const int pmax = FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
+
+    if (s->profile != FF_PROFILE_AAC_MAIN ||
+        !ics->predictor_present)
+        return;
+
+    put_bits(&s->pb, 1, !!ics->predictor_reset_group);
+    if (ics->predictor_reset_group)
+        put_bits(&s->pb, 5, ics->predictor_reset_group);
+    for (sfb = 0; sfb < pmax; sfb++)
+        put_bits(&s->pb, 1, ics->prediction_used[sfb]);
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacenc_pred.h

@@ -0,0 +1,47 @@
+/*
+ * AAC encoder main-type prediction
+ * Copyright (C) 2015 Rostislav Pehlivanov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder main-type prediction
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#ifndef AVCODEC_AACENC_PRED_H
+#define AVCODEC_AACENC_PRED_H
+
+#include "aacenc.h"
+
+/* Every predictor group needs to get reset at least once in this many frames */
+#define PRED_RESET_FRAME_MIN 240
+
+/* Any frame with less than this amount of frames since last reset is ok */
+#define PRED_RESET_MIN 64
+
+/* Raise to filter any low frequency artifacts due to prediction */
+#define PRED_SFB_START 10
+
+void ff_aac_apply_main_pred(AACEncContext *s, SingleChannelElement *sce);
+void ff_aac_adjust_common_pred(AACEncContext *s, ChannelElement *cpe);
+void ff_aac_search_for_pred(AACEncContext *s, SingleChannelElement *sce);
+void ff_aac_encode_main_pred(AACEncContext *s, SingleChannelElement *sce);
+
+#endif /* AVCODEC_AACENC_PRED_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacenc_quantization.h

@@ -0,0 +1,283 @@
+/*
+ * AAC encoder quantizer
+ * Copyright (C) 2015 Rostislav Pehlivanov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder quantizer
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#ifndef AVCODEC_AACENC_QUANTIZATION_H
+#define AVCODEC_AACENC_QUANTIZATION_H
+
+#include "aactab.h"
+#include "aacenc.h"
+#include "aacenctab.h"
+#include "aacenc_utils.h"
+
+/**
+ * Calculate rate distortion cost for quantizing with given codebook
+ *
+ * @return quantization distortion
+ */
+static av_always_inline float quantize_and_encode_band_cost_template(
+                                struct AACEncContext *s,
+                                PutBitContext *pb, const float *in, float *out,
+                                const float *scaled, int size, int scale_idx,
+                                int cb, const float lambda, const float uplim,
+                                int *bits, float *energy, int BT_ZERO, int BT_UNSIGNED,
+                                int BT_PAIR, int BT_ESC, int BT_NOISE, int BT_STEREO,
+                                const float ROUNDING)
+{
+    const int q_idx = POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512;
+    const float Q   = ff_aac_pow2sf_tab [q_idx];
+    const float Q34 = ff_aac_pow34sf_tab[q_idx];
+    const float IQ  = ff_aac_pow2sf_tab [POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
+    const float CLIPPED_ESCAPE = 165140.0f*IQ;
+    int i, j;
+    float cost = 0;
+    float qenergy = 0;
+    const int dim = BT_PAIR ? 2 : 4;
+    int resbits = 0;
+    int off;
+
+    if (BT_ZERO || BT_NOISE || BT_STEREO) {
+        for (i = 0; i < size; i++)
+            cost += in[i]*in[i];
+        if (bits)
+            *bits = 0;
+        if (energy)
+            *energy = qenergy;
+        if (out) {
+            for (i = 0; i < size; i += dim)
+                for (j = 0; j < dim; j++)
+                    out[i+j] = 0.0f;
+        }
+        return cost * lambda;
+    }
+    if (!scaled) {
+        s->abs_pow34(s->scoefs, in, size);
+        scaled = s->scoefs;
+    }
+    s->quant_bands(s->qcoefs, in, scaled, size, !BT_UNSIGNED, aac_cb_maxval[cb], Q34, ROUNDING);
+    if (BT_UNSIGNED) {
+        off = 0;
+    } else {
+        off = aac_cb_maxval[cb];
+    }
+    for (i = 0; i < size; i += dim) {
+        const float *vec;
+        int *quants = s->qcoefs + i;
+        int curidx = 0;
+        int curbits;
+        float quantized, rd = 0.0f;
+        for (j = 0; j < dim; j++) {
+            curidx *= aac_cb_range[cb];
+            curidx += quants[j] + off;
+        }
+        curbits =  ff_aac_spectral_bits[cb-1][curidx];
+        vec     = &ff_aac_codebook_vectors[cb-1][curidx*dim];
+        if (BT_UNSIGNED) {
+            for (j = 0; j < dim; j++) {
+                float t = fabsf(in[i+j]);
+                float di;
+                if (BT_ESC && vec[j] == 64.0f) { //FIXME: slow
+                    if (t >= CLIPPED_ESCAPE) {
+                        quantized = CLIPPED_ESCAPE;
+                        curbits += 21;
+                    } else {
+                        int c = av_clip_uintp2(quant(t, Q, ROUNDING), 13);
+                        quantized = c*cbrtf(c)*IQ;
+                        curbits += av_log2(c)*2 - 4 + 1;
+                    }
+                } else {
+                    quantized = vec[j]*IQ;
+                }
+                di = t - quantized;
+                if (out)
+                    out[i+j] = in[i+j] >= 0 ? quantized : -quantized;
+                if (vec[j] != 0.0f)
+                    curbits++;
+                qenergy += quantized*quantized;
+                rd += di*di;
+            }
+        } else {
+            for (j = 0; j < dim; j++) {
+                quantized = vec[j]*IQ;
+                qenergy += quantized*quantized;
+                if (out)
+                    out[i+j] = quantized;
+                rd += (in[i+j] - quantized)*(in[i+j] - quantized);
+            }
+        }
+        cost    += rd * lambda + curbits;
+        resbits += curbits;
+        if (cost >= uplim)
+            return uplim;
+        if (pb) {
+            put_bits(pb, ff_aac_spectral_bits[cb-1][curidx], ff_aac_spectral_codes[cb-1][curidx]);
+            if (BT_UNSIGNED)
+                for (j = 0; j < dim; j++)
+                    if (ff_aac_codebook_vectors[cb-1][curidx*dim+j] != 0.0f)
+                        put_bits(pb, 1, in[i+j] < 0.0f);
+            if (BT_ESC) {
+                for (j = 0; j < 2; j++) {
+                    if (ff_aac_codebook_vectors[cb-1][curidx*2+j] == 64.0f) {
+                        int coef = av_clip_uintp2(quant(fabsf(in[i+j]), Q, ROUNDING), 13);
+                        int len = av_log2(coef);
+
+                        put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2);
+                        put_sbits(pb, len, coef);
+                    }
+                }
+            }
+        }
+    }
+
+    if (bits)
+        *bits = resbits;
+    if (energy)
+        *energy = qenergy;
+    return cost;
+}
+
+static inline float quantize_and_encode_band_cost_NONE(struct AACEncContext *s, PutBitContext *pb,
+                                                const float *in, float *quant, const float *scaled,
+                                                int size, int scale_idx, int cb,
+                                                const float lambda, const float uplim,
+                                                int *bits, float *energy) {
+    av_assert0(0);
+    return 0.0f;
+}
+
+#define QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NAME, BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE, BT_STEREO, ROUNDING) \
+static float quantize_and_encode_band_cost_ ## NAME(                                         \
+                                struct AACEncContext *s,                                     \
+                                PutBitContext *pb, const float *in, float *quant,            \
+                                const float *scaled, int size, int scale_idx,                \
+                                int cb, const float lambda, const float uplim,               \
+                                int *bits, float *energy) {                                  \
+    return quantize_and_encode_band_cost_template(                                           \
+                                s, pb, in, quant, scaled, size, scale_idx,                   \
+                                BT_ESC ? ESC_BT : cb, lambda, uplim, bits, energy,           \
+                                BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE, BT_STEREO,  \
+                                ROUNDING);                                                   \
+}
+
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ZERO,  1, 0, 0, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SQUAD, 0, 0, 0, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UQUAD, 0, 1, 0, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SPAIR, 0, 0, 1, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UPAIR, 0, 1, 1, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC,   0, 1, 1, 1, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC_RTZ, 0, 1, 1, 1, 0, 0, ROUND_TO_ZERO)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NOISE, 0, 0, 0, 0, 1, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(STEREO,0, 0, 0, 0, 0, 1, ROUND_STANDARD)
+
+static float (*const quantize_and_encode_band_cost_arr[])(
+                                struct AACEncContext *s,
+                                PutBitContext *pb, const float *in, float *quant,
+                                const float *scaled, int size, int scale_idx,
+                                int cb, const float lambda, const float uplim,
+                                int *bits, float *energy) = {
+    quantize_and_encode_band_cost_ZERO,
+    quantize_and_encode_band_cost_SQUAD,
+    quantize_and_encode_band_cost_SQUAD,
+    quantize_and_encode_band_cost_UQUAD,
+    quantize_and_encode_band_cost_UQUAD,
+    quantize_and_encode_band_cost_SPAIR,
+    quantize_and_encode_band_cost_SPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_ESC,
+    quantize_and_encode_band_cost_NONE,     /* CB 12 doesn't exist */
+    quantize_and_encode_band_cost_NOISE,
+    quantize_and_encode_band_cost_STEREO,
+    quantize_and_encode_band_cost_STEREO,
+};
+
+static float (*const quantize_and_encode_band_cost_rtz_arr[])(
+                                struct AACEncContext *s,
+                                PutBitContext *pb, const float *in, float *quant,
+                                const float *scaled, int size, int scale_idx,
+                                int cb, const float lambda, const float uplim,
+                                int *bits, float *energy) = {
+    quantize_and_encode_band_cost_ZERO,
+    quantize_and_encode_band_cost_SQUAD,
+    quantize_and_encode_band_cost_SQUAD,
+    quantize_and_encode_band_cost_UQUAD,
+    quantize_and_encode_band_cost_UQUAD,
+    quantize_and_encode_band_cost_SPAIR,
+    quantize_and_encode_band_cost_SPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_ESC_RTZ,
+    quantize_and_encode_band_cost_NONE,     /* CB 12 doesn't exist */
+    quantize_and_encode_band_cost_NOISE,
+    quantize_and_encode_band_cost_STEREO,
+    quantize_and_encode_band_cost_STEREO,
+};
+
+#define quantize_and_encode_band_cost(                                  \
+                                s, pb, in, quant, scaled, size, scale_idx, cb, \
+                                lambda, uplim, bits, energy, rtz)               \
+    ((rtz) ? quantize_and_encode_band_cost_rtz_arr : quantize_and_encode_band_cost_arr)[cb]( \
+                                s, pb, in, quant, scaled, size, scale_idx, cb, \
+                                lambda, uplim, bits, energy)
+
+static inline float quantize_band_cost(struct AACEncContext *s, const float *in,
+                                const float *scaled, int size, int scale_idx,
+                                int cb, const float lambda, const float uplim,
+                                int *bits, float *energy, int rtz)
+{
+    return quantize_and_encode_band_cost(s, NULL, in, NULL, scaled, size, scale_idx,
+                                         cb, lambda, uplim, bits, energy, rtz);
+}
+
+static inline int quantize_band_cost_bits(struct AACEncContext *s, const float *in,
+                                const float *scaled, int size, int scale_idx,
+                                int cb, const float lambda, const float uplim,
+                                int *bits, float *energy, int rtz)
+{
+    int auxbits;
+    quantize_and_encode_band_cost(s, NULL, in, NULL, scaled, size, scale_idx,
+                                         cb, 0.0f, uplim, &auxbits, energy, rtz);
+    if (bits) {
+        *bits = auxbits;
+    }
+    return auxbits;
+}
+
+static inline void quantize_and_encode_band(struct AACEncContext *s, PutBitContext *pb,
+                                            const float *in, float *out, int size, int scale_idx,
+                                            int cb, const float lambda, int rtz)
+{
+    quantize_and_encode_band_cost(s, pb, in, out, NULL, size, scale_idx, cb, lambda,
+                                  INFINITY, NULL, NULL, rtz);
+}
+
+#include "aacenc_quantization_misc.h"
+
+#endif /* AVCODEC_AACENC_QUANTIZATION_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacenc_quantization_misc.h

@@ -0,0 +1,53 @@
+/*
+ * AAC encoder quantization
+ * Copyright (C) 2015 Claudio Freire
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder quantization misc reusable function templates
+ * @author Claudio Freire ( klaussfreire gmail com )
+ */
+
+#ifndef AVCODEC_AACENC_QUANTIZATION_MISC_H
+#define AVCODEC_AACENC_QUANTIZATION_MISC_H
+
+static inline float quantize_band_cost_cached(struct AACEncContext *s, int w, int g, const float *in,
+                                const float *scaled, int size, int scale_idx,
+                                int cb, const float lambda, const float uplim,
+                                int *bits, float *energy, int rtz)
+{
+    AACQuantizeBandCostCacheEntry *entry;
+    av_assert1(scale_idx >= 0 && scale_idx < 256);
+    entry = &s->quantize_band_cost_cache[scale_idx][w*16+g];
+    if (entry->generation != s->quantize_band_cost_cache_generation || entry->cb != cb || entry->rtz != rtz) {
+        entry->rd = quantize_band_cost(s, in, scaled, size, scale_idx,
+                                       cb, lambda, uplim, &entry->bits, &entry->energy, rtz);
+        entry->cb = cb;
+        entry->rtz = rtz;
+        entry->generation = s->quantize_band_cost_cache_generation;
+    }
+    if (bits)
+        *bits = entry->bits;
+    if (energy)
+        *energy = entry->energy;
+    return entry->rd;
+}
+
+#endif /* AVCODEC_AACENC_QUANTIZATION_MISC_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacenc_tns.c

@@ -0,0 +1,215 @@
+/*
+ * AAC encoder TNS
+ * Copyright (C) 2015 Rostislav Pehlivanov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder temporal noise shaping
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#include "libavutil/libm.h"
+#include "aacenc.h"
+#include "aacenc_tns.h"
+#include "aactab.h"
+#include "aacenc_utils.h"
+#include "aacenc_quantization.h"
+
+/* Could be set to 3 to save an additional bit at the cost of little quality */
+#define TNS_Q_BITS 4
+
+/* Coefficient resolution in short windows */
+#define TNS_Q_BITS_IS8 4
+
+/* We really need the bits we save here elsewhere */
+#define TNS_ENABLE_COEF_COMPRESSION
+
+/* TNS will only be used if the LPC gain is within these margins */
+#define TNS_GAIN_THRESHOLD_LOW      1.4f
+#define TNS_GAIN_THRESHOLD_HIGH     1.16f*TNS_GAIN_THRESHOLD_LOW
+
+static inline int compress_coeffs(int *coef, int order, int c_bits)
+{
+    int i;
+    const int low_idx   = c_bits ?  4 : 2;
+    const int shift_val = c_bits ?  8 : 4;
+    const int high_idx  = c_bits ? 11 : 5;
+#ifndef TNS_ENABLE_COEF_COMPRESSION
+    return 0;
+#endif /* TNS_ENABLE_COEF_COMPRESSION */
+    for (i = 0; i < order; i++)
+        if (coef[i] >= low_idx && coef[i] <= high_idx)
+            return 0;
+    for (i = 0; i < order; i++)
+        coef[i] -= (coef[i] > high_idx) ? shift_val : 0;
+    return 1;
+}
+
+/**
+ * Encode TNS data.
+ * Coefficient compression is simply not lossless as it should be
+ * on any decoder tested and as such is not active.
+ */
+void ff_aac_encode_tns_info(AACEncContext *s, SingleChannelElement *sce)
+{
+    TemporalNoiseShaping *tns = &sce->tns;
+    int i, w, filt, coef_compress = 0, coef_len;
+    const int is8 = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
+    const int c_bits = is8 ? TNS_Q_BITS_IS8 == 4 : TNS_Q_BITS == 4;
+
+    if (!sce->tns.present)
+        return;
+
+    for (i = 0; i < sce->ics.num_windows; i++) {
+        put_bits(&s->pb, 2 - is8, sce->tns.n_filt[i]);
+        if (!tns->n_filt[i])
+            continue;
+        put_bits(&s->pb, 1, c_bits);
+        for (filt = 0; filt < tns->n_filt[i]; filt++) {
+            put_bits(&s->pb, 6 - 2 * is8, tns->length[i][filt]);
+            put_bits(&s->pb, 5 - 2 * is8, tns->order[i][filt]);
+            if (!tns->order[i][filt])
+                continue;
+            put_bits(&s->pb, 1, tns->direction[i][filt]);
+            coef_compress = compress_coeffs(tns->coef_idx[i][filt],
+                                            tns->order[i][filt], c_bits);
+            put_bits(&s->pb, 1, coef_compress);
+            coef_len = c_bits + 3 - coef_compress;
+            for (w = 0; w < tns->order[i][filt]; w++)
+                put_bits(&s->pb, coef_len, tns->coef_idx[i][filt][w]);
+        }
+    }
+}
+
+/* Apply TNS filter */
+void ff_aac_apply_tns(AACEncContext *s, SingleChannelElement *sce)
+{
+    TemporalNoiseShaping *tns = &sce->tns;
+    IndividualChannelStream *ics = &sce->ics;
+    int w, filt, m, i, top, order, bottom, start, end, size, inc;
+    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
+    float lpc[TNS_MAX_ORDER];
+
+    for (w = 0; w < ics->num_windows; w++) {
+        bottom = ics->num_swb;
+        for (filt = 0; filt < tns->n_filt[w]; filt++) {
+            top    = bottom;
+            bottom = FFMAX(0, top - tns->length[w][filt]);
+            order  = tns->order[w][filt];
+            if (order == 0)
+                continue;
+
+            // tns_decode_coef
+            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
+
+            start = ics->swb_offset[FFMIN(bottom, mmm)];
+            end   = ics->swb_offset[FFMIN(   top, mmm)];
+            if ((size = end - start) <= 0)
+                continue;
+            if (tns->direction[w][filt]) {
+                inc = -1;
+                start = end - 1;
+            } else {
+                inc = 1;
+            }
+            start += w * 128;
+
+            /* AR filter */
+            for (m = 0; m < size; m++, start += inc) {
+                for (i = 1; i <= FFMIN(m, order); i++) {
+                    sce->coeffs[start] += lpc[i-1]*sce->pcoeffs[start - i*inc];
+                }
+            }
+        }
+    }
+}
+
+/*
+ * c_bits - 1 if 4 bit coefficients, 0 if 3 bit coefficients
+ */
+static inline void quantize_coefs(double *coef, int *idx, float *lpc, int order,
+                                  int c_bits)
+{
+    int i;
+    const float *quant_arr = tns_tmp2_map[c_bits];
+    for (i = 0; i < order; i++) {
+        idx[i] = quant_array_idx(coef[i], quant_arr, c_bits ? 16 : 8);
+        lpc[i] = quant_arr[idx[i]];
+    }
+}
+
+/*
+ * 3 bits per coefficient with 8 short windows
+ */
+void ff_aac_search_for_tns(AACEncContext *s, SingleChannelElement *sce)
+{
+    TemporalNoiseShaping *tns = &sce->tns;
+    int w, g, count = 0;
+    double gain, coefs[MAX_LPC_ORDER];
+    const int mmm = FFMIN(sce->ics.tns_max_bands, sce->ics.max_sfb);
+    const int is8 = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
+    const int c_bits = is8 ? TNS_Q_BITS_IS8 == 4 : TNS_Q_BITS == 4;
+    const int sfb_start = av_clip(tns_min_sfb[is8][s->samplerate_index], 0, mmm);
+    const int sfb_end   = av_clip(sce->ics.num_swb, 0, mmm);
+    const int order = is8 ? 7 : s->profile == FF_PROFILE_AAC_LOW ? 12 : TNS_MAX_ORDER;
+    const int slant = sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE  ? 1 :
+                      sce->ics.window_sequence[0] == LONG_START_SEQUENCE ? 0 : 2;
+    const int sfb_len = sfb_end - sfb_start;
+    const int coef_len = sce->ics.swb_offset[sfb_end] - sce->ics.swb_offset[sfb_start];
+
+    if (coef_len <= 0 || sfb_len <= 0) {
+        sce->tns.present = 0;
+        return;
+    }
+
+    for (w = 0; w < sce->ics.num_windows; w++) {
+        float en[2] = {0.0f, 0.0f};
+        int oc_start = 0, os_start = 0;
+        int coef_start = sce->ics.swb_offset[sfb_start];
+
+        for (g = sfb_start; g < sce->ics.num_swb && g <= sfb_end; g++) {
+            FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[w*16+g];
+            if (g > sfb_start + (sfb_len/2))
+                en[1] += band->energy;
+            else
+                en[0] += band->energy;
+        }
+
+        /* LPC */
+        gain = ff_lpc_calc_ref_coefs_f(&s->lpc, &sce->coeffs[w*128 + coef_start],
+                                       coef_len, order, coefs);
+
+        if (!order || !isfinite(gain) || gain < TNS_GAIN_THRESHOLD_LOW || gain > TNS_GAIN_THRESHOLD_HIGH)
+            continue;
+
+        tns->n_filt[w] = is8 ? 1 : order != TNS_MAX_ORDER ? 2 : 3;
+        for (g = 0; g < tns->n_filt[w]; g++) {
+            tns->direction[w][g] = slant != 2 ? slant : en[g] < en[!g];
+            tns->order[w][g] = g < tns->n_filt[w] ? order/tns->n_filt[w] : order - oc_start;
+            tns->length[w][g] = g < tns->n_filt[w] ? sfb_len/tns->n_filt[w] : sfb_len - os_start;
+            quantize_coefs(&coefs[oc_start], tns->coef_idx[w][g], tns->coef[w][g],
+                            tns->order[w][g], c_bits);
+            oc_start += tns->order[w][g];
+            os_start += tns->length[w][g];
+        }
+        count++;
+    }
+    sce->tns.present = !!count;
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacenc_tns.h

@@ -0,0 +1,37 @@
+/*
+ * AAC encoder TNS
+ * Copyright (C) 2015 Rostislav Pehlivanov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder temporal noise shaping
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#ifndef AVCODEC_AACENC_TNS_H
+#define AVCODEC_AACENC_TNS_H
+
+#include "aacenc.h"
+
+void ff_aac_encode_tns_info(AACEncContext *s, SingleChannelElement *sce);
+void ff_aac_apply_tns(AACEncContext *s, SingleChannelElement *sce);
+void ff_aac_search_for_tns(AACEncContext *s, SingleChannelElement *sce);
+
+#endif /* AVCODEC_AACENC_TNS_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacenc_utils.h

@@ -0,0 +1,279 @@
+/*
+ * AAC encoder utilities
+ * Copyright (C) 2015 Rostislav Pehlivanov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder utilities
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#ifndef AVCODEC_AACENC_UTILS_H
+#define AVCODEC_AACENC_UTILS_H
+
+#include "libavutil/ffmath.h"
+#include "aac.h"
+#include "aacenctab.h"
+#include "aactab.h"
+
+#define ROUND_STANDARD 0.4054f
+#define ROUND_TO_ZERO 0.1054f
+#define C_QUANT 0.4054f
+
+static inline void abs_pow34_v(float *out, const float *in, const int size)
+{
+    int i;
+    for (i = 0; i < size; i++) {
+        float a = fabsf(in[i]);
+        out[i] = sqrtf(a * sqrtf(a));
+    }
+}
+
+static inline float pos_pow34(float a)
+{
+    return sqrtf(a * sqrtf(a));
+}
+
+/**
+ * Quantize one coefficient.
+ * @return absolute value of the quantized coefficient
+ * @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
+ */
+static inline int quant(float coef, const float Q, const float rounding)
+{
+    float a = coef * Q;
+    return sqrtf(a * sqrtf(a)) + rounding;
+}
+
+static inline void quantize_bands(int *out, const float *in, const float *scaled,
+                                  int size, int is_signed, int maxval, const float Q34,
+                                  const float rounding)
+{
+    int i;
+    for (i = 0; i < size; i++) {
+        float qc = scaled[i] * Q34;
+        int tmp = (int)FFMIN(qc + rounding, (float)maxval);
+        if (is_signed && in[i] < 0.0f) {
+            tmp = -tmp;
+        }
+        out[i] = tmp;
+    }
+}
+
+static inline float find_max_val(int group_len, int swb_size, const float *scaled)
+{
+    float maxval = 0.0f;
+    int w2, i;
+    for (w2 = 0; w2 < group_len; w2++) {
+        for (i = 0; i < swb_size; i++) {
+            maxval = FFMAX(maxval, scaled[w2*128+i]);
+        }
+    }
+    return maxval;
+}
+
+static inline int find_min_book(float maxval, int sf)
+{
+    float Q34 = ff_aac_pow34sf_tab[POW_SF2_ZERO - sf + SCALE_ONE_POS - SCALE_DIV_512];
+    int qmaxval, cb;
+    qmaxval = maxval * Q34 + C_QUANT;
+    if (qmaxval >= (FF_ARRAY_ELEMS(aac_maxval_cb)))
+        cb = 11;
+    else
+        cb = aac_maxval_cb[qmaxval];
+    return cb;
+}
+
+static inline float find_form_factor(int group_len, int swb_size, float thresh,
+                                     const float *scaled, float nzslope) {
+    const float iswb_size = 1.0f / swb_size;
+    const float iswb_sizem1 = 1.0f / (swb_size - 1);
+    const float ethresh = thresh;
+    float form = 0.0f, weight = 0.0f;
+    int w2, i;
+    for (w2 = 0; w2 < group_len; w2++) {
+        float e = 0.0f, e2 = 0.0f, var = 0.0f, maxval = 0.0f;
+        float nzl = 0;
+        for (i = 0; i < swb_size; i++) {
+            float s = fabsf(scaled[w2*128+i]);
+            maxval = FFMAX(maxval, s);
+            e += s;
+            e2 += s *= s;
+            /* We really don't want a hard non-zero-line count, since
+             * even below-threshold lines do add up towards band spectral power.
+             * So, fall steeply towards zero, but smoothly
+             */
+            if (s >= ethresh) {
+                nzl += 1.0f;
+            } else {
+                if (nzslope == 2.f)
+                    nzl += (s / ethresh) * (s / ethresh);
+                else
+                    nzl += ff_fast_powf(s / ethresh, nzslope);
+            }
+        }
+        if (e2 > thresh) {
+            float frm;
+            e *= iswb_size;
+
+            /** compute variance */
+            for (i = 0; i < swb_size; i++) {
+                float d = fabsf(scaled[w2*128+i]) - e;
+                var += d*d;
+            }
+            var = sqrtf(var * iswb_sizem1);
+
+            e2 *= iswb_size;
+            frm = e / FFMIN(e+4*var,maxval);
+            form += e2 * sqrtf(frm) / FFMAX(0.5f,nzl);
+            weight += e2;
+        }
+    }
+    if (weight > 0) {
+        return form / weight;
+    } else {
+        return 1.0f;
+    }
+}
+
+/** Return the minimum scalefactor where the quantized coef does not clip. */
+static inline uint8_t coef2minsf(float coef)
+{
+    return av_clip_uint8(log2f(coef)*4 - 69 + SCALE_ONE_POS - SCALE_DIV_512);
+}
+
+/** Return the maximum scalefactor where the quantized coef is not zero. */
+static inline uint8_t coef2maxsf(float coef)
+{
+    return av_clip_uint8(log2f(coef)*4 +  6 + SCALE_ONE_POS - SCALE_DIV_512);
+}
+
+/*
+ * Returns the closest possible index to an array of float values, given a value.
+ */
+static inline int quant_array_idx(const float val, const float *arr, const int num)
+{
+    int i, index = 0;
+    float quant_min_err = INFINITY;
+    for (i = 0; i < num; i++) {
+        float error = (val - arr[i])*(val - arr[i]);
+        if (error < quant_min_err) {
+            quant_min_err = error;
+            index = i;
+        }
+    }
+    return index;
+}
+
+/**
+ * approximates exp10f(-3.0f*(0.5f + 0.5f * cosf(FFMIN(b,15.5f) / 15.5f)))
+ */
+static av_always_inline float bval2bmax(float b)
+{
+    return 0.001f + 0.0035f * (b*b*b) / (15.5f*15.5f*15.5f);
+}
+
+/*
+ * Compute a nextband map to be used with SF delta constraint utilities.
+ * The nextband array should contain 128 elements, and positions that don't
+ * map to valid, nonzero bands of the form w*16+g (with w being the initial
+ * window of the window group, only) are left indetermined.
+ */
+static inline void ff_init_nextband_map(const SingleChannelElement *sce, uint8_t *nextband)
+{
+    unsigned char prevband = 0;
+    int w, g;
+    /** Just a safe default */
+    for (g = 0; g < 128; g++)
+        nextband[g] = g;
+
+    /** Now really navigate the nonzero band chain */
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        for (g = 0; g < sce->ics.num_swb; g++) {
+            if (!sce->zeroes[w*16+g] && sce->band_type[w*16+g] < RESERVED_BT)
+                prevband = nextband[prevband] = w*16+g;
+        }
+    }
+    nextband[prevband] = prevband; /* terminate */
+}
+
+/*
+ * Updates nextband to reflect a removed band (equivalent to
+ * calling ff_init_nextband_map after marking a band as zero)
+ */
+static inline void ff_nextband_remove(uint8_t *nextband, int prevband, int band)
+{
+    nextband[prevband] = nextband[band];
+}
+
+/*
+ * Checks whether the specified band could be removed without inducing
+ * scalefactor delta that violates SF delta encoding constraints.
+ * prev_sf has to be the scalefactor of the previous nonzero, nonspecial
+ * band, in encoding order, or negative if there was no such band.
+ */
+static inline int ff_sfdelta_can_remove_band(const SingleChannelElement *sce,
+    const uint8_t *nextband, int prev_sf, int band)
+{
+    return prev_sf >= 0
+        && sce->sf_idx[nextband[band]] >= (prev_sf - SCALE_MAX_DIFF)
+        && sce->sf_idx[nextband[band]] <= (prev_sf + SCALE_MAX_DIFF);
+}
+
+/*
+ * Checks whether the specified band's scalefactor could be replaced
+ * with another one without violating SF delta encoding constraints.
+ * prev_sf has to be the scalefactor of the previous nonzero, nonsepcial
+ * band, in encoding order, or negative if there was no such band.
+ */
+static inline int ff_sfdelta_can_replace(const SingleChannelElement *sce,
+    const uint8_t *nextband, int prev_sf, int new_sf, int band)
+{
+    return new_sf >= (prev_sf - SCALE_MAX_DIFF)
+        && new_sf <= (prev_sf + SCALE_MAX_DIFF)
+        && sce->sf_idx[nextband[band]] >= (new_sf - SCALE_MAX_DIFF)
+        && sce->sf_idx[nextband[band]] <= (new_sf + SCALE_MAX_DIFF);
+}
+
+/**
+ * linear congruential pseudorandom number generator
+ *
+ * @param   previous_val    pointer to the current state of the generator
+ *
+ * @return  Returns a 32-bit pseudorandom integer
+ */
+static av_always_inline int lcg_random(unsigned previous_val)
+{
+    union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
+    return v.s;
+}
+
+#define ERROR_IF(cond, ...) \
+    if (cond) { \
+        av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
+        return AVERROR(EINVAL); \
+    }
+
+#define WARN_IF(cond, ...) \
+    if (cond) { \
+        av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \
+    }
+
+#endif /* AVCODEC_AACENC_UTILS_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacencdsp.asm

@@ -0,0 +1,86 @@
+;******************************************************************************
+;* SIMD optimized AAC encoder DSP functions
+;*
+;* Copyright (C) 2016 Rostislav Pehlivanov <atomnuker@gmail.com>
+;*
+;* This file is part of FFmpeg.
+;*
+;* FFmpeg is free software; you can redistribute it and/or
+;* modify it under the terms of the GNU Lesser General Public
+;* License as published by the Free Software Foundation; either
+;* version 2.1 of the License, or (at your option) any later version.
+;*
+;* FFmpeg is distributed in the hope that it will be useful,
+;* but WITHOUT ANY WARRANTY; without even the implied warranty of
+;* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+;* Lesser General Public License for more details.
+;*
+;* You should have received a copy of the GNU Lesser General Public
+;* License along with FFmpeg; if not, write to the Free Software
+;* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+;******************************************************************************
+
+%include "libavutil/x86/x86util.asm"
+
+SECTION_RODATA
+
+float_abs_mask: times 4 dd 0x7fffffff
+
+SECTION .text
+
+;*******************************************************************
+;void ff_abs_pow34(float *out, const float *in, const int size);
+;*******************************************************************
+INIT_XMM sse
+cglobal abs_pow34, 3, 3, 3, out, in, size
+    mova   m2, [float_abs_mask]
+    shl    sizeq, 2
+    add    inq, sizeq
+    add    outq, sizeq
+    neg    sizeq
+.loop:
+    andps  m0, m2, [inq+sizeq]
+    sqrtps m1, m0
+    mulps  m0, m1
+    sqrtps m0, m0
+    mova   [outq+sizeq], m0
+    add    sizeq, mmsize
+    jl    .loop
+    RET
+
+;*******************************************************************
+;void ff_aac_quantize_bands(int *out, const float *in, const float *scaled,
+;                           int size, int is_signed, int maxval, const float Q34,
+;                           const float rounding)
+;*******************************************************************
+INIT_XMM sse2
+cglobal aac_quantize_bands, 5, 5, 6, out, in, scaled, size, is_signed, maxval, Q34, rounding
+%if UNIX64 == 0
+    movss     m0, Q34m
+    movss     m1, roundingm
+    cvtsi2ss  m3, dword maxvalm
+%else
+    cvtsi2ss  m3, maxvald
+%endif
+    shufps    m0, m0, 0
+    shufps    m1, m1, 0
+    shufps    m3, m3, 0
+    shl       is_signedd, 31
+    movd      m4, is_signedd
+    shufps    m4, m4, 0
+    shl       sized,   2
+    add       inq, sizeq
+    add       outq, sizeq
+    add       scaledq, sizeq
+    neg       sizeq
+.loop:
+    mulps     m2, m0, [scaledq+sizeq]
+    addps     m2, m1
+    minps     m2, m3
+    andps     m5, m4, [inq+sizeq]
+    orps      m2, m5
+    cvttps2dq m2, m2
+    mova      [outq+sizeq], m2
+    add       sizeq, mmsize
+    jl       .loop
+    RET

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacencdsp_init.c

@@ -0,0 +1,43 @@
+/*
+ * AAC encoder assembly optimizations
+ * Copyright (C) 2016 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "config.h"
+
+#include "libavutil/float_dsp.h"
+#include "libavutil/x86/cpu.h"
+#include "libavcodec/aacenc.h"
+
+void ff_abs_pow34_sse(float *out, const float *in, const int size);
+
+void ff_aac_quantize_bands_sse2(int *out, const float *in, const float *scaled,
+                                int size, int is_signed, int maxval, const float Q34,
+                                const float rounding);
+
+av_cold void ff_aac_dsp_init_x86(AACEncContext *s)
+{
+    int cpu_flags = av_get_cpu_flags();
+
+    if (EXTERNAL_SSE(cpu_flags))
+        s->abs_pow34   = ff_abs_pow34_sse;
+
+    if (EXTERNAL_SSE2(cpu_flags))
+        s->quant_bands = ff_aac_quantize_bands_sse2;
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacenctab.c

@@ -0,0 +1,108 @@
+/*
+ * AAC encoder data
+ * Copyright (c) 2015 Rostislav Pehlivanov ( atomnuker gmail com )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "aacenctab.h"
+
+static const uint8_t swb_size_128_96[] = {
+    4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
+};
+
+static const uint8_t swb_size_128_64[] = {
+    4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
+};
+
+static const uint8_t swb_size_128_48[] = {
+    4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
+};
+
+static const uint8_t swb_size_128_24[] = {
+    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
+};
+
+static const uint8_t swb_size_128_16[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
+};
+
+static const uint8_t swb_size_128_8[] = {
+    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
+};
+
+static const uint8_t swb_size_1024_96[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
+    64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
+};
+
+static const uint8_t swb_size_1024_64[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
+    12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
+    40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
+};
+
+static const uint8_t swb_size_1024_48[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
+    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
+    96
+};
+
+static const uint8_t swb_size_1024_32[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
+    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
+};
+
+static const uint8_t swb_size_1024_24[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
+    32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
+};
+
+static const uint8_t swb_size_1024_16[] = {
+    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
+    32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
+};
+
+static const uint8_t swb_size_1024_8[] = {
+    12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
+    16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
+    32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
+};
+
+const uint8_t *ff_aac_swb_size_128[] = {
+    swb_size_128_96, swb_size_128_96, swb_size_128_64,
+    swb_size_128_48, swb_size_128_48, swb_size_128_48,
+    swb_size_128_24, swb_size_128_24, swb_size_128_16,
+    swb_size_128_16, swb_size_128_16, swb_size_128_8,
+    swb_size_128_8
+};
+
+const uint8_t *ff_aac_swb_size_1024[] = {
+    swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
+    swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
+    swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
+    swb_size_1024_16, swb_size_1024_16, swb_size_1024_8,
+    swb_size_1024_8
+};
+
+const int ff_aac_swb_size_128_len  = FF_ARRAY_ELEMS(ff_aac_swb_size_128);
+const int ff_aac_swb_size_1024_len = FF_ARRAY_ELEMS(ff_aac_swb_size_1024);

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacenctab.h

@@ -0,0 +1,139 @@
+/*
+ * AAC encoder data
+ * Copyright (c) 2015 Rostislav Pehlivanov ( atomnuker gmail com )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder data
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#ifndef AVCODEC_AACENCTAB_H
+#define AVCODEC_AACENCTAB_H
+
+#include "aac.h"
+
+/** Total number of usable codebooks **/
+#define CB_TOT 12
+
+/** Total number of codebooks, including special ones **/
+#define CB_TOT_ALL 15
+
+#define AAC_MAX_CHANNELS 16
+
+extern const uint8_t *ff_aac_swb_size_1024[];
+extern const int      ff_aac_swb_size_1024_len;
+extern const uint8_t *ff_aac_swb_size_128[];
+extern const int      ff_aac_swb_size_128_len;
+
+/* Supported layouts without using a PCE */
+static const int64_t aac_normal_chan_layouts[7] = {
+    AV_CH_LAYOUT_MONO,
+    AV_CH_LAYOUT_STEREO,
+    AV_CH_LAYOUT_SURROUND,
+    AV_CH_LAYOUT_4POINT0,
+    AV_CH_LAYOUT_5POINT0_BACK,
+    AV_CH_LAYOUT_5POINT1_BACK,
+    AV_CH_LAYOUT_7POINT1,
+};
+
+/** default channel configurations */
+static const uint8_t aac_chan_configs[AAC_MAX_CHANNELS][6] = {
+    {1, TYPE_SCE},                                         // 1 channel  - single channel element
+    {1, TYPE_CPE},                                         // 2 channels - channel pair
+    {2, TYPE_SCE, TYPE_CPE},                               // 3 channels - center + stereo
+    {3, TYPE_SCE, TYPE_CPE, TYPE_SCE},                     // 4 channels - front center + stereo + back center
+    {3, TYPE_SCE, TYPE_CPE, TYPE_CPE},                     // 5 channels - front center + stereo + back stereo
+    {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE},           // 6 channels - front center + stereo + back stereo + LFE
+    {0},                                                   // 7 channels - invalid without PCE
+    {5, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 8 channels - front center + front stereo + side stereo + back stereo + LFE
+};
+
+/**
+ * Table to remap channels from libavcodec's default order to AAC order.
+ */
+static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
+    { 0 },
+    { 0, 1 },
+    { 2, 0, 1 },
+    { 2, 0, 1, 3 },
+    { 2, 0, 1, 3, 4 },
+    { 2, 0, 1, 4, 5, 3 },
+    { 0 },
+    { 2, 0, 1, 6, 7, 4, 5, 3 },
+};
+
+/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
+ * failures */
+static const int mpeg4audio_sample_rates[16] = {
+    96000, 88200, 64000, 48000, 44100, 32000,
+    24000, 22050, 16000, 12000, 11025, 8000, 7350
+};
+
+/** bits needed to code codebook run value for long windows */
+static const uint8_t run_value_bits_long[64] = {
+     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,
+     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5, 10,
+    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
+    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
+};
+
+/** bits needed to code codebook run value for short windows */
+static const uint8_t run_value_bits_short[16] = {
+    3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
+};
+
+/* TNS starting SFBs for long and short windows */
+static const uint8_t tns_min_sfb_short[16] = {
+    2, 2, 2, 3, 3, 4, 6, 6, 8, 10, 10, 12, 12, 12, 12, 12
+};
+
+static const uint8_t tns_min_sfb_long[16] = {
+    12, 13, 15, 16, 17, 20, 25, 26, 24, 28, 30, 31, 31, 31, 31, 31
+};
+
+static const uint8_t * const tns_min_sfb[2] = {
+    tns_min_sfb_long, tns_min_sfb_short
+};
+
+static const uint8_t * const run_value_bits[2] = {
+    run_value_bits_long, run_value_bits_short
+};
+
+/** Map to convert values from BandCodingPath index to a codebook index **/
+static const uint8_t aac_cb_out_map[CB_TOT_ALL]  = {0,1,2,3,4,5,6,7,8,9,10,11,13,14,15};
+/** Inverse map to convert from codebooks to BandCodingPath indices **/
+static const uint8_t aac_cb_in_map[CB_TOT_ALL+1] = {0,1,2,3,4,5,6,7,8,9,10,11,0,12,13,14};
+
+static const uint8_t aac_cb_range [12] = {0, 3, 3, 3, 3, 9, 9, 8, 8, 13, 13, 17};
+static const uint8_t aac_cb_maxval[12] = {0, 1, 1, 2, 2, 4, 4, 7, 7, 12, 12, 16};
+
+static const unsigned char aac_maxval_cb[] = {
+    0, 1, 3, 5, 5, 7, 7, 7, 9, 9, 9, 9, 9, 11
+};
+
+static const int aacenc_profiles[] = {
+    FF_PROFILE_AAC_MAIN,
+    FF_PROFILE_AAC_LOW,
+    FF_PROFILE_AAC_LTP,
+    FF_PROFILE_MPEG2_AAC_LOW,
+};
+
+#endif /* AVCODEC_AACENCTAB_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacps.h

@@ -0,0 +1,86 @@
+/*
+ * MPEG-4 Parametric Stereo definitions and declarations
+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AACPS_H
+#define AVCODEC_AACPS_H
+
+#include <stdint.h>
+
+#include "aacpsdsp.h"
+#include "avcodec.h"
+#include "get_bits.h"
+
+#define PS_MAX_NUM_ENV 5
+#define PS_MAX_NR_IIDICC 34
+#define PS_MAX_NR_IPDOPD 17
+#define PS_MAX_SSB 91
+#define PS_MAX_AP_BANDS 50
+#define PS_QMF_TIME_SLOTS 32
+#define PS_MAX_DELAY 14
+#define PS_AP_LINKS 3
+#define PS_MAX_AP_DELAY 5
+
+typedef struct PSContext {
+    int    start;
+    int    enable_iid;
+    int    iid_quant;
+    int    nr_iid_par;
+    int    nr_ipdopd_par;
+    int    enable_icc;
+    int    icc_mode;
+    int    nr_icc_par;
+    int    enable_ext;
+    int    frame_class;
+    int    num_env_old;
+    int    num_env;
+    int    enable_ipdopd;
+    int    border_position[PS_MAX_NUM_ENV+1];
+    int8_t iid_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; ///< Inter-channel Intensity Difference Parameters
+    int8_t icc_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; ///< Inter-Channel Coherence Parameters
+    /* ipd/opd is iid/icc sized so that the same functions can handle both */
+    int8_t ipd_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; ///< Inter-channel Phase Difference Parameters
+    int8_t opd_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; ///< Overall Phase Difference Parameters
+    int    is34bands;
+    int    is34bands_old;
+
+    DECLARE_ALIGNED(16, INTFLOAT, in_buf)[5][44][2];
+    DECLARE_ALIGNED(16, INTFLOAT, delay)[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
+    DECLARE_ALIGNED(16, INTFLOAT, ap_delay)[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
+    DECLARE_ALIGNED(16, INTFLOAT, peak_decay_nrg)[34];
+    DECLARE_ALIGNED(16, INTFLOAT, power_smooth)[34];
+    DECLARE_ALIGNED(16, INTFLOAT, peak_decay_diff_smooth)[34];
+    DECLARE_ALIGNED(16, INTFLOAT, H11)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    DECLARE_ALIGNED(16, INTFLOAT, H12)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    DECLARE_ALIGNED(16, INTFLOAT, H21)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    DECLARE_ALIGNED(16, INTFLOAT, H22)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    DECLARE_ALIGNED(16, INTFLOAT, Lbuf)[91][32][2];
+    DECLARE_ALIGNED(16, INTFLOAT, Rbuf)[91][32][2];
+    int8_t opd_hist[PS_MAX_NR_IIDICC];
+    int8_t ipd_hist[PS_MAX_NR_IIDICC];
+    PSDSPContext dsp;
+} PSContext;
+
+void AAC_RENAME(ff_ps_init)(void);
+void AAC_RENAME(ff_ps_ctx_init)(PSContext *ps);
+int AAC_RENAME(ff_ps_read_data)(AVCodecContext *avctx, GetBitContext *gb, PSContext *ps, int bits_left);
+int AAC_RENAME(ff_ps_apply)(AVCodecContext *avctx, PSContext *ps, INTFLOAT L[2][38][64], INTFLOAT R[2][38][64], int top);
+
+#endif /* AVCODEC_AACPS_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacps_fixed.c

@@ -0,0 +1,24 @@
+/*
+ * MPEG-4 Parametric Stereo decoding functions
+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#define USE_FIXED 1
+
+#include "aacps.c"

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacps_fixed_tablegen.c

@@ -0,0 +1,24 @@
+/*
+ * Generate a header file for hardcoded Parametric Stereo tables
+ *
+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#define USE_FIXED 1
+#include "aacps_tablegen_template.c"

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacps_fixed_tablegen.h

@@ -0,0 +1,403 @@
+/*
+ * Header file for hardcoded Parametric Stereo tables
+ *
+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Note: Rounding-to-nearest used unless otherwise stated
+ *
+ */
+
+#ifndef AVCODEC_AACPS_FIXED_TABLEGEN_H
+#define AVCODEC_AACPS_FIXED_TABLEGEN_H
+
+#include <math.h>
+#include <stdint.h>
+
+#if CONFIG_HARDCODED_TABLES
+#define ps_tableinit()
+#define TABLE_CONST const
+#include "libavcodec/aacps_fixed_tables.h"
+#else
+#include "libavutil/common.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/mem.h"
+
+#include "aac_defines.h"
+#include "libavutil/softfloat.h"
+#define NR_ALLPASS_BANDS20 30
+#define NR_ALLPASS_BANDS34 50
+#define PS_AP_LINKS 3
+#define TABLE_CONST
+static int pd_re_smooth[8*8*8];
+static int pd_im_smooth[8*8*8];
+static int HA[46][8][4];
+static int HB[46][8][4];
+static DECLARE_ALIGNED(16, int, f20_0_8) [ 8][8][2];
+static DECLARE_ALIGNED(16, int, f34_0_12)[12][8][2];
+static DECLARE_ALIGNED(16, int, f34_1_8) [ 8][8][2];
+static DECLARE_ALIGNED(16, int, f34_2_4) [ 4][8][2];
+static TABLE_CONST DECLARE_ALIGNED(16, int, Q_fract_allpass)[2][50][3][2];
+static DECLARE_ALIGNED(16, int, phi_fract)[2][50][2];
+
+static const int g0_Q8[] = {
+    Q31(0.00746082949812f), Q31(0.02270420949825f), Q31(0.04546865930473f), Q31(0.07266113929591f),
+    Q31(0.09885108575264f), Q31(0.11793710567217f), Q31(0.125f)
+};
+
+static const int g0_Q12[] = {
+    Q31(0.04081179924692f), Q31(0.03812810994926f), Q31(0.05144908135699f), Q31(0.06399831151592f),
+    Q31(0.07428313801106f), Q31(0.08100347892914f), Q31(0.08333333333333f)
+};
+
+static const int g1_Q8[] = {
+    Q31(0.01565675600122f), Q31(0.03752716391991f), Q31(0.05417891378782f), Q31(0.08417044116767f),
+    Q31(0.10307344158036f), Q31(0.12222452249753f), Q31(0.125f)
+};
+
+static const int g2_Q4[] = {
+    Q31(-0.05908211155639f), Q31(-0.04871498374946f), Q31(0.0f),   Q31(0.07778723915851f),
+    Q31( 0.16486303567403f), Q31( 0.23279856662996f), Q31(0.25f)
+};
+
+static const int sintbl_4[4]   = {           0,  1073741824,           0, -1073741824 };
+static const int costbl_4[4]   = {  1073741824,           0, -1073741824,           0 };
+static const int sintbl_8[8]   = {           0,   759250125,  1073741824,   759250125,
+                                             0,  -759250125, -1073741824,  -759250125 };
+static const int costbl_8[8]   = {  1073741824,   759250125,           0,  -759250125,
+                                   -1073741824,  -759250125,           0,   759250125 };
+static const int sintbl_12[12] = {           0,   536870912,   929887697,  1073741824,
+                                     929887697,   536870912,           0,  -536870912,
+                                    -929887697, -1073741824,  -929887697,  -536870912 };
+static const int costbl_12[12] = {  1073741824,   929887697,   536870912,           0,
+                                    -536870912,  -929887697, -1073741824,  -929887697,
+                                    -536870912,           0,   536870912,   929887697 };
+
+static void make_filters_from_proto(int (*filter)[8][2], const int *proto, int bands)
+{
+
+    const int *sinptr, *cosptr;
+    int s, c, sinhalf, coshalf;
+    int q, n;
+
+    if (bands == 4) {
+        sinptr = sintbl_4;
+        cosptr = costbl_4;
+        sinhalf = 759250125;
+        coshalf = 759250125;
+    } else if (bands == 8) {
+        sinptr = sintbl_8;
+        cosptr = costbl_8;
+        sinhalf = 410903207;
+        coshalf = 992008094;
+    } else {
+        sinptr = sintbl_12;
+        cosptr = costbl_12;
+        sinhalf = 277904834;
+        coshalf = 1037154959;
+    }
+
+    for (q = 0; q < bands; q++) {
+        for (n = 0; n < 7; n++) {
+            int theta = (q*(n-6) + (n>>1) - 3) % bands;
+
+            if (theta < 0)
+                theta += bands;
+            s = sinptr[theta];
+            c = cosptr[theta];
+
+            if (n & 1) {
+                theta = (int)(((int64_t)c * coshalf - (int64_t)s * sinhalf + 0x20000000) >> 30);
+                s = (int)(((int64_t)s * coshalf + (int64_t)c * sinhalf + 0x20000000) >> 30);
+                c = theta;
+            }
+            filter[q][n][0] = (int)(((int64_t)proto[n] * c + 0x20000000) >> 30);
+            filter[q][n][1] = -(int)(((int64_t)proto[n] * s + 0x20000000) >> 30);
+        }
+    }
+}
+
+static void ps_tableinit(void)
+{
+    static const int ipdopd_sin[] = { Q30(0), Q30(M_SQRT1_2), Q30(1), Q30( M_SQRT1_2), Q30( 0), Q30(-M_SQRT1_2), Q30(-1), Q30(-M_SQRT1_2) };
+    static const int ipdopd_cos[] = { Q30(1), Q30(M_SQRT1_2), Q30(0), Q30(-M_SQRT1_2), Q30(-1), Q30(-M_SQRT1_2), Q30( 0), Q30( M_SQRT1_2) };
+    int pd0, pd1, pd2;
+    int idx;
+
+    static const int alpha_tab[] =
+    {
+      Q30(1.5146213770f/M_PI), Q30(1.5181334019f/M_PI), Q30(1.5234849453f/M_PI), Q30(1.5369486809f/M_PI), Q30(1.5500687361f/M_PI), Q30(1.5679757595f/M_PI),
+      Q30(1.4455626011f/M_PI), Q30(1.4531552792f/M_PI), Q30(1.4648091793f/M_PI), Q30(1.4945238829f/M_PI), Q30(1.5239057541f/M_PI), Q30(1.5644006729f/M_PI),
+      Q30(1.3738563061f/M_PI), Q30(1.3851221800f/M_PI), Q30(1.4026404619f/M_PI), Q30(1.4484288692f/M_PI), Q30(1.4949874878f/M_PI), Q30(1.5604078770f/M_PI),
+      Q30(1.2645189762f/M_PI), Q30(1.2796478271f/M_PI), Q30(1.3038636446f/M_PI), Q30(1.3710125685f/M_PI), Q30(1.4443849325f/M_PI), Q30(1.5532352924f/M_PI),
+      Q30(1.1507037878f/M_PI), Q30(1.1669205427f/M_PI), Q30(1.1938756704f/M_PI), Q30(1.2754167318f/M_PI), Q30(1.3761177063f/M_PI), Q30(1.5429240465f/M_PI),
+      Q30(1.0079245567f/M_PI), Q30(1.0208238363f/M_PI), Q30(1.0433073044f/M_PI), Q30(1.1208510399f/M_PI), Q30(1.2424604893f/M_PI), Q30(1.5185726881f/M_PI),
+      Q30(0.8995233774f/M_PI), Q30(0.9069069624f/M_PI), Q30(0.9201194048f/M_PI), Q30(0.9698365927f/M_PI), Q30(1.0671583414f/M_PI), Q30(1.4647934437f/M_PI),
+      Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI),
+      Q30(0.6712729335f/M_PI), Q30(0.6638893485f/M_PI), Q30(0.6506769061f/M_PI), Q30(0.6009597182f/M_PI), Q30(0.5036380291f/M_PI), Q30(0.1060028747f/M_PI),
+      Q30(0.5628717542f/M_PI), Q30(0.5499725342f/M_PI), Q30(0.5274890065f/M_PI), Q30(0.4499453008f/M_PI), Q30(0.3283358216f/M_PI), Q30(0.0522236861f/M_PI),
+      Q30(0.4200925827f/M_PI), Q30(0.4038758278f/M_PI), Q30(0.3769206405f/M_PI), Q30(0.2953795493f/M_PI), Q30(0.1946786791f/M_PI), Q30(0.0278722942f/M_PI),
+      Q30(0.3062773645f/M_PI), Q30(0.2911485136f/M_PI), Q30(0.2669326365f/M_PI), Q30(0.1997837722f/M_PI), Q30(0.1264114529f/M_PI), Q30(0.0175609849f/M_PI),
+      Q30(0.1969399750f/M_PI), Q30(0.1856741160f/M_PI), Q30(0.1681558639f/M_PI), Q30(0.1223674342f/M_PI), Q30(0.0758088827f/M_PI), Q30(0.0103884479f/M_PI),
+      Q30(0.1252337098f/M_PI), Q30(0.1176410317f/M_PI), Q30(0.1059871912f/M_PI), Q30(0.0762724727f/M_PI), Q30(0.0468905345f/M_PI), Q30(0.0063956482f/M_PI),
+      Q30(0.0561749674f/M_PI), Q30(0.0526629239f/M_PI), Q30(0.0473113805f/M_PI), Q30(0.0338476151f/M_PI), Q30(0.0207276177f/M_PI), Q30(0.0028205961f/M_PI),
+      Q30(1.5676341057f/M_PI), Q30(1.5678333044f/M_PI), Q30(1.5681363344f/M_PI), Q30(1.5688960552f/M_PI), Q30(1.5696337223f/M_PI), Q30(1.5706381798f/M_PI),
+      Q30(1.5651730299f/M_PI), Q30(1.5655272007f/M_PI), Q30(1.5660660267f/M_PI), Q30(1.5674170256f/M_PI), Q30(1.5687289238f/M_PI), Q30(1.5705151558f/M_PI),
+      Q30(1.5607966185f/M_PI), Q30(1.5614265203f/M_PI), Q30(1.5623844862f/M_PI), Q30(1.5647867918f/M_PI), Q30(1.5671195984f/M_PI), Q30(1.5702962875f/M_PI),
+      Q30(1.5530153513f/M_PI), Q30(1.5541347265f/M_PI), Q30(1.5558375120f/M_PI), Q30(1.5601085424f/M_PI), Q30(1.5642569065f/M_PI), Q30(1.5699069500f/M_PI),
+      Q30(1.5391840935f/M_PI), Q30(1.5411708355f/M_PI), Q30(1.5441943407f/M_PI), Q30(1.5517836809f/M_PI), Q30(1.5591609478f/M_PI), Q30(1.5692136288f/M_PI),
+      Q30(1.5146213770f/M_PI), Q30(1.5181334019f/M_PI), Q30(1.5234849453f/M_PI), Q30(1.5369486809f/M_PI), Q30(1.5500687361f/M_PI), Q30(1.5679757595f/M_PI),
+      Q30(1.4915299416f/M_PI), Q30(1.4964480400f/M_PI), Q30(1.5039558411f/M_PI), Q30(1.5229074955f/M_PI), Q30(1.5414420366f/M_PI), Q30(1.5667995214f/M_PI),
+      Q30(1.4590617418f/M_PI), Q30(1.4658898115f/M_PI), Q30(1.4763505459f/M_PI), Q30(1.5029321909f/M_PI), Q30(1.5291173458f/M_PI), Q30(1.5651149750f/M_PI),
+      Q30(1.4136143923f/M_PI), Q30(1.4229322672f/M_PI), Q30(1.4373078346f/M_PI), Q30(1.4743183851f/M_PI), Q30(1.5113102198f/M_PI), Q30(1.5626684427f/M_PI),
+      Q30(1.3505556583f/M_PI), Q30(1.3628427982f/M_PI), Q30(1.3820509911f/M_PI), Q30(1.4327841997f/M_PI), Q30(1.4850014448f/M_PI), Q30(1.5590143204f/M_PI),
+      Q30(1.2645189762f/M_PI), Q30(1.2796478271f/M_PI), Q30(1.3038636446f/M_PI), Q30(1.3710125685f/M_PI), Q30(1.4443849325f/M_PI), Q30(1.5532352924f/M_PI),
+      Q30(1.1919227839f/M_PI), Q30(1.2081253529f/M_PI), Q30(1.2346779108f/M_PI), Q30(1.3123005629f/M_PI), Q30(1.4034168720f/M_PI), Q30(1.5471596718f/M_PI),
+      Q30(1.1061993837f/M_PI), Q30(1.1219338179f/M_PI), Q30(1.1484941244f/M_PI), Q30(1.2320860624f/M_PI), Q30(1.3421301842f/M_PI), Q30(1.5373806953f/M_PI),
+      Q30(1.0079245567f/M_PI), Q30(1.0208238363f/M_PI), Q30(1.0433073044f/M_PI), Q30(1.1208510399f/M_PI), Q30(1.2424604893f/M_PI), Q30(1.5185726881f/M_PI),
+      Q30(0.8995233774f/M_PI), Q30(0.9069069624f/M_PI), Q30(0.9201194048f/M_PI), Q30(0.9698365927f/M_PI), Q30(1.0671583414f/M_PI), Q30(1.4647934437f/M_PI),
+      Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI),
+      Q30(0.6712729335f/M_PI), Q30(0.6638893485f/M_PI), Q30(0.6506769061f/M_PI), Q30(0.6009597182f/M_PI), Q30(0.5036380291f/M_PI), Q30(0.1060028747f/M_PI),
+      Q30(0.5628717542f/M_PI), Q30(0.5499725342f/M_PI), Q30(0.5274890065f/M_PI), Q30(0.4499453008f/M_PI), Q30(0.3283358216f/M_PI), Q30(0.0522236861f/M_PI),
+      Q30(0.4645969570f/M_PI), Q30(0.4488625824f/M_PI), Q30(0.4223022461f/M_PI), Q30(0.3387103081f/M_PI), Q30(0.2286661267f/M_PI), Q30(0.0334156826f/M_PI),
+      Q30(0.3788735867f/M_PI), Q30(0.3626709878f/M_PI), Q30(0.3361184299f/M_PI), Q30(0.2584958076f/M_PI), Q30(0.1673794836f/M_PI), Q30(0.0236366931f/M_PI),
+      Q30(0.3062773645f/M_PI), Q30(0.2911485136f/M_PI), Q30(0.2669326365f/M_PI), Q30(0.1997837722f/M_PI), Q30(0.1264114529f/M_PI), Q30(0.0175609849f/M_PI),
+      Q30(0.2202406377f/M_PI), Q30(0.2079535723f/M_PI), Q30(0.1887452900f/M_PI), Q30(0.1380121708f/M_PI), Q30(0.0857949182f/M_PI), Q30(0.0117820343f/M_PI),
+      Q30(0.1571819335f/M_PI), Q30(0.1478640437f/M_PI), Q30(0.1334884763f/M_PI), Q30(0.0964778885f/M_PI), Q30(0.0594860613f/M_PI), Q30(0.0081279324f/M_PI),
+      Q30(0.1117345318f/M_PI), Q30(0.1049065739f/M_PI), Q30(0.0944457650f/M_PI), Q30(0.0678641573f/M_PI), Q30(0.0416790098f/M_PI), Q30(0.0056813755f/M_PI),
+      Q30(0.0792663917f/M_PI), Q30(0.0743482932f/M_PI), Q30(0.0668405443f/M_PI), Q30(0.0478888862f/M_PI), Q30(0.0293543357f/M_PI), Q30(0.0039967746f/M_PI),
+      Q30(0.0561749674f/M_PI), Q30(0.0526629239f/M_PI), Q30(0.0473113805f/M_PI), Q30(0.0338476151f/M_PI), Q30(0.0207276177f/M_PI), Q30(0.0028205961f/M_PI),
+      Q30(0.0316122435f/M_PI), Q30(0.0296254847f/M_PI), Q30(0.0266019460f/M_PI), Q30(0.0190126132f/M_PI), Q30(0.0116353342f/M_PI), Q30(0.0015827164f/M_PI),
+      Q30(0.0177809205f/M_PI), Q30(0.0166615788f/M_PI), Q30(0.0149587989f/M_PI), Q30(0.0106877899f/M_PI), Q30(0.0065393616f/M_PI), Q30(0.0008894200f/M_PI),
+      Q30(0.0099996664f/M_PI), Q30(0.0093698399f/M_PI), Q30(0.0084118480f/M_PI), Q30(0.0060095116f/M_PI), Q30(0.0036767013f/M_PI), Q30(0.0005000498f/M_PI),
+      Q30(0.0056233541f/M_PI), Q30(0.0052691097f/M_PI), Q30(0.0047303112f/M_PI), Q30(0.0033792770f/M_PI), Q30(0.0020674451f/M_PI), Q30(0.0002811795f/M_PI),
+      Q30(0.0031622672f/M_PI), Q30(0.0029630491f/M_PI), Q30(0.0026600463f/M_PI), Q30(0.0019002859f/M_PI), Q30(0.0011625893f/M_PI), Q30(0.0001581155f/M_PI)
+    };
+
+    static const int gamma_tab[] =
+    {
+      Q30(0.0000000000f/M_PI), Q30(0.0195873566f/M_PI), Q30(0.0303316917f/M_PI), Q30(0.0448668823f/M_PI), Q30(0.0522258915f/M_PI), Q30(0.0561044961f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0433459543f/M_PI), Q30(0.0672172382f/M_PI), Q30(0.0997167900f/M_PI), Q30(0.1162951663f/M_PI), Q30(0.1250736862f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0672341362f/M_PI), Q30(0.1045235619f/M_PI), Q30(0.1558904350f/M_PI), Q30(0.1824723780f/M_PI), Q30(0.1966800541f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1011129096f/M_PI), Q30(0.1580764502f/M_PI), Q30(0.2387557179f/M_PI), Q30(0.2820728719f/M_PI), Q30(0.3058380187f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1315985769f/M_PI), Q30(0.2072522491f/M_PI), Q30(0.3188187480f/M_PI), Q30(0.3825501204f/M_PI), Q30(0.4193951190f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1603866369f/M_PI), Q30(0.2549437582f/M_PI), Q30(0.4029446840f/M_PI), Q30(0.4980689585f/M_PI), Q30(0.5615641475f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1736015975f/M_PI), Q30(0.2773745656f/M_PI), Q30(0.4461984038f/M_PI), Q30(0.5666890144f/M_PI), Q30(0.6686112881f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1784276664f/M_PI), Q30(0.2856673002f/M_PI), Q30(0.4630723596f/M_PI), Q30(0.5971632004f/M_PI), Q30(0.7603877187f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1736015975f/M_PI), Q30(0.2773745656f/M_PI), Q30(0.4461984038f/M_PI), Q30(0.5666890144f/M_PI), Q30(0.6686112881f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1603866369f/M_PI), Q30(0.2549437582f/M_PI), Q30(0.4029446840f/M_PI), Q30(0.4980689585f/M_PI), Q30(0.5615641475f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1315985769f/M_PI), Q30(0.2072522491f/M_PI), Q30(0.3188187480f/M_PI), Q30(0.3825501204f/M_PI), Q30(0.4193951190f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1011129096f/M_PI), Q30(0.1580764502f/M_PI), Q30(0.2387557179f/M_PI), Q30(0.2820728719f/M_PI), Q30(0.3058380187f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0672341362f/M_PI), Q30(0.1045235619f/M_PI), Q30(0.1558904350f/M_PI), Q30(0.1824723780f/M_PI), Q30(0.1966800541f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0433459543f/M_PI), Q30(0.0672172382f/M_PI), Q30(0.0997167900f/M_PI), Q30(0.1162951663f/M_PI), Q30(0.1250736862f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0195873566f/M_PI), Q30(0.0303316917f/M_PI), Q30(0.0448668823f/M_PI), Q30(0.0522258915f/M_PI), Q30(0.0561044961f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0011053939f/M_PI), Q30(0.0017089852f/M_PI), Q30(0.0025254129f/M_PI), Q30(0.0029398468f/M_PI), Q30(0.0031597170f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0019607407f/M_PI), Q30(0.0030395309f/M_PI), Q30(0.0044951206f/M_PI), Q30(0.0052305623f/M_PI), Q30(0.0056152637f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0034913034f/M_PI), Q30(0.0054070661f/M_PI), Q30(0.0079917293f/M_PI), Q30(0.0092999367f/M_PI), Q30(0.0099875759f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0062100487f/M_PI), Q30(0.0096135242f/M_PI), Q30(0.0142110568f/M_PI), Q30(0.0165348612f/M_PI), Q30(0.0177587029f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0110366223f/M_PI), Q30(0.0170863140f/M_PI), Q30(0.0252620988f/M_PI), Q30(0.0293955617f/M_PI), Q30(0.0315726399f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0195873566f/M_PI), Q30(0.0303316917f/M_PI), Q30(0.0448668823f/M_PI), Q30(0.0522258915f/M_PI), Q30(0.0561044961f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0275881495f/M_PI), Q30(0.0427365713f/M_PI), Q30(0.0632618815f/M_PI), Q30(0.0736731067f/M_PI), Q30(0.0791663304f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0387469754f/M_PI), Q30(0.0600636788f/M_PI), Q30(0.0890387669f/M_PI), Q30(0.1037906483f/M_PI), Q30(0.1115923747f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0541138873f/M_PI), Q30(0.0839984417f/M_PI), Q30(0.1248718798f/M_PI), Q30(0.1458375156f/M_PI), Q30(0.1569785923f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0747506917f/M_PI), Q30(0.1163287833f/M_PI), Q30(0.1738867164f/M_PI), Q30(0.2038587779f/M_PI), Q30(0.2199459076f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1011129096f/M_PI), Q30(0.1580764502f/M_PI), Q30(0.2387557179f/M_PI), Q30(0.2820728719f/M_PI), Q30(0.3058380187f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1212290376f/M_PI), Q30(0.1903949380f/M_PI), Q30(0.2907958031f/M_PI), Q30(0.3466993868f/M_PI), Q30(0.3782821596f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1418247074f/M_PI), Q30(0.2240308374f/M_PI), Q30(0.3474813402f/M_PI), Q30(0.4202919006f/M_PI), Q30(0.4637607038f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1603866369f/M_PI), Q30(0.2549437582f/M_PI), Q30(0.4029446840f/M_PI), Q30(0.4980689585f/M_PI), Q30(0.5615641475f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1736015975f/M_PI), Q30(0.2773745656f/M_PI), Q30(0.4461984038f/M_PI), Q30(0.5666890144f/M_PI), Q30(0.6686112881f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1784276664f/M_PI), Q30(0.2856673002f/M_PI), Q30(0.4630723596f/M_PI), Q30(0.5971632004f/M_PI), Q30(0.7603877187f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1736015975f/M_PI), Q30(0.2773745656f/M_PI), Q30(0.4461984038f/M_PI), Q30(0.5666890144f/M_PI), Q30(0.6686112881f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1603866369f/M_PI), Q30(0.2549437582f/M_PI), Q30(0.4029446840f/M_PI), Q30(0.4980689585f/M_PI), Q30(0.5615641475f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1418247074f/M_PI), Q30(0.2240308374f/M_PI), Q30(0.3474813402f/M_PI), Q30(0.4202919006f/M_PI), Q30(0.4637607038f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1212290376f/M_PI), Q30(0.1903949380f/M_PI), Q30(0.2907958031f/M_PI), Q30(0.3466993868f/M_PI), Q30(0.3782821596f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1011129096f/M_PI), Q30(0.1580764502f/M_PI), Q30(0.2387557179f/M_PI), Q30(0.2820728719f/M_PI), Q30(0.3058380187f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0747506917f/M_PI), Q30(0.1163287833f/M_PI), Q30(0.1738867164f/M_PI), Q30(0.2038587779f/M_PI), Q30(0.2199459076f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0541138873f/M_PI), Q30(0.0839984417f/M_PI), Q30(0.1248718798f/M_PI), Q30(0.1458375156f/M_PI), Q30(0.1569785923f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0387469754f/M_PI), Q30(0.0600636788f/M_PI), Q30(0.0890387669f/M_PI), Q30(0.1037906483f/M_PI), Q30(0.1115923747f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0275881495f/M_PI), Q30(0.0427365713f/M_PI), Q30(0.0632618815f/M_PI), Q30(0.0736731067f/M_PI), Q30(0.0791663304f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0195873566f/M_PI), Q30(0.0303316917f/M_PI), Q30(0.0448668823f/M_PI), Q30(0.0522258915f/M_PI), Q30(0.0561044961f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0110366223f/M_PI), Q30(0.0170863140f/M_PI), Q30(0.0252620988f/M_PI), Q30(0.0293955617f/M_PI), Q30(0.0315726399f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0062100487f/M_PI), Q30(0.0096135242f/M_PI), Q30(0.0142110568f/M_PI), Q30(0.0165348612f/M_PI), Q30(0.0177587029f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0034913034f/M_PI), Q30(0.0054070661f/M_PI), Q30(0.0079917293f/M_PI), Q30(0.0092999367f/M_PI), Q30(0.0099875759f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0019607407f/M_PI), Q30(0.0030395309f/M_PI), Q30(0.0044951206f/M_PI), Q30(0.0052305623f/M_PI), Q30(0.0056152637f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0011053939f/M_PI), Q30(0.0017089852f/M_PI), Q30(0.0025254129f/M_PI), Q30(0.0029398468f/M_PI), Q30(0.0031597170f/M_PI)
+    };
+
+    static const int iid_par_dequant_c1[] = {
+        //iid_par_dequant_default
+        Q30(1.41198278375959f), Q30(1.40313815268360f), Q30(1.38687670404960f), Q30(1.34839972492648f),
+        Q30(1.29124937110028f), Q30(1.19603741667993f), Q30(1.10737240362323f), Q30(1),
+        Q30(0.87961716655242f), Q30(0.75464859232732f), Q30(0.57677990744575f), Q30(0.42640143271122f),
+        Q30(0.27671828230984f), Q30(0.17664462766713f), Q30(0.07940162697653f),
+        //iid_par_dequant_fine
+        Q30(1.41420649135832f), Q30(1.41419120222364f), Q30(1.41414285699784f), Q30(1.41399000859438f),
+        Q30(1.41350698548044f), Q30(1.41198278375959f), Q30(1.40977302262355f), Q30(1.40539479488545f),
+        Q30(1.39677960498402f), Q30(1.38005309967827f), Q30(1.34839972492648f), Q30(1.31392017367631f),
+        Q30(1.26431008149654f), Q30(1.19603741667993f), Q30(1.10737240362323f), Q30(1),
+        Q30(0.87961716655242f), Q30(0.75464859232732f), Q30(0.63365607219232f), Q30(0.52308104267543f),
+        Q30(0.42640143271122f), Q30(0.30895540465965f), Q30(0.22137464873077f), Q30(0.15768788954414f),
+        Q30(0.11198225164225f), Q30(0.07940162697653f), Q30(0.04469901562677f), Q30(0.02514469318284f),
+        Q30(0.01414142856998f), Q30(0.00795258154731f), Q30(0.00447211359449f),
+    };
+
+    static const int acos_icc_invq[] = {
+        Q31(0), Q31(0.178427635f/M_PI), Q31(0.28566733f/M_PI), Q31(0.46307236f/M_PI), Q31(0.59716315f/M_PI), Q31(0.78539816f/M_PI), Q31(1.10030855f/M_PI), Q31(1.57079633f/M_PI)
+    };
+    int iid, icc;
+
+    int k, m;
+    static const int8_t f_center_20[] = {
+        -3, -1, 1, 3, 5, 7, 10, 14, 18, 22,
+    };
+    static const int32_t f_center_34[] = {
+      Q31(  2/768.0),Q31(  6/768.0),Q31(10/768.0),Q31(14/768.0),Q31( 18/768.0),Q31( 22/768.0),Q31( 26/768.0),Q31(30/768.0),
+      Q31( 34/768.0),Q31(-10/768.0),Q31(-6/768.0),Q31(-2/768.0),Q31( 51/768.0),Q31( 57/768.0),Q31( 15/768.0),Q31(21/768.0),
+      Q31( 27/768.0),Q31( 33/768.0),Q31(39/768.0),Q31(45/768.0),Q31( 54/768.0),Q31( 66/768.0),Q31( 78/768.0),Q31(42/768.0),
+      Q31(102/768.0),Q31( 66/768.0),Q31(78/768.0),Q31(90/768.0),Q31(102/768.0),Q31(114/768.0),Q31(126/768.0),Q31(90/768.0)
+    };
+    static const int fractional_delay_links[] = { Q31(0.43f), Q31(0.75f), Q31(0.347f) };
+    const int fractional_delay_gain = Q31(0.39f);
+
+    for (pd0 = 0; pd0 < 8; pd0++) {
+        int pd0_re = (ipdopd_cos[pd0]+2)>>2;
+        int pd0_im = (ipdopd_sin[pd0]+2)>>2;
+        for (pd1 = 0; pd1 < 8; pd1++) {
+            int pd1_re = ipdopd_cos[pd1] >> 1;
+            int pd1_im = ipdopd_sin[pd1] >> 1;
+            for (pd2 = 0; pd2 < 8; pd2++) {
+                int shift, round;
+                int pd2_re = ipdopd_cos[pd2];
+                int pd2_im = ipdopd_sin[pd2];
+                int re_smooth = pd0_re + pd1_re + pd2_re;
+                int im_smooth = pd0_im + pd1_im + pd2_im;
+
+                SoftFloat pd_mag = av_int2sf(((ipdopd_cos[(pd0-pd1)&7]+8)>>4) + ((ipdopd_cos[(pd0-pd2)&7]+4)>>3) +
+                                               ((ipdopd_cos[(pd1-pd2)&7]+2)>>2) + 0x15000000, 28);
+                pd_mag = av_div_sf(FLOAT_1, av_sqrt_sf(pd_mag));
+                shift = 30 - pd_mag.exp;
+                round = 1 << (shift-1);
+                pd_re_smooth[pd0*64+pd1*8+pd2] = (int)(((int64_t)re_smooth * pd_mag.mant + round) >> shift);
+                pd_im_smooth[pd0*64+pd1*8+pd2] = (int)(((int64_t)im_smooth * pd_mag.mant + round) >> shift);
+            }
+        }
+    }
+
+    idx = 0;
+    for (iid = 0; iid < 46; iid++) {
+        int c1, c2;
+
+        c1 = iid_par_dequant_c1[iid];
+        if (iid < 15)
+          c2 = iid_par_dequant_c1[14-iid];
+        else
+          c2 = iid_par_dequant_c1[60-iid];
+
+        for (icc = 0; icc < 8; icc++) {
+            /*if (PS_BASELINE || ps->icc_mode < 3)*/{
+                int alpha, beta;
+                int ca, sa, cb, sb;
+
+                alpha = acos_icc_invq[icc];
+                beta = (int)(((int64_t)alpha * 1518500250 + 0x40000000) >> 31);
+                alpha >>= 1;
+                beta = (int)(((int64_t)beta * (c1 - c2) + 0x40000000) >> 31);
+                av_sincos_sf(beta + alpha, &sa, &ca);
+                av_sincos_sf(beta - alpha, &sb, &cb);
+
+                HA[iid][icc][0] = (int)(((int64_t)c2 * ca + 0x20000000) >> 30);
+                HA[iid][icc][1] = (int)(((int64_t)c1 * cb + 0x20000000) >> 30);
+                HA[iid][icc][2] = (int)(((int64_t)c2 * sa + 0x20000000) >> 30);
+                HA[iid][icc][3] = (int)(((int64_t)c1 * sb + 0x20000000) >> 30);
+            } /* else */ {
+                int alpha_int, gamma_int;
+                int alpha_c_int, alpha_s_int, gamma_c_int, gamma_s_int;
+
+                alpha_int = alpha_tab[idx];
+                gamma_int = gamma_tab[idx];
+
+                av_sincos_sf(alpha_int, &alpha_s_int, &alpha_c_int);
+                av_sincos_sf(gamma_int, &gamma_s_int, &gamma_c_int);
+
+                alpha_c_int = (int)(((int64_t)alpha_c_int * 1518500250 + 0x20000000) >> 30);
+                alpha_s_int = (int)(((int64_t)alpha_s_int * 1518500250 + 0x20000000) >> 30);
+
+                HB[iid][icc][0] = (int)(((int64_t)alpha_c_int * gamma_c_int + 0x20000000) >> 30);
+                HB[iid][icc][1] = (int)(((int64_t)alpha_s_int * gamma_c_int + 0x20000000) >> 30);
+                HB[iid][icc][2] = -(int)(((int64_t)alpha_s_int * gamma_s_int + 0x20000000) >> 30);
+                HB[iid][icc][3] = (int)(((int64_t)alpha_c_int * gamma_s_int + 0x20000000) >> 30);
+            }
+
+            if (icc < 5 || icc > 6)
+              idx++;
+        }
+    }
+
+    for (k = 0; k < NR_ALLPASS_BANDS20; k++) {
+        int theta;
+        int64_t f_center;
+        int c, s;
+
+        if (k < FF_ARRAY_ELEMS(f_center_20))
+          f_center = f_center_20[k];
+        else
+          f_center = (k << 3) - 52;
+
+        for (m = 0; m < PS_AP_LINKS; m++) {
+            theta = (int)(((int64_t)fractional_delay_links[m] * f_center + 8) >> 4);
+            av_sincos_sf(-theta, &s, &c);
+            Q_fract_allpass[0][k][m][0] = c;
+            Q_fract_allpass[0][k][m][1] = s;
+        }
+
+        theta = (int)(((int64_t)fractional_delay_gain * f_center + 8) >> 4);
+        av_sincos_sf(-theta, &s, &c);
+        phi_fract[0][k][0] = c;
+        phi_fract[0][k][1] = s;
+    }
+
+    for (k = 0; k < NR_ALLPASS_BANDS34; k++) {
+        int theta, f_center;
+        int c, s;
+
+        if (k < FF_ARRAY_ELEMS(f_center_34))
+            f_center = f_center_34[k];
+        else
+            f_center = ((int64_t)k << 26) - (53 << 25);
+
+        for (m = 0; m < PS_AP_LINKS; m++) {
+            theta = (int)(((int64_t)fractional_delay_links[m] * f_center + 0x10000000) >> 27);
+            av_sincos_sf(-theta, &s, &c);
+            Q_fract_allpass[1][k][m][0] = c;
+            Q_fract_allpass[1][k][m][1] = s;
+        }
+
+        theta = (int)(((int64_t)fractional_delay_gain * f_center + 0x10000000) >> 27);
+        av_sincos_sf(-theta, &s, &c);
+        phi_fract[1][k][0] = c;
+        phi_fract[1][k][1] = s;
+    }
+
+    make_filters_from_proto(f20_0_8,  g0_Q8,   8);
+    make_filters_from_proto(f34_0_12, g0_Q12, 12);
+    make_filters_from_proto(f34_1_8,  g1_Q8,   8);
+    make_filters_from_proto(f34_2_4,  g2_Q4,   4);
+}
+#endif /* CONFIG_HARDCODED_TABLES */
+
+#endif /* AVCODEC_AACPS_FIXED_TABLEGEN_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacps_float.c

@@ -0,0 +1,24 @@
+/*
+ * MPEG-4 Parametric Stereo decoding functions
+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#define USE_FIXED 0
+
+#include "aacps.c"

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacps_tablegen.c

@@ -0,0 +1,24 @@
+/*
+ * Generate a header file for hardcoded Parametric Stereo tables
+ *
+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#define USE_FIXED 0
+#include "aacps_tablegen_template.c"

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacps_tablegen.h

@@ -0,0 +1,217 @@
+/*
+ * Header file for hardcoded Parametric Stereo tables
+ *
+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AACPS_TABLEGEN_H
+#define AVCODEC_AACPS_TABLEGEN_H
+
+#include <math.h>
+#include <stdint.h>
+
+#if CONFIG_HARDCODED_TABLES
+#define ps_tableinit()
+#define TABLE_CONST const
+#include "libavcodec/aacps_tables.h"
+#else
+#include "libavutil/common.h"
+#include "libavutil/libm.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/mem.h"
+#define NR_ALLPASS_BANDS20 30
+#define NR_ALLPASS_BANDS34 50
+#define PS_AP_LINKS 3
+#define TABLE_CONST
+static float pd_re_smooth[8*8*8];
+static float pd_im_smooth[8*8*8];
+static float HA[46][8][4];
+static float HB[46][8][4];
+static DECLARE_ALIGNED(16, float, f20_0_8) [ 8][8][2];
+static DECLARE_ALIGNED(16, float, f34_0_12)[12][8][2];
+static DECLARE_ALIGNED(16, float, f34_1_8) [ 8][8][2];
+static DECLARE_ALIGNED(16, float, f34_2_4) [ 4][8][2];
+static TABLE_CONST DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2];
+static DECLARE_ALIGNED(16, float, phi_fract)[2][50][2];
+
+static const float g0_Q8[] = {
+    0.00746082949812f, 0.02270420949825f, 0.04546865930473f, 0.07266113929591f,
+    0.09885108575264f, 0.11793710567217f, 0.125f
+};
+
+static const float g0_Q12[] = {
+    0.04081179924692f, 0.03812810994926f, 0.05144908135699f, 0.06399831151592f,
+    0.07428313801106f, 0.08100347892914f, 0.08333333333333f
+};
+
+static const float g1_Q8[] = {
+    0.01565675600122f, 0.03752716391991f, 0.05417891378782f, 0.08417044116767f,
+    0.10307344158036f, 0.12222452249753f, 0.125f
+};
+
+static const float g2_Q4[] = {
+    -0.05908211155639f, -0.04871498374946f, 0.0f,   0.07778723915851f,
+     0.16486303567403f,  0.23279856662996f, 0.25f
+};
+
+static av_cold void make_filters_from_proto(float (*filter)[8][2], const float *proto, int bands)
+{
+    int q, n;
+    for (q = 0; q < bands; q++) {
+        for (n = 0; n < 7; n++) {
+            double theta = 2 * M_PI * (q + 0.5) * (n - 6) / bands;
+            filter[q][n][0] = proto[n] *  cos(theta);
+            filter[q][n][1] = proto[n] * -sin(theta);
+        }
+    }
+}
+
+static av_cold void ps_tableinit(void)
+{
+    static const float ipdopd_sin[] = { 0, M_SQRT1_2, 1,  M_SQRT1_2,  0, -M_SQRT1_2, -1, -M_SQRT1_2 };
+    static const float ipdopd_cos[] = { 1, M_SQRT1_2, 0, -M_SQRT1_2, -1, -M_SQRT1_2,  0,  M_SQRT1_2 };
+    int pd0, pd1, pd2;
+
+    static const float iid_par_dequant[] = {
+        //iid_par_dequant_default
+        0.05623413251903, 0.12589254117942, 0.19952623149689, 0.31622776601684,
+        0.44668359215096, 0.63095734448019, 0.79432823472428, 1,
+        1.25892541179417, 1.58489319246111, 2.23872113856834, 3.16227766016838,
+        5.01187233627272, 7.94328234724282, 17.7827941003892,
+        //iid_par_dequant_fine
+        0.00316227766017, 0.00562341325190, 0.01,             0.01778279410039,
+        0.03162277660168, 0.05623413251903, 0.07943282347243, 0.11220184543020,
+        0.15848931924611, 0.22387211385683, 0.31622776601684, 0.39810717055350,
+        0.50118723362727, 0.63095734448019, 0.79432823472428, 1,
+        1.25892541179417, 1.58489319246111, 1.99526231496888, 2.51188643150958,
+        3.16227766016838, 4.46683592150963, 6.30957344480193, 8.91250938133745,
+        12.5892541179417, 17.7827941003892, 31.6227766016838, 56.2341325190349,
+        100,              177.827941003892, 316.227766016837,
+    };
+    static const float icc_invq[] = {
+        1, 0.937,      0.84118,    0.60092,    0.36764,   0,      -0.589,    -1
+    };
+    static const float acos_icc_invq[] = {
+        0, 0.35685527, 0.57133466, 0.92614472, 1.1943263, M_PI/2, 2.2006171, M_PI
+    };
+    int iid, icc;
+
+    int k, m;
+    static const int8_t f_center_20[] = {
+        -3, -1, 1, 3, 5, 7, 10, 14, 18, 22,
+    };
+    static const int8_t f_center_34[] = {
+         2,  6, 10, 14, 18, 22, 26, 30,
+        34,-10, -6, -2, 51, 57, 15, 21,
+        27, 33, 39, 45, 54, 66, 78, 42,
+       102, 66, 78, 90,102,114,126, 90,
+    };
+    static const float fractional_delay_links[] = { 0.43f, 0.75f, 0.347f };
+    const float fractional_delay_gain = 0.39f;
+
+    for (pd0 = 0; pd0 < 8; pd0++) {
+        float pd0_re = ipdopd_cos[pd0];
+        float pd0_im = ipdopd_sin[pd0];
+        for (pd1 = 0; pd1 < 8; pd1++) {
+            float pd1_re = ipdopd_cos[pd1];
+            float pd1_im = ipdopd_sin[pd1];
+            for (pd2 = 0; pd2 < 8; pd2++) {
+                float pd2_re = ipdopd_cos[pd2];
+                float pd2_im = ipdopd_sin[pd2];
+                float re_smooth = 0.25f * pd0_re + 0.5f * pd1_re + pd2_re;
+                float im_smooth = 0.25f * pd0_im + 0.5f * pd1_im + pd2_im;
+                float pd_mag = 1 / hypot(im_smooth, re_smooth);
+                pd_re_smooth[pd0*64+pd1*8+pd2] = re_smooth * pd_mag;
+                pd_im_smooth[pd0*64+pd1*8+pd2] = im_smooth * pd_mag;
+            }
+        }
+    }
+
+    for (iid = 0; iid < 46; iid++) {
+        float c = iid_par_dequant[iid]; ///< Linear Inter-channel Intensity Difference
+        float c1 = (float)M_SQRT2 / sqrtf(1.0f + c*c);
+        float c2 = c * c1;
+        for (icc = 0; icc < 8; icc++) {
+            /*if (PS_BASELINE || ps->icc_mode < 3)*/ {
+                float alpha = 0.5f * acos_icc_invq[icc];
+                float beta  = alpha * (c1 - c2) * (float)M_SQRT1_2;
+                HA[iid][icc][0] = c2 * cosf(beta + alpha);
+                HA[iid][icc][1] = c1 * cosf(beta - alpha);
+                HA[iid][icc][2] = c2 * sinf(beta + alpha);
+                HA[iid][icc][3] = c1 * sinf(beta - alpha);
+            } /* else */ {
+                float alpha, gamma, mu, rho;
+                float alpha_c, alpha_s, gamma_c, gamma_s;
+                rho = FFMAX(icc_invq[icc], 0.05f);
+                alpha = 0.5f * atan2f(2.0f * c * rho, c*c - 1.0f);
+                mu = c + 1.0f / c;
+                mu = sqrtf(1 + (4 * rho * rho - 4)/(mu * mu));
+                gamma = atanf(sqrtf((1.0f - mu)/(1.0f + mu)));
+                if (alpha < 0) alpha += M_PI/2;
+                alpha_c = cosf(alpha);
+                alpha_s = sinf(alpha);
+                gamma_c = cosf(gamma);
+                gamma_s = sinf(gamma);
+                HB[iid][icc][0] =  M_SQRT2 * alpha_c * gamma_c;
+                HB[iid][icc][1] =  M_SQRT2 * alpha_s * gamma_c;
+                HB[iid][icc][2] = -M_SQRT2 * alpha_s * gamma_s;
+                HB[iid][icc][3] =  M_SQRT2 * alpha_c * gamma_s;
+            }
+        }
+    }
+
+    for (k = 0; k < NR_ALLPASS_BANDS20; k++) {
+        double f_center, theta;
+        if (k < FF_ARRAY_ELEMS(f_center_20))
+            f_center = f_center_20[k] * 0.125;
+        else
+            f_center = k - 6.5f;
+        for (m = 0; m < PS_AP_LINKS; m++) {
+            theta = -M_PI * fractional_delay_links[m] * f_center;
+            Q_fract_allpass[0][k][m][0] = cos(theta);
+            Q_fract_allpass[0][k][m][1] = sin(theta);
+        }
+        theta = -M_PI*fractional_delay_gain*f_center;
+        phi_fract[0][k][0] = cos(theta);
+        phi_fract[0][k][1] = sin(theta);
+    }
+    for (k = 0; k < NR_ALLPASS_BANDS34; k++) {
+        double f_center, theta;
+        if (k < FF_ARRAY_ELEMS(f_center_34))
+            f_center = f_center_34[k] / 24.0;
+        else
+            f_center = k - 26.5f;
+        for (m = 0; m < PS_AP_LINKS; m++) {
+            theta = -M_PI * fractional_delay_links[m] * f_center;
+            Q_fract_allpass[1][k][m][0] = cos(theta);
+            Q_fract_allpass[1][k][m][1] = sin(theta);
+        }
+        theta = -M_PI*fractional_delay_gain*f_center;
+        phi_fract[1][k][0] = cos(theta);
+        phi_fract[1][k][1] = sin(theta);
+    }
+
+    make_filters_from_proto(f20_0_8,  g0_Q8,   8);
+    make_filters_from_proto(f34_0_12, g0_Q12, 12);
+    make_filters_from_proto(f34_1_8,  g1_Q8,   8);
+    make_filters_from_proto(f34_2_4,  g2_Q4,   4);
+}
+#endif /* CONFIG_HARDCODED_TABLES */
+
+#endif /* AVCODEC_AACPS_TABLEGEN_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacps_tablegen_template.c

@@ -0,0 +1,107 @@
+/*
+ * Generate a header file for hardcoded Parametric Stereo tables
+ *
+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdlib.h>
+#define CONFIG_HARDCODED_TABLES 0
+#include "aac_defines.h"
+
+#if USE_FIXED
+#define TYPE_NAME "int32_t"
+typedef int32_t INT32FLOAT;
+#define ARRAY_RENAME(x) write_int32_t_ ## x
+#define ARRAY_URENAME(x) write_uint32_t_ ## x
+#include "aacps_fixed_tablegen.h"
+#else
+#define TYPE_NAME "float"
+typedef float INT32FLOAT;
+#define ARRAY_RENAME(x) write_float_ ## x
+#define ARRAY_URENAME(x) write_float_ ## x
+#include "aacps_tablegen.h"
+#endif /* USE_FIXED */
+#include "tableprint.h"
+
+void ARRAY_RENAME(3d_array) (const void *p, int b, int c, int d)
+{
+    int i;
+    const INT32FLOAT *f = p;
+    for (i = 0; i < b; i++) {
+        printf("{\n");
+        ARRAY_URENAME(2d_array)(f, c, d);
+        printf("},\n");
+        f += c * d;
+    }
+}
+
+void ARRAY_RENAME(4d_array) (const void *p, int a, int b, int c, int d)
+{
+    int i;
+    const INT32FLOAT *f = p;
+    for (i = 0; i < a; i++) {
+        printf("{\n");
+        ARRAY_RENAME(3d_array)(f, b, c, d);
+        printf("},\n");
+        f += b * c * d;
+    }
+}
+
+int main(void)
+{
+    ps_tableinit();
+
+    write_fileheader();
+
+    printf("static const %s pd_re_smooth[8*8*8] = {\n", TYPE_NAME);
+    ARRAY_RENAME(array)(pd_re_smooth, 8*8*8);
+    printf("};\n");
+    printf("static const %s pd_im_smooth[8*8*8] = {\n", TYPE_NAME);
+    ARRAY_RENAME(array)(pd_im_smooth, 8*8*8);
+    printf("};\n");
+
+    printf("static const %s HA[46][8][4] = {\n", TYPE_NAME);
+    ARRAY_RENAME(3d_array)(HA, 46, 8, 4);
+    printf("};\n");
+    printf("static const %s HB[46][8][4] = {\n", TYPE_NAME);
+    ARRAY_RENAME(3d_array)(HB, 46, 8, 4);
+    printf("};\n");
+
+    printf("static const DECLARE_ALIGNED(16, %s, f20_0_8)[8][8][2] = {\n", TYPE_NAME);
+    ARRAY_RENAME(3d_array)(f20_0_8, 8, 8, 2);
+    printf("};\n");
+    printf("static const DECLARE_ALIGNED(16, %s, f34_0_12)[12][8][2] = {\n", TYPE_NAME);
+    ARRAY_RENAME(3d_array)(f34_0_12, 12, 8, 2);
+    printf("};\n");
+    printf("static const DECLARE_ALIGNED(16, %s, f34_1_8)[8][8][2] = {\n", TYPE_NAME);
+    ARRAY_RENAME(3d_array)(f34_1_8, 8, 8, 2);
+    printf("};\n");
+    printf("static const DECLARE_ALIGNED(16, %s, f34_2_4)[4][8][2] = {\n", TYPE_NAME);
+    ARRAY_RENAME(3d_array)(f34_2_4, 4, 8, 2);
+    printf("};\n");
+
+    printf("static const DECLARE_ALIGNED(16, %s, Q_fract_allpass)[2][50][3][2] = {\n", TYPE_NAME);
+    ARRAY_RENAME(4d_array)(Q_fract_allpass, 2, 50, 3, 2);
+    printf("};\n");
+    printf("static const DECLARE_ALIGNED(16, %s, phi_fract)[2][50][2] = {\n", TYPE_NAME);
+    ARRAY_RENAME(3d_array)(phi_fract, 2, 50, 2);
+    printf("};\n");
+
+    return 0;
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacpsdata.c

@@ -0,0 +1,163 @@
+/*
+ * MPEG-4 Parametric Stereo data tables
+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+static const uint8_t huff_iid_df1_bits[] = {
+    18, 18, 18, 18, 18, 18, 18, 18, 18, 17, 18, 17, 17, 16, 16, 15, 14, 14,
+    13, 12, 12, 11, 10, 10,  8,  7,  6,  5,  4,  3,  1,  3,  4,  5,  6,  7,
+     8,  9, 10, 11, 11, 12, 13, 14, 14, 15, 16, 16, 17, 17, 18, 17, 18, 18,
+    18, 18, 18, 18, 18, 18, 18,
+};
+
+static const uint32_t huff_iid_df1_codes[] = {
+    0x01FEB4, 0x01FEB5, 0x01FD76, 0x01FD77, 0x01FD74, 0x01FD75, 0x01FE8A,
+    0x01FE8B, 0x01FE88, 0x00FE80, 0x01FEB6, 0x00FE82, 0x00FEB8, 0x007F42,
+    0x007FAE, 0x003FAF, 0x001FD1, 0x001FE9, 0x000FE9, 0x0007EA, 0x0007FB,
+    0x0003FB, 0x0001FB, 0x0001FF, 0x00007C, 0x00003C, 0x00001C, 0x00000C,
+    0x000000, 0x000001, 0x000001, 0x000002, 0x000001, 0x00000D, 0x00001D,
+    0x00003D, 0x00007D, 0x0000FC, 0x0001FC, 0x0003FC, 0x0003F4, 0x0007EB,
+    0x000FEA, 0x001FEA, 0x001FD6, 0x003FD0, 0x007FAF, 0x007F43, 0x00FEB9,
+    0x00FE83, 0x01FEB7, 0x00FE81, 0x01FE89, 0x01FE8E, 0x01FE8F, 0x01FE8C,
+    0x01FE8D, 0x01FEB2, 0x01FEB3, 0x01FEB0, 0x01FEB1,
+};
+
+static const uint8_t huff_iid_dt1_bits[] = {
+    16, 16, 16, 16, 16, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 13,
+    13, 13, 12, 12, 11, 10,  9,  9,  7,  6,  5,  3,  1,  2,  5,  6,  7,  8,
+     9, 10, 11, 11, 12, 12, 13, 13, 14, 14, 15, 15, 15, 15, 16, 16, 16, 16,
+    16, 16, 16, 16, 16, 16, 16,
+};
+
+static const uint16_t huff_iid_dt1_codes[] = {
+    0x004ED4, 0x004ED5, 0x004ECE, 0x004ECF, 0x004ECC, 0x004ED6, 0x004ED8,
+    0x004F46, 0x004F60, 0x002718, 0x002719, 0x002764, 0x002765, 0x00276D,
+    0x0027B1, 0x0013B7, 0x0013D6, 0x0009C7, 0x0009E9, 0x0009ED, 0x0004EE,
+    0x0004F7, 0x000278, 0x000139, 0x00009A, 0x00009F, 0x000020, 0x000011,
+    0x00000A, 0x000003, 0x000001, 0x000000, 0x00000B, 0x000012, 0x000021,
+    0x00004C, 0x00009B, 0x00013A, 0x000279, 0x000270, 0x0004EF, 0x0004E2,
+    0x0009EA, 0x0009D8, 0x0013D7, 0x0013D0, 0x0027B2, 0x0027A2, 0x00271A,
+    0x00271B, 0x004F66, 0x004F67, 0x004F61, 0x004F47, 0x004ED9, 0x004ED7,
+    0x004ECD, 0x004ED2, 0x004ED3, 0x004ED0, 0x004ED1,
+};
+
+static const uint8_t huff_iid_df0_bits[] = {
+    17, 17, 17, 17, 16, 15, 13, 10,  9,  7,  6,  5,  4,  3,  1,  3,  4,  5,
+     6,  6,  8, 11, 13, 14, 14, 15, 17, 18, 18,
+};
+
+static const uint32_t huff_iid_df0_codes[] = {
+    0x01FFFB, 0x01FFFC, 0x01FFFD, 0x01FFFA, 0x00FFFC, 0x007FFC, 0x001FFD,
+    0x0003FE, 0x0001FE, 0x00007E, 0x00003C, 0x00001D, 0x00000D, 0x000005,
+    0x000000, 0x000004, 0x00000C, 0x00001C, 0x00003D, 0x00003E, 0x0000FE,
+    0x0007FE, 0x001FFC, 0x003FFC, 0x003FFD, 0x007FFD, 0x01FFFE, 0x03FFFE,
+    0x03FFFF,
+};
+
+static const uint8_t huff_iid_dt0_bits[] = {
+    19, 19, 19, 20, 20, 20, 17, 15, 12, 10,  8,  6,  4,  2,  1,  3,  5,  7,
+     9, 11, 13, 14, 17, 19, 20, 20, 20, 20, 20,
+};
+
+static const uint32_t huff_iid_dt0_codes[] = {
+    0x07FFF9, 0x07FFFA, 0x07FFFB, 0x0FFFF8, 0x0FFFF9, 0x0FFFFA, 0x01FFFD,
+    0x007FFE, 0x000FFE, 0x0003FE, 0x0000FE, 0x00003E, 0x00000E, 0x000002,
+    0x000000, 0x000006, 0x00001E, 0x00007E, 0x0001FE, 0x0007FE, 0x001FFE,
+    0x003FFE, 0x01FFFC, 0x07FFF8, 0x0FFFFB, 0x0FFFFC, 0x0FFFFD, 0x0FFFFE,
+    0x0FFFFF,
+};
+
+static const uint8_t huff_icc_df_bits[] = {
+    14, 14, 12, 10, 7, 5, 3, 1, 2, 4, 6, 8, 9, 11, 13,
+};
+
+static const uint16_t huff_icc_df_codes[] = {
+    0x3FFF, 0x3FFE, 0x0FFE, 0x03FE, 0x007E, 0x001E, 0x0006, 0x0000,
+    0x0002, 0x000E, 0x003E, 0x00FE, 0x01FE, 0x07FE, 0x1FFE,
+};
+
+static const uint8_t huff_icc_dt_bits[] = {
+    14, 13, 11, 9, 7, 5, 3, 1, 2, 4, 6, 8, 10, 12, 14,
+};
+
+static const uint16_t huff_icc_dt_codes[] = {
+    0x3FFE, 0x1FFE, 0x07FE, 0x01FE, 0x007E, 0x001E, 0x0006, 0x0000,
+    0x0002, 0x000E, 0x003E, 0x00FE, 0x03FE, 0x0FFE, 0x3FFF,
+};
+
+static const uint8_t huff_ipd_df_bits[] = {
+    1, 3, 4, 4, 4, 4, 4, 4,
+};
+
+static const uint8_t huff_ipd_df_codes[] = {
+    0x01, 0x00, 0x06, 0x04, 0x02, 0x03, 0x05, 0x07,
+};
+
+static const uint8_t huff_ipd_dt_bits[] = {
+    1, 3, 4, 5, 5, 4, 4, 3,
+};
+
+static const uint8_t huff_ipd_dt_codes[] = {
+    0x01, 0x02, 0x02, 0x03, 0x02, 0x00, 0x03, 0x03,
+};
+
+static const uint8_t huff_opd_df_bits[] = {
+    1, 3, 4, 4, 5, 5, 4, 3,
+};
+
+static const uint8_t huff_opd_df_codes[] = {
+    0x01, 0x01, 0x06, 0x04, 0x0F, 0x0E, 0x05, 0x00,
+};
+
+static const uint8_t huff_opd_dt_bits[] = {
+    1, 3, 4, 5, 5, 4, 4, 3,
+};
+
+static const uint8_t huff_opd_dt_codes[] = {
+    0x01, 0x02, 0x01, 0x07, 0x06, 0x00, 0x02, 0x03,
+};
+
+static const int8_t huff_offset[] = {
+    30, 30,
+    14, 14,
+    7, 7,
+    0, 0,
+    0, 0,
+};
+
+///Table 8.48
+static const int8_t k_to_i_20[] = {
+     1,  0,  0,  1,  2,  3,  4,  5,  6,  7,  8,  9, 10, 11, 12, 13, 14, 14, 15,
+    15, 15, 16, 16, 16, 16, 17, 17, 17, 17, 17, 18, 18, 18, 18, 18, 18, 18, 18,
+    18, 18, 18, 18, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19,
+    19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19
+};
+///Table 8.49
+static const int8_t k_to_i_34[] = {
+     0,  1,  2,  3,  4,  5,  6,  6,  7,  2,  1,  0, 10, 10,  4,  5,  6,  7,  8,
+     9, 10, 11, 12,  9, 14, 11, 12, 13, 14, 15, 16, 13, 16, 17, 18, 19, 20, 21,
+    22, 22, 23, 23, 24, 24, 25, 25, 26, 26, 27, 27, 27, 28, 28, 28, 29, 29, 29,
+    30, 30, 30, 31, 31, 31, 31, 32, 32, 32, 32, 33, 33, 33, 33, 33, 33, 33, 33,
+    33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33
+};
+
+static const INTFLOAT g1_Q2[] = {
+    Q31(0.0f),  Q31(0.01899487526049f), Q31(0.0f), Q31(-0.07293139167538f),
+    Q31(0.0f),  Q31(0.30596630545168f), Q31(0.5f)
+};

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacpsdsp.asm

@@ -0,0 +1,487 @@
+;******************************************************************************
+;* SIMD optimized MPEG-4 Parametric Stereo decoding functions
+;*
+;* Copyright (C) 2015 James Almer
+;*
+;* This file is part of FFmpeg.
+;*
+;* FFmpeg is free software; you can redistribute it and/or
+;* modify it under the terms of the GNU Lesser General Public
+;* License as published by the Free Software Foundation; either
+;* version 2.1 of the License, or (at your option) any later version.
+;*
+;* FFmpeg is distributed in the hope that it will be useful,
+;* but WITHOUT ANY WARRANTY; without even the implied warranty of
+;* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+;* Lesser General Public License for more details.
+;*
+;* You should have received a copy of the GNU Lesser General Public
+;* License along with FFmpeg; if not, write to the Free Software
+;* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+;******************************************************************************
+
+%include "libavutil/x86/x86util.asm"
+
+SECTION_RODATA
+
+ps_p1m1p1m1: dd 0, 0x80000000, 0, 0x80000000
+
+SECTION .text
+
+;*************************************************************************
+;void ff_ps_add_squares_<opt>(float *dst, const float (*src)[2], int n);
+;*************************************************************************
+%macro PS_ADD_SQUARES 1
+cglobal ps_add_squares, 3, 3, %1, dst, src, n
+    shl    nd, 3
+    add  srcq, nq
+    neg    nq
+
+align 16
+.loop:
+    movaps m0, [srcq+nq]
+    movaps m1, [srcq+nq+mmsize]
+    mulps  m0, m0
+    mulps  m1, m1
+    HADDPS m0, m1, m2
+    addps  m0, [dstq]
+    movaps [dstq], m0
+    add  dstq, mmsize
+    add    nq, mmsize*2
+    jl .loop
+    REP_RET
+%endmacro
+
+INIT_XMM sse
+PS_ADD_SQUARES 2
+INIT_XMM sse3
+PS_ADD_SQUARES 3
+
+;*******************************************************************
+;void ff_ps_mul_pair_single_sse(float (*dst)[2], float (*src0)[2],
+;                                   float *src1, int n);
+;*******************************************************************
+INIT_XMM sse
+cglobal ps_mul_pair_single, 4, 4, 4, dst, src1, src2, n
+    shl      nd, 3
+    add   src1q, nq
+    add    dstq, nq
+    neg      nq
+
+align 16
+.loop:
+    movu     m0, [src1q+nq]
+    movu     m1, [src1q+nq+mmsize]
+    mova     m2, [src2q]
+    mova     m3, m2
+    unpcklps m2, m2
+    unpckhps m3, m3
+    mulps    m0, m2
+    mulps    m1, m3
+    mova [dstq+nq], m0
+    mova [dstq+nq+mmsize], m1
+    add   src2q, mmsize
+    add      nq, mmsize*2
+    jl .loop
+    REP_RET
+
+;***********************************************************************
+;void ff_ps_stereo_interpolate_sse3(float (*l)[2], float (*r)[2],
+;                                   float h[2][4], float h_step[2][4],
+;                                   int len);
+;***********************************************************************
+INIT_XMM sse3
+cglobal ps_stereo_interpolate, 5, 5, 6, l, r, h, h_step, n
+    movaps   m0, [hq]
+    movaps   m1, [h_stepq]
+    unpcklps m4, m0, m0
+    unpckhps m0, m0
+    unpcklps m5, m1, m1
+    unpckhps m1, m1
+    shl      nd, 3
+    add      lq, nq
+    add      rq, nq
+    neg      nq
+
+align 16
+.loop:
+    addps    m4, m5
+    addps    m0, m1
+    movddup  m2, [lq+nq]
+    movddup  m3, [rq+nq]
+    mulps    m2, m4
+    mulps    m3, m0
+    addps    m2, m3
+    movsd  [lq+nq], m2
+    movhps [rq+nq], m2
+    add      nq, 8
+    jl .loop
+    REP_RET
+
+;***************************************************************************
+;void ps_stereo_interpolate_ipdopd_sse3(float (*l)[2], float (*r)[2],
+;                                       float h[2][4], float h_step[2][4],
+;                                       int len);
+;***************************************************************************
+INIT_XMM sse3
+cglobal ps_stereo_interpolate_ipdopd, 5, 5, 10, l, r, h, h_step, n
+    movaps   m0, [hq]
+    movaps   m1, [hq+mmsize]
+%if ARCH_X86_64
+    movaps   m8, [h_stepq]
+    movaps   m9, [h_stepq+mmsize]
+    %define  H_STEP0 m8
+    %define  H_STEP1 m9
+%else
+    %define  H_STEP0 [h_stepq]
+    %define  H_STEP1 [h_stepq+mmsize]
+%endif
+    shl      nd, 3
+    add      lq, nq
+    add      rq, nq
+    neg      nq
+
+align 16
+.loop:
+    addps    m0, H_STEP0
+    addps    m1, H_STEP1
+    movddup  m2, [lq+nq]
+    movddup  m3, [rq+nq]
+    shufps   m4, m2, m2, q2301
+    shufps   m5, m3, m3, q2301
+    unpcklps m6, m0, m0
+    unpckhps m7, m0, m0
+    mulps    m2, m6
+    mulps    m3, m7
+    unpcklps m6, m1, m1
+    unpckhps m7, m1, m1
+    mulps    m4, m6
+    mulps    m5, m7
+    addps    m2, m3
+    addsubps m2, m4
+    addsubps m2, m5
+    movsd  [lq+nq], m2
+    movhps [rq+nq], m2
+    add      nq, 8
+    jl .loop
+    REP_RET
+
+;**********************************************************
+;void ps_hybrid_analysis_ileave_sse(float out[2][38][64],
+;                                   float (*in)[32][2],
+;                                   int i, int len)
+;**********************************************************
+INIT_XMM sse
+cglobal ps_hybrid_analysis_ileave, 3, 7, 5, out, in, i, len, in0, in1, tmp
+    movsxdifnidn        iq, id
+    mov               lend, 32 << 3
+    lea                inq, [inq+iq*4]
+    mov               tmpd, id
+    shl               tmpd, 8
+    add               outq, tmpq
+    mov               tmpd, 64
+    sub               tmpd, id
+    mov                 id, tmpd
+
+    test                id, 1
+    jne .loop4
+    test                id, 2
+    jne .loop8
+
+align 16
+.loop16:
+    mov               in0q, inq
+    mov               in1q, 38*64*4
+    add               in1q, in0q
+    mov               tmpd, lend
+
+.inner_loop16:
+    movaps              m0, [in0q]
+    movaps              m1, [in1q]
+    movaps              m2, [in0q+lenq]
+    movaps              m3, [in1q+lenq]
+    TRANSPOSE4x4PS 0, 1, 2, 3, 4
+    movaps          [outq], m0
+    movaps     [outq+lenq], m1
+    movaps   [outq+lenq*2], m2
+    movaps [outq+3*32*2*4], m3
+    lea               in0q, [in0q+lenq*2]
+    lea               in1q, [in1q+lenq*2]
+    add               outq, mmsize
+    sub               tmpd, mmsize
+    jg .inner_loop16
+    add                inq, 16
+    add               outq, 3*32*2*4
+    sub                 id, 4
+    jg .loop16
+    RET
+
+align 16
+.loop8:
+    mov               in0q, inq
+    mov               in1q, 38*64*4
+    add               in1q, in0q
+    mov               tmpd, lend
+
+.inner_loop8:
+    movlps              m0, [in0q]
+    movlps              m1, [in1q]
+    movhps              m0, [in0q+lenq]
+    movhps              m1, [in1q+lenq]
+    SBUTTERFLYPS 0, 1, 2
+    SBUTTERFLYPD 0, 1, 2
+    movaps          [outq], m0
+    movaps     [outq+lenq], m1
+    lea               in0q, [in0q+lenq*2]
+    lea               in1q, [in1q+lenq*2]
+    add               outq, mmsize
+    sub               tmpd, mmsize
+    jg .inner_loop8
+    add                inq, 8
+    add               outq, lenq
+    sub                 id, 2
+    jg .loop16
+    RET
+
+align 16
+.loop4:
+    mov               in0q, inq
+    mov               in1q, 38*64*4
+    add               in1q, in0q
+    mov               tmpd, lend
+
+.inner_loop4:
+    movss               m0, [in0q]
+    movss               m1, [in1q]
+    movss               m2, [in0q+lenq]
+    movss               m3, [in1q+lenq]
+    movlhps             m0, m1
+    movlhps             m2, m3
+    shufps              m0, m2, q2020
+    movaps          [outq], m0
+    lea               in0q, [in0q+lenq*2]
+    lea               in1q, [in1q+lenq*2]
+    add               outq, mmsize
+    sub               tmpd, mmsize
+    jg .inner_loop4
+    add                inq, 4
+    sub                 id, 1
+    test                id, 2
+    jne .loop8
+    cmp                 id, 4
+    jge .loop16
+    RET
+
+;***********************************************************
+;void ps_hybrid_synthesis_deint_sse4(float out[2][38][64],
+;                                    float (*in)[32][2],
+;                                    int i, int len)
+;***********************************************************
+%macro HYBRID_SYNTHESIS_DEINT 0
+cglobal ps_hybrid_synthesis_deint, 3, 7, 5, out, in, i, len, out0, out1, tmp
+%if cpuflag(sse4)
+%define MOVH movsd
+%else
+%define MOVH movlps
+%endif
+    movsxdifnidn        iq, id
+    mov               lend, 32 << 3
+    lea               outq, [outq+iq*4]
+    mov               tmpd, id
+    shl               tmpd, 8
+    add                inq, tmpq
+    mov               tmpd, 64
+    sub               tmpd, id
+    mov                 id, tmpd
+
+    test                id, 1
+    jne .loop4
+    test                id, 2
+    jne .loop8
+
+align 16
+.loop16:
+    mov              out0q, outq
+    mov              out1q, 38*64*4
+    add              out1q, out0q
+    mov               tmpd, lend
+
+.inner_loop16:
+    movaps              m0, [inq]
+    movaps              m1, [inq+lenq]
+    movaps              m2, [inq+lenq*2]
+    movaps              m3, [inq+3*32*2*4]
+    TRANSPOSE4x4PS 0, 1, 2, 3, 4
+    movaps         [out0q], m0
+    movaps         [out1q], m1
+    movaps    [out0q+lenq], m2
+    movaps    [out1q+lenq], m3
+    lea              out0q, [out0q+lenq*2]
+    lea              out1q, [out1q+lenq*2]
+    add                inq, mmsize
+    sub               tmpd, mmsize
+    jg .inner_loop16
+    add               outq, 16
+    add                inq, 3*32*2*4
+    sub                 id, 4
+    jg .loop16
+    RET
+
+align 16
+.loop8:
+    mov              out0q, outq
+    mov              out1q, 38*64*4
+    add              out1q, out0q
+    mov               tmpd, lend
+
+.inner_loop8:
+    movaps              m0, [inq]
+    movaps              m1, [inq+lenq]
+    SBUTTERFLYPS 0, 1, 2
+    SBUTTERFLYPD 0, 1, 2
+    MOVH           [out0q], m0
+    MOVH           [out1q], m1
+    movhps    [out0q+lenq], m0
+    movhps    [out1q+lenq], m1
+    lea              out0q, [out0q+lenq*2]
+    lea              out1q, [out1q+lenq*2]
+    add                inq, mmsize
+    sub               tmpd, mmsize
+    jg .inner_loop8
+    add               outq, 8
+    add                inq, lenq
+    sub                 id, 2
+    jg .loop16
+    RET
+
+align 16
+.loop4:
+    mov              out0q, outq
+    mov              out1q, 38*64*4
+    add              out1q, out0q
+    mov               tmpd, lend
+
+.inner_loop4:
+    movaps              m0, [inq]
+    movss          [out0q], m0
+%if cpuflag(sse4)
+    extractps      [out1q], m0, 1
+    extractps [out0q+lenq], m0, 2
+    extractps [out1q+lenq], m0, 3
+%else
+    movhlps             m1, m0
+    movss     [out0q+lenq], m1
+    shufps              m0, m0, 0xb1
+    movss          [out1q], m0
+    movhlps             m1, m0
+    movss     [out1q+lenq], m1
+%endif
+    lea              out0q, [out0q+lenq*2]
+    lea              out1q, [out1q+lenq*2]
+    add                inq, mmsize
+    sub               tmpd, mmsize
+    jg .inner_loop4
+    add               outq, 4
+    sub                 id, 1
+    test                id, 2
+    jne .loop8
+    cmp                 id, 4
+    jge .loop16
+    RET
+%endmacro
+
+INIT_XMM sse
+HYBRID_SYNTHESIS_DEINT
+INIT_XMM sse4
+HYBRID_SYNTHESIS_DEINT
+
+;*******************************************************************
+;void ff_ps_hybrid_analysis_<opt>(float (*out)[2], float (*in)[2],
+;                                 const float (*filter)[8][2],
+;                                 ptrdiff_t stride, int n);
+;*******************************************************************
+%macro PS_HYBRID_ANALYSIS_LOOP 3
+    movu     %1, [inq+mmsize*%3]
+    movu     m1, [inq+mmsize*(5-%3)+8]
+%if cpuflag(sse3)
+    pshufd   %2, %1, q2301
+    pshufd   m4, m1, q0123
+    pshufd   m1, m1, q1032
+    pshufd   m2, [filterq+nq+mmsize*%3], q2301
+    addsubps %2, m4
+    addsubps %1, m1
+%else
+    mova     m2, [filterq+nq+mmsize*%3]
+    mova     %2, %1
+    mova     m4, m1
+    shufps   %2, %2, q2301
+    shufps   m4, m4, q0123
+    shufps   m1, m1, q1032
+    shufps   m2, m2, q2301
+    xorps    m4, m7
+    xorps    m1, m7
+    subps    %2, m4
+    subps    %1, m1
+%endif
+    mulps    %2, m2
+    mulps    %1, m2
+%if %3
+    addps    m3, %2
+    addps    m0, %1
+%endif
+%endmacro
+
+%macro PS_HYBRID_ANALYSIS 0
+cglobal ps_hybrid_analysis, 5, 5, 8, out, in, filter, stride, n
+%if cpuflag(sse3)
+%define MOVH movsd
+%else
+%define MOVH movlps
+%endif
+    shl strideq, 3
+    shl nd, 6
+    add filterq, nq
+    neg nq
+    mova m7, [ps_p1m1p1m1]
+
+align 16
+.loop:
+    PS_HYBRID_ANALYSIS_LOOP m0, m3, 0
+    PS_HYBRID_ANALYSIS_LOOP m5, m6, 1
+    PS_HYBRID_ANALYSIS_LOOP m5, m6, 2
+
+%if cpuflag(sse3)
+    pshufd   m3, m3, q2301
+    xorps    m0, m7
+    hsubps   m3, m0
+    pshufd   m1, m3, q0020
+    pshufd   m3, m3, q0031
+    addps    m1, m3
+    movsd    m2, [inq+6*8]
+%else
+    mova     m1, m3
+    mova     m2, m0
+    shufps   m1, m1, q2301
+    shufps   m2, m2, q2301
+    subps    m1, m3
+    addps    m2, m0
+    unpcklps m3, m1, m2
+    unpckhps m1, m2
+    addps    m1, m3
+    movu     m2, [inq+6*8] ; faster than movlps and no risk of overread
+%endif
+    movss    m3, [filterq+nq+8*6]
+    SPLATD   m3
+    mulps    m2, m3
+    addps    m1, m2
+    MOVH [outq], m1
+    add    outq, strideq
+    add      nq, 64
+    jl .loop
+    REP_RET
+%endmacro
+
+INIT_XMM sse
+PS_HYBRID_ANALYSIS
+INIT_XMM sse3
+PS_HYBRID_ANALYSIS

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacpsdsp.h

@@ -0,0 +1,60 @@
+/*
+ * Copyright (c) 2012 Mans Rullgard
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AACPSDSP_H
+#define AVCODEC_AACPSDSP_H
+
+#include <stddef.h>
+
+#include "aac_defines.h"
+
+#define PS_QMF_TIME_SLOTS 32
+#define PS_AP_LINKS 3
+#define PS_MAX_AP_DELAY 5
+
+typedef struct PSDSPContext {
+    void (*add_squares)(INTFLOAT *dst, const INTFLOAT (*src)[2], int n);
+    void (*mul_pair_single)(INTFLOAT (*dst)[2], INTFLOAT (*src0)[2], INTFLOAT *src1,
+                            int n);
+    void (*hybrid_analysis)(INTFLOAT (*out)[2], INTFLOAT (*in)[2],
+                            const INTFLOAT (*filter)[8][2],
+                            ptrdiff_t stride, int n);
+    void (*hybrid_analysis_ileave)(INTFLOAT (*out)[32][2], INTFLOAT L[2][38][64],
+                                   int i, int len);
+    void (*hybrid_synthesis_deint)(INTFLOAT out[2][38][64], INTFLOAT (*in)[32][2],
+                                   int i, int len);
+    void (*decorrelate)(INTFLOAT (*out)[2], INTFLOAT (*delay)[2],
+                        INTFLOAT (*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2],
+                        const INTFLOAT phi_fract[2], const INTFLOAT (*Q_fract)[2],
+                        const INTFLOAT *transient_gain,
+                        INTFLOAT g_decay_slope,
+                        int len);
+    void (*stereo_interpolate[2])(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
+                                  INTFLOAT h[2][4], INTFLOAT h_step[2][4],
+                                  int len);
+} PSDSPContext;
+
+void AAC_RENAME(ff_psdsp_init)(PSDSPContext *s);
+void ff_psdsp_init_arm(PSDSPContext *s);
+void ff_psdsp_init_aarch64(PSDSPContext *s);
+void ff_psdsp_init_mips(PSDSPContext *s);
+void ff_psdsp_init_x86(PSDSPContext *s);
+
+#endif /* AVCODEC_AACPSDSP_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacpsdsp_fixed.c

@@ -0,0 +1,23 @@
+/*
+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#define USE_FIXED 1
+
+#include "aacpsdsp_template.c"

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacpsdsp_float.c

@@ -0,0 +1,23 @@
+/*
+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#define USE_FIXED 0
+
+#include "aacpsdsp_template.c"

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacpsdsp_init.c

@@ -0,0 +1,72 @@
+/*
+ * SIMD optimized MPEG-4 Parametric Stereo decoding functions
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stddef.h>
+
+#include "config.h"
+
+#include "libavutil/x86/cpu.h"
+#include "libavutil/attributes.h"
+#include "libavcodec/aacpsdsp.h"
+
+void ff_ps_add_squares_sse  (float *dst, const float (*src)[2], int n);
+void ff_ps_add_squares_sse3 (float *dst, const float (*src)[2], int n);
+void ff_ps_mul_pair_single_sse (float (*dst)[2], float (*src0)[2],
+                                float *src1, int n);
+void ff_ps_hybrid_analysis_sse (float (*out)[2], float (*in)[2],
+                                const float (*filter)[8][2],
+                                ptrdiff_t stride, int n);
+void ff_ps_hybrid_analysis_sse3(float (*out)[2], float (*in)[2],
+                                const float (*filter)[8][2],
+                                ptrdiff_t stride, int n);
+void ff_ps_stereo_interpolate_sse3(float (*l)[2], float (*r)[2],
+                                   float h[2][4], float h_step[2][4],
+                                   int len);
+void ff_ps_stereo_interpolate_ipdopd_sse3(float (*l)[2], float (*r)[2],
+                                          float h[2][4], float h_step[2][4],
+                                          int len);
+void ff_ps_hybrid_synthesis_deint_sse(float out[2][38][64], float (*in)[32][2],
+                                      int i, int len);
+void ff_ps_hybrid_synthesis_deint_sse4(float out[2][38][64], float (*in)[32][2],
+                                       int i, int len);
+void ff_ps_hybrid_analysis_ileave_sse(float (*out)[32][2], float L[2][38][64],
+                                      int i, int len);
+
+av_cold void ff_psdsp_init_x86(PSDSPContext *s)
+{
+    int cpu_flags = av_get_cpu_flags();
+
+    if (EXTERNAL_SSE(cpu_flags)) {
+        s->add_squares            = ff_ps_add_squares_sse;
+        s->mul_pair_single        = ff_ps_mul_pair_single_sse;
+        s->hybrid_analysis_ileave = ff_ps_hybrid_analysis_ileave_sse;
+        s->hybrid_synthesis_deint = ff_ps_hybrid_synthesis_deint_sse;
+        s->hybrid_analysis        = ff_ps_hybrid_analysis_sse;
+    }
+    if (EXTERNAL_SSE3(cpu_flags)) {
+        s->add_squares            = ff_ps_add_squares_sse3;
+        s->stereo_interpolate[0]  = ff_ps_stereo_interpolate_sse3;
+        s->stereo_interpolate[1]  = ff_ps_stereo_interpolate_ipdopd_sse3;
+        s->hybrid_analysis        = ff_ps_hybrid_analysis_sse3;
+    }
+    if (EXTERNAL_SSE4(cpu_flags)) {
+        s->hybrid_synthesis_deint = ff_ps_hybrid_synthesis_deint_sse4;
+    }
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacpsdsp_template.c

@@ -0,0 +1,233 @@
+/*
+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Note: Rounding-to-nearest used unless otherwise stated
+ *
+ */
+#include <stdint.h>
+
+#include "config.h"
+#include "libavutil/attributes.h"
+#include "aacpsdsp.h"
+
+static void ps_add_squares_c(INTFLOAT *dst, const INTFLOAT (*src)[2], int n)
+{
+    int i;
+    for (i = 0; i < n; i++)
+        dst[i] += (UINTFLOAT)AAC_MADD28(src[i][0], src[i][0], src[i][1], src[i][1]);
+}
+
+static void ps_mul_pair_single_c(INTFLOAT (*dst)[2], INTFLOAT (*src0)[2], INTFLOAT *src1,
+                                 int n)
+{
+    int i;
+    for (i = 0; i < n; i++) {
+        dst[i][0] = AAC_MUL16(src0[i][0], src1[i]);
+        dst[i][1] = AAC_MUL16(src0[i][1], src1[i]);
+    }
+}
+
+static void ps_hybrid_analysis_c(INTFLOAT (*out)[2], INTFLOAT (*in)[2],
+                                 const INTFLOAT (*filter)[8][2],
+                                 ptrdiff_t stride, int n)
+{
+    int i, j;
+
+    for (i = 0; i < n; i++) {
+        INT64FLOAT sum_re = (INT64FLOAT)filter[i][6][0] * in[6][0];
+        INT64FLOAT sum_im = (INT64FLOAT)filter[i][6][0] * in[6][1];
+
+        for (j = 0; j < 6; j++) {
+            INT64FLOAT in0_re = in[j][0];
+            INT64FLOAT in0_im = in[j][1];
+            INT64FLOAT in1_re = in[12-j][0];
+            INT64FLOAT in1_im = in[12-j][1];
+            sum_re += (INT64FLOAT)filter[i][j][0] * (in0_re + in1_re) -
+                      (INT64FLOAT)filter[i][j][1] * (in0_im - in1_im);
+            sum_im += (INT64FLOAT)filter[i][j][0] * (in0_im + in1_im) +
+                      (INT64FLOAT)filter[i][j][1] * (in0_re - in1_re);
+        }
+#if USE_FIXED
+        out[i * stride][0] = (int)((sum_re + 0x40000000) >> 31);
+        out[i * stride][1] = (int)((sum_im + 0x40000000) >> 31);
+#else
+        out[i * stride][0] = sum_re;
+        out[i * stride][1] = sum_im;
+#endif /* USE_FIXED */
+    }
+}
+
+static void ps_hybrid_analysis_ileave_c(INTFLOAT (*out)[32][2], INTFLOAT L[2][38][64],
+                                      int i, int len)
+{
+    int j;
+
+    for (; i < 64; i++) {
+        for (j = 0; j < len; j++) {
+            out[i][j][0] = L[0][j][i];
+            out[i][j][1] = L[1][j][i];
+        }
+    }
+}
+
+static void ps_hybrid_synthesis_deint_c(INTFLOAT out[2][38][64],
+                                      INTFLOAT (*in)[32][2],
+                                      int i, int len)
+{
+    int n;
+
+    for (; i < 64; i++) {
+        for (n = 0; n < len; n++) {
+            out[0][n][i] = in[i][n][0];
+            out[1][n][i] = in[i][n][1];
+        }
+    }
+}
+
+static void ps_decorrelate_c(INTFLOAT (*out)[2], INTFLOAT (*delay)[2],
+                             INTFLOAT (*ap_delay)[PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2],
+                             const INTFLOAT phi_fract[2], const INTFLOAT (*Q_fract)[2],
+                             const INTFLOAT *transient_gain,
+                             INTFLOAT g_decay_slope,
+                             int len)
+{
+    static const INTFLOAT a[] = { Q31(0.65143905753106f),
+                               Q31(0.56471812200776f),
+                               Q31(0.48954165955695f) };
+    INTFLOAT ag[PS_AP_LINKS];
+    int m, n;
+
+    for (m = 0; m < PS_AP_LINKS; m++)
+        ag[m] = AAC_MUL30(a[m], g_decay_slope);
+
+    for (n = 0; n < len; n++) {
+        INTFLOAT in_re = AAC_MSUB30(delay[n][0], phi_fract[0], delay[n][1], phi_fract[1]);
+        INTFLOAT in_im = AAC_MADD30(delay[n][0], phi_fract[1], delay[n][1], phi_fract[0]);
+        for (m = 0; m < PS_AP_LINKS; m++) {
+            INTFLOAT a_re                = AAC_MUL31(ag[m], in_re);
+            INTFLOAT a_im                = AAC_MUL31(ag[m], in_im);
+            INTFLOAT link_delay_re       = ap_delay[m][n+2-m][0];
+            INTFLOAT link_delay_im       = ap_delay[m][n+2-m][1];
+            INTFLOAT fractional_delay_re = Q_fract[m][0];
+            INTFLOAT fractional_delay_im = Q_fract[m][1];
+            INTFLOAT apd_re = in_re;
+            INTFLOAT apd_im = in_im;
+            in_re = AAC_MSUB30(link_delay_re, fractional_delay_re,
+                    link_delay_im, fractional_delay_im);
+            in_re -= (UINTFLOAT)a_re;
+            in_im = AAC_MADD30(link_delay_re, fractional_delay_im,
+                    link_delay_im, fractional_delay_re);
+            in_im -= (UINTFLOAT)a_im;
+            ap_delay[m][n+5][0] = apd_re + (UINTFLOAT)AAC_MUL31(ag[m], in_re);
+            ap_delay[m][n+5][1] = apd_im + (UINTFLOAT)AAC_MUL31(ag[m], in_im);
+        }
+        out[n][0] = AAC_MUL16(transient_gain[n], in_re);
+        out[n][1] = AAC_MUL16(transient_gain[n], in_im);
+    }
+}
+
+static void ps_stereo_interpolate_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
+                                    INTFLOAT h[2][4], INTFLOAT h_step[2][4],
+                                    int len)
+{
+    INTFLOAT h0 = h[0][0];
+    INTFLOAT h1 = h[0][1];
+    INTFLOAT h2 = h[0][2];
+    INTFLOAT h3 = h[0][3];
+    UINTFLOAT hs0 = h_step[0][0];
+    UINTFLOAT hs1 = h_step[0][1];
+    UINTFLOAT hs2 = h_step[0][2];
+    UINTFLOAT hs3 = h_step[0][3];
+    int n;
+
+    for (n = 0; n < len; n++) {
+        //l is s, r is d
+        INTFLOAT l_re = l[n][0];
+        INTFLOAT l_im = l[n][1];
+        INTFLOAT r_re = r[n][0];
+        INTFLOAT r_im = r[n][1];
+        h0 += hs0;
+        h1 += hs1;
+        h2 += hs2;
+        h3 += hs3;
+        l[n][0] = AAC_MADD30(h0, l_re, h2, r_re);
+        l[n][1] = AAC_MADD30(h0, l_im, h2, r_im);
+        r[n][0] = AAC_MADD30(h1, l_re, h3, r_re);
+        r[n][1] = AAC_MADD30(h1, l_im, h3, r_im);
+    }
+}
+
+static void ps_stereo_interpolate_ipdopd_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
+                                           INTFLOAT h[2][4], INTFLOAT h_step[2][4],
+                                           int len)
+{
+    INTFLOAT h00  = h[0][0],      h10  = h[1][0];
+    INTFLOAT h01  = h[0][1],      h11  = h[1][1];
+    INTFLOAT h02  = h[0][2],      h12  = h[1][2];
+    INTFLOAT h03  = h[0][3],      h13  = h[1][3];
+    UINTFLOAT hs00 = h_step[0][0], hs10 = h_step[1][0];
+    UINTFLOAT hs01 = h_step[0][1], hs11 = h_step[1][1];
+    UINTFLOAT hs02 = h_step[0][2], hs12 = h_step[1][2];
+    UINTFLOAT hs03 = h_step[0][3], hs13 = h_step[1][3];
+    int n;
+
+    for (n = 0; n < len; n++) {
+        //l is s, r is d
+        INTFLOAT l_re = l[n][0];
+        INTFLOAT l_im = l[n][1];
+        INTFLOAT r_re = r[n][0];
+        INTFLOAT r_im = r[n][1];
+        h00 += hs00;
+        h01 += hs01;
+        h02 += hs02;
+        h03 += hs03;
+        h10 += hs10;
+        h11 += hs11;
+        h12 += hs12;
+        h13 += hs13;
+
+        l[n][0] = AAC_MSUB30_V8(h00, l_re, h02, r_re, h10, l_im, h12, r_im);
+        l[n][1] = AAC_MADD30_V8(h00, l_im, h02, r_im, h10, l_re, h12, r_re);
+        r[n][0] = AAC_MSUB30_V8(h01, l_re, h03, r_re, h11, l_im, h13, r_im);
+        r[n][1] = AAC_MADD30_V8(h01, l_im, h03, r_im, h11, l_re, h13, r_re);
+    }
+}
+
+av_cold void AAC_RENAME(ff_psdsp_init)(PSDSPContext *s)
+{
+    s->add_squares            = ps_add_squares_c;
+    s->mul_pair_single        = ps_mul_pair_single_c;
+    s->hybrid_analysis        = ps_hybrid_analysis_c;
+    s->hybrid_analysis_ileave = ps_hybrid_analysis_ileave_c;
+    s->hybrid_synthesis_deint = ps_hybrid_synthesis_deint_c;
+    s->decorrelate            = ps_decorrelate_c;
+    s->stereo_interpolate[0]  = ps_stereo_interpolate_c;
+    s->stereo_interpolate[1]  = ps_stereo_interpolate_ipdopd_c;
+
+#if !USE_FIXED
+    if (ARCH_ARM)
+        ff_psdsp_init_arm(s);
+    if (ARCH_AARCH64)
+        ff_psdsp_init_aarch64(s);
+    if (ARCH_MIPS)
+        ff_psdsp_init_mips(s);
+    if (ARCH_X86)
+        ff_psdsp_init_x86(s);
+#endif /* !USE_FIXED */
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacsbr.c

@@ -0,0 +1,370 @@
+/*
+ * AAC Spectral Band Replication decoding functions
+ * Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
+ * Copyright (c) 2009-2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC Spectral Band Replication decoding functions
+ * @author Robert Swain ( rob opendot cl )
+ */
+#define USE_FIXED 0
+
+#include "aac.h"
+#include "sbr.h"
+#include "aacsbr.h"
+#include "aacsbrdata.h"
+#include "aacsbr_tablegen.h"
+#include "fft.h"
+#include "internal.h"
+#include "aacps.h"
+#include "sbrdsp.h"
+#include "libavutil/internal.h"
+#include "libavutil/libm.h"
+#include "libavutil/avassert.h"
+
+#include <stdint.h>
+#include <float.h>
+#include <math.h>
+
+#if ARCH_MIPS
+#include "mips/aacsbr_mips.h"
+#endif /* ARCH_MIPS */
+
+static VLC vlc_sbr[10];
+static void aacsbr_func_ptr_init(AACSBRContext *c);
+
+static void make_bands(int16_t* bands, int start, int stop, int num_bands)
+{
+    int k, previous, present;
+    float base, prod;
+
+    base = powf((float)stop / start, 1.0f / num_bands);
+    prod = start;
+    previous = start;
+
+    for (k = 0; k < num_bands-1; k++) {
+        prod *= base;
+        present  = lrintf(prod);
+        bands[k] = present - previous;
+        previous = present;
+    }
+    bands[num_bands-1] = stop - previous;
+}
+
+/// Dequantization and stereo decoding (14496-3 sp04 p203)
+static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
+{
+    int k, e;
+    int ch;
+    static const double exp2_tab[2] = {1, M_SQRT2};
+    if (id_aac == TYPE_CPE && sbr->bs_coupling) {
+        int pan_offset = sbr->data[0].bs_amp_res ? 12 : 24;
+        for (e = 1; e <= sbr->data[0].bs_num_env; e++) {
+            for (k = 0; k < sbr->n[sbr->data[0].bs_freq_res[e]]; k++) {
+                float temp1, temp2, fac;
+                if (sbr->data[0].bs_amp_res) {
+                    temp1 = ff_exp2fi(sbr->data[0].env_facs_q[e][k] + 7);
+                    temp2 = ff_exp2fi(pan_offset - sbr->data[1].env_facs_q[e][k]);
+                }
+                else {
+                    temp1 = ff_exp2fi((sbr->data[0].env_facs_q[e][k]>>1) + 7) *
+                            exp2_tab[sbr->data[0].env_facs_q[e][k] & 1];
+                    temp2 = ff_exp2fi((pan_offset - sbr->data[1].env_facs_q[e][k])>>1) *
+                            exp2_tab[(pan_offset - sbr->data[1].env_facs_q[e][k]) & 1];
+                }
+                if (temp1 > 1E20) {
+                    av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
+                    temp1 = 1;
+                }
+                fac   = temp1 / (1.0f + temp2);
+                sbr->data[0].env_facs[e][k] = fac;
+                sbr->data[1].env_facs[e][k] = fac * temp2;
+            }
+        }
+        for (e = 1; e <= sbr->data[0].bs_num_noise; e++) {
+            for (k = 0; k < sbr->n_q; k++) {
+                float temp1 = ff_exp2fi(NOISE_FLOOR_OFFSET - sbr->data[0].noise_facs_q[e][k] + 1);
+                float temp2 = ff_exp2fi(12 - sbr->data[1].noise_facs_q[e][k]);
+                float fac;
+                av_assert0(temp1 <= 1E20);
+                fac = temp1 / (1.0f + temp2);
+                sbr->data[0].noise_facs[e][k] = fac;
+                sbr->data[1].noise_facs[e][k] = fac * temp2;
+            }
+        }
+    } else { // SCE or one non-coupled CPE
+        for (ch = 0; ch < (id_aac == TYPE_CPE) + 1; ch++) {
+            for (e = 1; e <= sbr->data[ch].bs_num_env; e++)
+                for (k = 0; k < sbr->n[sbr->data[ch].bs_freq_res[e]]; k++){
+                    if (sbr->data[ch].bs_amp_res)
+                        sbr->data[ch].env_facs[e][k] = ff_exp2fi(sbr->data[ch].env_facs_q[e][k] + 6);
+                    else
+                        sbr->data[ch].env_facs[e][k] = ff_exp2fi((sbr->data[ch].env_facs_q[e][k]>>1) + 6)
+                                                       * exp2_tab[sbr->data[ch].env_facs_q[e][k] & 1];
+                    if (sbr->data[ch].env_facs[e][k] > 1E20) {
+                        av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
+                        sbr->data[ch].env_facs[e][k] = 1;
+                    }
+                }
+
+            for (e = 1; e <= sbr->data[ch].bs_num_noise; e++)
+                for (k = 0; k < sbr->n_q; k++)
+                    sbr->data[ch].noise_facs[e][k] =
+                        ff_exp2fi(NOISE_FLOOR_OFFSET - sbr->data[ch].noise_facs_q[e][k]);
+        }
+    }
+}
+
+/** High Frequency Generation (14496-3 sp04 p214+) and Inverse Filtering
+ * (14496-3 sp04 p214)
+ * Warning: This routine does not seem numerically stable.
+ */
+static void sbr_hf_inverse_filter(SBRDSPContext *dsp,
+                                  float (*alpha0)[2], float (*alpha1)[2],
+                                  const float X_low[32][40][2], int k0)
+{
+    int k;
+    for (k = 0; k < k0; k++) {
+        LOCAL_ALIGNED_16(float, phi, [3], [2][2]);
+        float dk;
+
+        dsp->autocorrelate(X_low[k], phi);
+
+        dk =  phi[2][1][0] * phi[1][0][0] -
+             (phi[1][1][0] * phi[1][1][0] + phi[1][1][1] * phi[1][1][1]) / 1.000001f;
+
+        if (!dk) {
+            alpha1[k][0] = 0;
+            alpha1[k][1] = 0;
+        } else {
+            float temp_real, temp_im;
+            temp_real = phi[0][0][0] * phi[1][1][0] -
+                        phi[0][0][1] * phi[1][1][1] -
+                        phi[0][1][0] * phi[1][0][0];
+            temp_im   = phi[0][0][0] * phi[1][1][1] +
+                        phi[0][0][1] * phi[1][1][0] -
+                        phi[0][1][1] * phi[1][0][0];
+
+            alpha1[k][0] = temp_real / dk;
+            alpha1[k][1] = temp_im   / dk;
+        }
+
+        if (!phi[1][0][0]) {
+            alpha0[k][0] = 0;
+            alpha0[k][1] = 0;
+        } else {
+            float temp_real, temp_im;
+            temp_real = phi[0][0][0] + alpha1[k][0] * phi[1][1][0] +
+                                       alpha1[k][1] * phi[1][1][1];
+            temp_im   = phi[0][0][1] + alpha1[k][1] * phi[1][1][0] -
+                                       alpha1[k][0] * phi[1][1][1];
+
+            alpha0[k][0] = -temp_real / phi[1][0][0];
+            alpha0[k][1] = -temp_im   / phi[1][0][0];
+        }
+
+        if (alpha1[k][0] * alpha1[k][0] + alpha1[k][1] * alpha1[k][1] >= 16.0f ||
+           alpha0[k][0] * alpha0[k][0] + alpha0[k][1] * alpha0[k][1] >= 16.0f) {
+            alpha1[k][0] = 0;
+            alpha1[k][1] = 0;
+            alpha0[k][0] = 0;
+            alpha0[k][1] = 0;
+        }
+    }
+}
+
+/// Chirp Factors (14496-3 sp04 p214)
+static void sbr_chirp(SpectralBandReplication *sbr, SBRData *ch_data)
+{
+    int i;
+    float new_bw;
+    static const float bw_tab[] = { 0.0f, 0.75f, 0.9f, 0.98f };
+
+    for (i = 0; i < sbr->n_q; i++) {
+        if (ch_data->bs_invf_mode[0][i] + ch_data->bs_invf_mode[1][i] == 1) {
+            new_bw = 0.6f;
+        } else
+            new_bw = bw_tab[ch_data->bs_invf_mode[0][i]];
+
+        if (new_bw < ch_data->bw_array[i]) {
+            new_bw = 0.75f    * new_bw + 0.25f    * ch_data->bw_array[i];
+        } else
+            new_bw = 0.90625f * new_bw + 0.09375f * ch_data->bw_array[i];
+        ch_data->bw_array[i] = new_bw < 0.015625f ? 0.0f : new_bw;
+    }
+}
+
+/**
+ * Calculation of levels of additional HF signal components (14496-3 sp04 p219)
+ * and Calculation of gain (14496-3 sp04 p219)
+ */
+static void sbr_gain_calc(AACContext *ac, SpectralBandReplication *sbr,
+                          SBRData *ch_data, const int e_a[2])
+{
+    int e, k, m;
+    // max gain limits : -3dB, 0dB, 3dB, inf dB (limiter off)
+    static const float limgain[4] = { 0.70795, 1.0, 1.41254, 10000000000 };
+
+    for (e = 0; e < ch_data->bs_num_env; e++) {
+        int delta = !((e == e_a[1]) || (e == e_a[0]));
+        for (k = 0; k < sbr->n_lim; k++) {
+            float gain_boost, gain_max;
+            float sum[2] = { 0.0f, 0.0f };
+            for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+                const float temp = sbr->e_origmapped[e][m] / (1.0f + sbr->q_mapped[e][m]);
+                sbr->q_m[e][m] = sqrtf(temp * sbr->q_mapped[e][m]);
+                sbr->s_m[e][m] = sqrtf(temp * ch_data->s_indexmapped[e + 1][m]);
+                if (!sbr->s_mapped[e][m]) {
+                    sbr->gain[e][m] = sqrtf(sbr->e_origmapped[e][m] /
+                                            ((1.0f + sbr->e_curr[e][m]) *
+                                             (1.0f + sbr->q_mapped[e][m] * delta)));
+                } else {
+                    sbr->gain[e][m] = sqrtf(sbr->e_origmapped[e][m] * sbr->q_mapped[e][m] /
+                                            ((1.0f + sbr->e_curr[e][m]) *
+                                             (1.0f + sbr->q_mapped[e][m])));
+                }
+                sbr->gain[e][m] += FLT_MIN;
+            }
+            for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+                sum[0] += sbr->e_origmapped[e][m];
+                sum[1] += sbr->e_curr[e][m];
+            }
+            gain_max = limgain[sbr->bs_limiter_gains] * sqrtf((FLT_EPSILON + sum[0]) / (FLT_EPSILON + sum[1]));
+            gain_max = FFMIN(100000.f, gain_max);
+            for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+                float q_m_max   = sbr->q_m[e][m] * gain_max / sbr->gain[e][m];
+                sbr->q_m[e][m]  = FFMIN(sbr->q_m[e][m], q_m_max);
+                sbr->gain[e][m] = FFMIN(sbr->gain[e][m], gain_max);
+            }
+            sum[0] = sum[1] = 0.0f;
+            for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+                sum[0] += sbr->e_origmapped[e][m];
+                sum[1] += sbr->e_curr[e][m] * sbr->gain[e][m] * sbr->gain[e][m]
+                          + sbr->s_m[e][m] * sbr->s_m[e][m]
+                          + (delta && !sbr->s_m[e][m]) * sbr->q_m[e][m] * sbr->q_m[e][m];
+            }
+            gain_boost = sqrtf((FLT_EPSILON + sum[0]) / (FLT_EPSILON + sum[1]));
+            gain_boost = FFMIN(1.584893192f, gain_boost);
+            for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+                sbr->gain[e][m] *= gain_boost;
+                sbr->q_m[e][m]  *= gain_boost;
+                sbr->s_m[e][m]  *= gain_boost;
+            }
+        }
+    }
+}
+
+/// Assembling HF Signals (14496-3 sp04 p220)
+static void sbr_hf_assemble(float Y1[38][64][2],
+                            const float X_high[64][40][2],
+                            SpectralBandReplication *sbr, SBRData *ch_data,
+                            const int e_a[2])
+{
+    int e, i, j, m;
+    const int h_SL = 4 * !sbr->bs_smoothing_mode;
+    const int kx = sbr->kx[1];
+    const int m_max = sbr->m[1];
+    static const float h_smooth[5] = {
+        0.33333333333333,
+        0.30150283239582,
+        0.21816949906249,
+        0.11516383427084,
+        0.03183050093751,
+    };
+    float (*g_temp)[48] = ch_data->g_temp, (*q_temp)[48] = ch_data->q_temp;
+    int indexnoise = ch_data->f_indexnoise;
+    int indexsine  = ch_data->f_indexsine;
+
+    if (sbr->reset) {
+        for (i = 0; i < h_SL; i++) {
+            memcpy(g_temp[i + 2*ch_data->t_env[0]], sbr->gain[0], m_max * sizeof(sbr->gain[0][0]));
+            memcpy(q_temp[i + 2*ch_data->t_env[0]], sbr->q_m[0],  m_max * sizeof(sbr->q_m[0][0]));
+        }
+    } else if (h_SL) {
+        for (i = 0; i < 4; i++) {
+            memcpy(g_temp[i + 2 * ch_data->t_env[0]],
+                   g_temp[i + 2 * ch_data->t_env_num_env_old],
+                   sizeof(g_temp[0]));
+            memcpy(q_temp[i + 2 * ch_data->t_env[0]],
+                   q_temp[i + 2 * ch_data->t_env_num_env_old],
+                   sizeof(q_temp[0]));
+        }
+    }
+
+    for (e = 0; e < ch_data->bs_num_env; e++) {
+        for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
+            memcpy(g_temp[h_SL + i], sbr->gain[e], m_max * sizeof(sbr->gain[0][0]));
+            memcpy(q_temp[h_SL + i], sbr->q_m[e],  m_max * sizeof(sbr->q_m[0][0]));
+        }
+    }
+
+    for (e = 0; e < ch_data->bs_num_env; e++) {
+        for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
+            LOCAL_ALIGNED_16(float, g_filt_tab, [48]);
+            LOCAL_ALIGNED_16(float, q_filt_tab, [48]);
+            float *g_filt, *q_filt;
+
+            if (h_SL && e != e_a[0] && e != e_a[1]) {
+                g_filt = g_filt_tab;
+                q_filt = q_filt_tab;
+                for (m = 0; m < m_max; m++) {
+                    const int idx1 = i + h_SL;
+                    g_filt[m] = 0.0f;
+                    q_filt[m] = 0.0f;
+                    for (j = 0; j <= h_SL; j++) {
+                        g_filt[m] += g_temp[idx1 - j][m] * h_smooth[j];
+                        q_filt[m] += q_temp[idx1 - j][m] * h_smooth[j];
+                    }
+                }
+            } else {
+                g_filt = g_temp[i + h_SL];
+                q_filt = q_temp[i];
+            }
+
+            sbr->dsp.hf_g_filt(Y1[i] + kx, X_high + kx, g_filt, m_max,
+                               i + ENVELOPE_ADJUSTMENT_OFFSET);
+
+            if (e != e_a[0] && e != e_a[1]) {
+                sbr->dsp.hf_apply_noise[indexsine](Y1[i] + kx, sbr->s_m[e],
+                                                   q_filt, indexnoise,
+                                                   kx, m_max);
+            } else {
+                int idx = indexsine&1;
+                int A = (1-((indexsine+(kx & 1))&2));
+                int B = (A^(-idx)) + idx;
+                float *out = &Y1[i][kx][idx];
+                float *in  = sbr->s_m[e];
+                for (m = 0; m+1 < m_max; m+=2) {
+                    out[2*m  ] += in[m  ] * A;
+                    out[2*m+2] += in[m+1] * B;
+                }
+                if(m_max&1)
+                    out[2*m  ] += in[m  ] * A;
+            }
+            indexnoise = (indexnoise + m_max) & 0x1ff;
+            indexsine = (indexsine + 1) & 3;
+        }
+    }
+    ch_data->f_indexnoise = indexnoise;
+    ch_data->f_indexsine  = indexsine;
+}
+
+#include "aacsbr_template.c"

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacsbr.h

@@ -0,0 +1,96 @@
+/*
+ * AAC Spectral Band Replication function declarations
+ * Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
+ * Copyright (c) 2010      Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC Spectral Band Replication function declarations
+ * @author Robert Swain ( rob opendot cl )
+ */
+
+#ifndef AVCODEC_AACSBR_H
+#define AVCODEC_AACSBR_H
+
+#include "get_bits.h"
+#include "aac.h"
+#include "sbr.h"
+
+#define ENVELOPE_ADJUSTMENT_OFFSET 2
+#define NOISE_FLOOR_OFFSET 6
+
+/**
+ * SBR VLC tables
+ */
+enum {
+    T_HUFFMAN_ENV_1_5DB,
+    F_HUFFMAN_ENV_1_5DB,
+    T_HUFFMAN_ENV_BAL_1_5DB,
+    F_HUFFMAN_ENV_BAL_1_5DB,
+    T_HUFFMAN_ENV_3_0DB,
+    F_HUFFMAN_ENV_3_0DB,
+    T_HUFFMAN_ENV_BAL_3_0DB,
+    F_HUFFMAN_ENV_BAL_3_0DB,
+    T_HUFFMAN_NOISE_3_0DB,
+    T_HUFFMAN_NOISE_BAL_3_0DB,
+};
+
+/**
+ * bs_frame_class - frame class of current SBR frame (14496-3 sp04 p98)
+ */
+enum {
+    FIXFIX,
+    FIXVAR,
+    VARFIX,
+    VARVAR,
+};
+
+enum {
+    EXTENSION_ID_PS = 2,
+};
+
+static const int8_t vlc_sbr_lav[10] =
+    { 60, 60, 24, 24, 31, 31, 12, 12, 31, 12 };
+
+#define SBR_INIT_VLC_STATIC(num, size) \
+    INIT_VLC_STATIC(&vlc_sbr[num], 9, sbr_tmp[num].table_size / sbr_tmp[num].elem_size,     \
+                    sbr_tmp[num].sbr_bits ,                      1,                      1, \
+                    sbr_tmp[num].sbr_codes, sbr_tmp[num].elem_size, sbr_tmp[num].elem_size, \
+                    size)
+
+#define SBR_VLC_ROW(name) \
+    { name ## _codes, name ## _bits, sizeof(name ## _codes), sizeof(name ## _codes[0]) }
+
+/** Initialize SBR. */
+void AAC_RENAME(ff_aac_sbr_init)(void);
+/** Initialize one SBR context. */
+void AAC_RENAME(ff_aac_sbr_ctx_init)(AACContext *ac, SpectralBandReplication *sbr, int id_aac);
+/** Close one SBR context. */
+void AAC_RENAME(ff_aac_sbr_ctx_close)(SpectralBandReplication *sbr);
+/** Decode one SBR element. */
+int AAC_RENAME(ff_decode_sbr_extension)(AACContext *ac, SpectralBandReplication *sbr,
+                            GetBitContext *gb, int crc, int cnt, int id_aac);
+/** Apply one SBR element to one AAC element. */
+void AAC_RENAME(ff_sbr_apply)(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
+                  INTFLOAT* L, INTFLOAT *R);
+
+void ff_aacsbr_func_ptr_init_mips(AACSBRContext *c);
+
+#endif /* AVCODEC_AACSBR_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacsbr_fixed.c

@@ -0,0 +1,613 @@
+/*
+ * Copyright (c) 2013
+ *      MIPS Technologies, Inc., California.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
+ *    contributors may be used to endorse or promote products derived from
+ *    this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED.  IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
+ * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ * AAC Spectral Band Replication decoding functions (fixed-point)
+ * Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
+ * Copyright (c) 2009-2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC Spectral Band Replication decoding functions (fixed-point)
+ * Note: Rounding-to-nearest used unless otherwise stated
+ * @author Robert Swain ( rob opendot cl )
+ * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
+ */
+#define USE_FIXED 1
+
+#include "aac.h"
+#include "sbr.h"
+#include "aacsbr.h"
+#include "aacsbrdata.h"
+#include "aacsbr_fixed_tablegen.h"
+#include "fft.h"
+#include "aacps.h"
+#include "sbrdsp.h"
+#include "libavutil/internal.h"
+#include "libavutil/libm.h"
+#include "libavutil/avassert.h"
+
+#include <stdint.h>
+#include <float.h>
+#include <math.h>
+
+static VLC vlc_sbr[10];
+static void aacsbr_func_ptr_init(AACSBRContext *c);
+static const int CONST_LN2       = Q31(0.6931471806/256);  // ln(2)/256
+static const int CONST_RECIP_LN2 = Q31(0.7213475204);      // 0.5/ln(2)
+static const int CONST_076923    = Q31(0.76923076923076923077f);
+
+static const int fixed_log_table[10] =
+{
+    Q31(1.0/2), Q31(1.0/3), Q31(1.0/4), Q31(1.0/5), Q31(1.0/6),
+    Q31(1.0/7), Q31(1.0/8), Q31(1.0/9), Q31(1.0/10), Q31(1.0/11)
+};
+
+static int fixed_log(int x)
+{
+    int i, ret, xpow, tmp;
+
+    ret = x;
+    xpow = x;
+    for (i=0; i<10; i+=2){
+        xpow = (int)(((int64_t)xpow * x + 0x40000000) >> 31);
+        tmp = (int)(((int64_t)xpow * fixed_log_table[i] + 0x40000000) >> 31);
+        ret -= tmp;
+
+        xpow = (int)(((int64_t)xpow * x + 0x40000000) >> 31);
+        tmp = (int)(((int64_t)xpow * fixed_log_table[i+1] + 0x40000000) >> 31);
+        ret += tmp;
+    }
+
+    return ret;
+}
+
+static const int fixed_exp_table[7] =
+{
+    Q31(1.0/2), Q31(1.0/6), Q31(1.0/24), Q31(1.0/120),
+    Q31(1.0/720), Q31(1.0/5040), Q31(1.0/40320)
+};
+
+static int fixed_exp(int x)
+{
+    int i, ret, xpow, tmp;
+
+    ret = 0x800000 + x;
+    xpow = x;
+    for (i=0; i<7; i++){
+        xpow = (int)(((int64_t)xpow * x + 0x400000) >> 23);
+        tmp = (int)(((int64_t)xpow * fixed_exp_table[i] + 0x40000000) >> 31);
+        ret += tmp;
+    }
+
+    return ret;
+}
+
+static void make_bands(int16_t* bands, int start, int stop, int num_bands)
+{
+    int k, previous, present;
+    int base, prod, nz = 0;
+
+    base = (stop << 23) / start;
+    while (base < 0x40000000){
+        base <<= 1;
+        nz++;
+    }
+    base = fixed_log(base - 0x80000000);
+    base = (((base + 0x80) >> 8) + (8-nz)*CONST_LN2) / num_bands;
+    base = fixed_exp(base);
+
+    previous = start;
+    prod = start << 23;
+
+    for (k = 0; k < num_bands-1; k++) {
+        prod = (int)(((int64_t)prod * base + 0x400000) >> 23);
+        present = (prod + 0x400000) >> 23;
+        bands[k] = present - previous;
+        previous = present;
+    }
+    bands[num_bands-1] = stop - previous;
+}
+
+/// Dequantization and stereo decoding (14496-3 sp04 p203)
+static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
+{
+    int k, e;
+    int ch;
+
+    if (id_aac == TYPE_CPE && sbr->bs_coupling) {
+        int alpha      = sbr->data[0].bs_amp_res ?  2 :  1;
+        int pan_offset = sbr->data[0].bs_amp_res ? 12 : 24;
+        for (e = 1; e <= sbr->data[0].bs_num_env; e++) {
+            for (k = 0; k < sbr->n[sbr->data[0].bs_freq_res[e]]; k++) {
+                SoftFloat temp1, temp2, fac;
+
+                temp1.exp = sbr->data[0].env_facs_q[e][k] * alpha + 14;
+                if (temp1.exp & 1)
+                  temp1.mant = 759250125;
+                else
+                  temp1.mant = 0x20000000;
+                temp1.exp = (temp1.exp >> 1) + 1;
+                if (temp1.exp > 66) { // temp1 > 1E20
+                    av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
+                    temp1 = FLOAT_1;
+                }
+
+                temp2.exp = (pan_offset - sbr->data[1].env_facs_q[e][k]) * alpha;
+                if (temp2.exp & 1)
+                  temp2.mant = 759250125;
+                else
+                  temp2.mant = 0x20000000;
+                temp2.exp = (temp2.exp >> 1) + 1;
+                fac   = av_div_sf(temp1, av_add_sf(FLOAT_1, temp2));
+                sbr->data[0].env_facs[e][k] = fac;
+                sbr->data[1].env_facs[e][k] = av_mul_sf(fac, temp2);
+            }
+        }
+        for (e = 1; e <= sbr->data[0].bs_num_noise; e++) {
+            for (k = 0; k < sbr->n_q; k++) {
+                SoftFloat temp1, temp2, fac;
+
+                temp1.exp = NOISE_FLOOR_OFFSET - \
+                    sbr->data[0].noise_facs_q[e][k] + 2;
+                temp1.mant = 0x20000000;
+                av_assert0(temp1.exp <= 66);
+                temp2.exp = 12 - sbr->data[1].noise_facs_q[e][k] + 1;
+                temp2.mant = 0x20000000;
+                fac   = av_div_sf(temp1, av_add_sf(FLOAT_1, temp2));
+                sbr->data[0].noise_facs[e][k] = fac;
+                sbr->data[1].noise_facs[e][k] = av_mul_sf(fac, temp2);
+            }
+        }
+    } else { // SCE or one non-coupled CPE
+        for (ch = 0; ch < (id_aac == TYPE_CPE) + 1; ch++) {
+            int alpha = sbr->data[ch].bs_amp_res ? 2 : 1;
+            for (e = 1; e <= sbr->data[ch].bs_num_env; e++)
+                for (k = 0; k < sbr->n[sbr->data[ch].bs_freq_res[e]]; k++){
+                    SoftFloat temp1;
+
+                    temp1.exp = alpha * sbr->data[ch].env_facs_q[e][k] + 12;
+                    if (temp1.exp & 1)
+                        temp1.mant = 759250125;
+                    else
+                        temp1.mant = 0x20000000;
+                    temp1.exp = (temp1.exp >> 1) + 1;
+                    if (temp1.exp > 66) { // temp1 > 1E20
+                        av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
+                        temp1 = FLOAT_1;
+                    }
+                    sbr->data[ch].env_facs[e][k] = temp1;
+                }
+            for (e = 1; e <= sbr->data[ch].bs_num_noise; e++)
+                for (k = 0; k < sbr->n_q; k++){
+                    sbr->data[ch].noise_facs[e][k].exp = NOISE_FLOOR_OFFSET - \
+                        sbr->data[ch].noise_facs_q[e][k] + 1;
+                    sbr->data[ch].noise_facs[e][k].mant = 0x20000000;
+                }
+        }
+    }
+}
+
+/** High Frequency Generation (14496-3 sp04 p214+) and Inverse Filtering
+ * (14496-3 sp04 p214)
+ * Warning: This routine does not seem numerically stable.
+ */
+static void sbr_hf_inverse_filter(SBRDSPContext *dsp,
+                                  int (*alpha0)[2], int (*alpha1)[2],
+                                  const int X_low[32][40][2], int k0)
+{
+    int k;
+    int shift, round;
+
+    for (k = 0; k < k0; k++) {
+        SoftFloat phi[3][2][2];
+        SoftFloat a00, a01, a10, a11;
+        SoftFloat dk;
+
+        dsp->autocorrelate(X_low[k], phi);
+
+        dk = av_sub_sf(av_mul_sf(phi[2][1][0], phi[1][0][0]),
+             av_mul_sf(av_add_sf(av_mul_sf(phi[1][1][0], phi[1][1][0]),
+             av_mul_sf(phi[1][1][1], phi[1][1][1])), FLOAT_0999999));
+
+        if (!dk.mant) {
+            a10 = FLOAT_0;
+            a11 = FLOAT_0;
+        } else {
+            SoftFloat temp_real, temp_im;
+            temp_real = av_sub_sf(av_sub_sf(av_mul_sf(phi[0][0][0], phi[1][1][0]),
+                                            av_mul_sf(phi[0][0][1], phi[1][1][1])),
+                                  av_mul_sf(phi[0][1][0], phi[1][0][0]));
+            temp_im   = av_sub_sf(av_add_sf(av_mul_sf(phi[0][0][0], phi[1][1][1]),
+                                            av_mul_sf(phi[0][0][1], phi[1][1][0])),
+                                  av_mul_sf(phi[0][1][1], phi[1][0][0]));
+
+            a10 = av_div_sf(temp_real, dk);
+            a11 = av_div_sf(temp_im,   dk);
+        }
+
+        if (!phi[1][0][0].mant) {
+            a00 = FLOAT_0;
+            a01 = FLOAT_0;
+        } else {
+            SoftFloat temp_real, temp_im;
+            temp_real = av_add_sf(phi[0][0][0],
+                                  av_add_sf(av_mul_sf(a10, phi[1][1][0]),
+                                            av_mul_sf(a11, phi[1][1][1])));
+            temp_im   = av_add_sf(phi[0][0][1],
+                                  av_sub_sf(av_mul_sf(a11, phi[1][1][0]),
+                                            av_mul_sf(a10, phi[1][1][1])));
+
+            temp_real.mant = -temp_real.mant;
+            temp_im.mant   = -temp_im.mant;
+            a00 = av_div_sf(temp_real, phi[1][0][0]);
+            a01 = av_div_sf(temp_im,   phi[1][0][0]);
+        }
+
+        shift = a00.exp;
+        if (shift >= 3)
+            alpha0[k][0] = 0x7fffffff;
+        else if (shift <= -30)
+            alpha0[k][0] = 0;
+        else {
+            shift = 1-shift;
+            if (shift <= 0)
+                alpha0[k][0] = a00.mant * (1<<-shift);
+            else {
+                round = 1 << (shift-1);
+                alpha0[k][0] = (a00.mant + round) >> shift;
+            }
+        }
+
+        shift = a01.exp;
+        if (shift >= 3)
+            alpha0[k][1] = 0x7fffffff;
+        else if (shift <= -30)
+            alpha0[k][1] = 0;
+        else {
+            shift = 1-shift;
+            if (shift <= 0)
+                alpha0[k][1] = a01.mant * (1<<-shift);
+            else {
+                round = 1 << (shift-1);
+                alpha0[k][1] = (a01.mant + round) >> shift;
+            }
+        }
+        shift = a10.exp;
+        if (shift >= 3)
+            alpha1[k][0] = 0x7fffffff;
+        else if (shift <= -30)
+            alpha1[k][0] = 0;
+        else {
+            shift = 1-shift;
+            if (shift <= 0)
+                alpha1[k][0] = a10.mant * (1<<-shift);
+            else {
+                round = 1 << (shift-1);
+                alpha1[k][0] = (a10.mant + round) >> shift;
+            }
+        }
+
+        shift = a11.exp;
+        if (shift >= 3)
+            alpha1[k][1] = 0x7fffffff;
+        else if (shift <= -30)
+            alpha1[k][1] = 0;
+        else {
+            shift = 1-shift;
+            if (shift <= 0)
+                alpha1[k][1] = a11.mant * (1<<-shift);
+            else {
+                round = 1 << (shift-1);
+                alpha1[k][1] = (a11.mant + round) >> shift;
+            }
+        }
+
+        shift = (int)(((int64_t)(alpha1[k][0]>>1) * (alpha1[k][0]>>1) + \
+                       (int64_t)(alpha1[k][1]>>1) * (alpha1[k][1]>>1) + \
+                       0x40000000) >> 31);
+        if (shift >= 0x20000000){
+            alpha1[k][0] = 0;
+            alpha1[k][1] = 0;
+            alpha0[k][0] = 0;
+            alpha0[k][1] = 0;
+        }
+
+        shift = (int)(((int64_t)(alpha0[k][0]>>1) * (alpha0[k][0]>>1) + \
+                       (int64_t)(alpha0[k][1]>>1) * (alpha0[k][1]>>1) + \
+                       0x40000000) >> 31);
+        if (shift >= 0x20000000){
+            alpha1[k][0] = 0;
+            alpha1[k][1] = 0;
+            alpha0[k][0] = 0;
+            alpha0[k][1] = 0;
+        }
+    }
+}
+
+/// Chirp Factors (14496-3 sp04 p214)
+static void sbr_chirp(SpectralBandReplication *sbr, SBRData *ch_data)
+{
+    int i;
+    int new_bw;
+    static const int bw_tab[] = { 0, 1610612736, 1932735283, 2104533975 };
+    int64_t accu;
+
+    for (i = 0; i < sbr->n_q; i++) {
+        if (ch_data->bs_invf_mode[0][i] + ch_data->bs_invf_mode[1][i] == 1)
+            new_bw = 1288490189;
+        else
+            new_bw = bw_tab[ch_data->bs_invf_mode[0][i]];
+
+        if (new_bw < ch_data->bw_array[i]){
+            accu  = (int64_t)new_bw * 1610612736;
+            accu += (int64_t)ch_data->bw_array[i] * 0x20000000;
+            new_bw = (int)((accu + 0x40000000) >> 31);
+        } else {
+            accu  = (int64_t)new_bw * 1946157056;
+            accu += (int64_t)ch_data->bw_array[i] * 201326592;
+            new_bw = (int)((accu + 0x40000000) >> 31);
+        }
+        ch_data->bw_array[i] = new_bw < 0x2000000 ? 0 : new_bw;
+    }
+}
+
+/**
+ * Calculation of levels of additional HF signal components (14496-3 sp04 p219)
+ * and Calculation of gain (14496-3 sp04 p219)
+ */
+static void sbr_gain_calc(AACContext *ac, SpectralBandReplication *sbr,
+                          SBRData *ch_data, const int e_a[2])
+{
+    int e, k, m;
+    // max gain limits : -3dB, 0dB, 3dB, inf dB (limiter off)
+    static const SoftFloat limgain[4] = { { 760155524,  0 }, { 0x20000000,  1 },
+                                            { 758351638,  1 }, { 625000000, 34 } };
+
+    for (e = 0; e < ch_data->bs_num_env; e++) {
+        int delta = !((e == e_a[1]) || (e == e_a[0]));
+        for (k = 0; k < sbr->n_lim; k++) {
+            SoftFloat gain_boost, gain_max;
+            SoftFloat sum[2];
+            sum[0] = sum[1] = FLOAT_0;
+            for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+                const SoftFloat temp = av_div_sf(sbr->e_origmapped[e][m],
+                                            av_add_sf(FLOAT_1, sbr->q_mapped[e][m]));
+                sbr->q_m[e][m] = av_sqrt_sf(av_mul_sf(temp, sbr->q_mapped[e][m]));
+                sbr->s_m[e][m] = av_sqrt_sf(av_mul_sf(temp, av_int2sf(ch_data->s_indexmapped[e + 1][m], 0)));
+                if (!sbr->s_mapped[e][m]) {
+                    if (delta) {
+                      sbr->gain[e][m] = av_sqrt_sf(av_div_sf(sbr->e_origmapped[e][m],
+                                            av_mul_sf(av_add_sf(FLOAT_1, sbr->e_curr[e][m]),
+                                            av_add_sf(FLOAT_1, sbr->q_mapped[e][m]))));
+                    } else {
+                      sbr->gain[e][m] = av_sqrt_sf(av_div_sf(sbr->e_origmapped[e][m],
+                                            av_add_sf(FLOAT_1, sbr->e_curr[e][m])));
+                    }
+                } else {
+                    sbr->gain[e][m] = av_sqrt_sf(
+                                        av_div_sf(
+                                            av_mul_sf(sbr->e_origmapped[e][m], sbr->q_mapped[e][m]),
+                                            av_mul_sf(
+                                                av_add_sf(FLOAT_1, sbr->e_curr[e][m]),
+                                                av_add_sf(FLOAT_1, sbr->q_mapped[e][m]))));
+                }
+                sbr->gain[e][m] = av_add_sf(sbr->gain[e][m], FLOAT_MIN);
+            }
+            for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+                sum[0] = av_add_sf(sum[0], sbr->e_origmapped[e][m]);
+                sum[1] = av_add_sf(sum[1], sbr->e_curr[e][m]);
+            }
+            gain_max = av_mul_sf(limgain[sbr->bs_limiter_gains],
+                            av_sqrt_sf(
+                                av_div_sf(
+                                    av_add_sf(FLOAT_EPSILON, sum[0]),
+                                    av_add_sf(FLOAT_EPSILON, sum[1]))));
+            if (av_gt_sf(gain_max, FLOAT_100000))
+              gain_max = FLOAT_100000;
+            for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+                SoftFloat q_m_max = av_div_sf(
+                                        av_mul_sf(sbr->q_m[e][m], gain_max),
+                                        sbr->gain[e][m]);
+                if (av_gt_sf(sbr->q_m[e][m], q_m_max))
+                  sbr->q_m[e][m] = q_m_max;
+                if (av_gt_sf(sbr->gain[e][m], gain_max))
+                  sbr->gain[e][m] = gain_max;
+            }
+            sum[0] = sum[1] = FLOAT_0;
+            for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+                sum[0] = av_add_sf(sum[0], sbr->e_origmapped[e][m]);
+                sum[1] = av_add_sf(sum[1],
+                            av_mul_sf(
+                                av_mul_sf(sbr->e_curr[e][m],
+                                          sbr->gain[e][m]),
+                                sbr->gain[e][m]));
+                sum[1] = av_add_sf(sum[1],
+                            av_mul_sf(sbr->s_m[e][m], sbr->s_m[e][m]));
+                if (delta && !sbr->s_m[e][m].mant)
+                  sum[1] = av_add_sf(sum[1],
+                                av_mul_sf(sbr->q_m[e][m], sbr->q_m[e][m]));
+            }
+            gain_boost = av_sqrt_sf(
+                            av_div_sf(
+                                av_add_sf(FLOAT_EPSILON, sum[0]),
+                                av_add_sf(FLOAT_EPSILON, sum[1])));
+            if (av_gt_sf(gain_boost, FLOAT_1584893192))
+              gain_boost = FLOAT_1584893192;
+
+            for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+                sbr->gain[e][m] = av_mul_sf(sbr->gain[e][m], gain_boost);
+                sbr->q_m[e][m]  = av_mul_sf(sbr->q_m[e][m], gain_boost);
+                sbr->s_m[e][m]  = av_mul_sf(sbr->s_m[e][m], gain_boost);
+            }
+        }
+    }
+}
+
+/// Assembling HF Signals (14496-3 sp04 p220)
+static void sbr_hf_assemble(int Y1[38][64][2],
+                            const int X_high[64][40][2],
+                            SpectralBandReplication *sbr, SBRData *ch_data,
+                            const int e_a[2])
+{
+    int e, i, j, m;
+    const int h_SL = 4 * !sbr->bs_smoothing_mode;
+    const int kx = sbr->kx[1];
+    const int m_max = sbr->m[1];
+    static const SoftFloat h_smooth[5] = {
+      { 715827883, -1 },
+      { 647472402, -1 },
+      { 937030863, -2 },
+      { 989249804, -3 },
+      { 546843842, -4 },
+    };
+    SoftFloat (*g_temp)[48] = ch_data->g_temp, (*q_temp)[48] = ch_data->q_temp;
+    int indexnoise = ch_data->f_indexnoise;
+    int indexsine  = ch_data->f_indexsine;
+
+    if (sbr->reset) {
+        for (i = 0; i < h_SL; i++) {
+            memcpy(g_temp[i + 2*ch_data->t_env[0]], sbr->gain[0], m_max * sizeof(sbr->gain[0][0]));
+            memcpy(q_temp[i + 2*ch_data->t_env[0]], sbr->q_m[0],  m_max * sizeof(sbr->q_m[0][0]));
+        }
+    } else if (h_SL) {
+        for (i = 0; i < 4; i++) {
+            memcpy(g_temp[i + 2 * ch_data->t_env[0]],
+                   g_temp[i + 2 * ch_data->t_env_num_env_old],
+                   sizeof(g_temp[0]));
+            memcpy(q_temp[i + 2 * ch_data->t_env[0]],
+                   q_temp[i + 2 * ch_data->t_env_num_env_old],
+                   sizeof(q_temp[0]));
+        }
+    }
+
+    for (e = 0; e < ch_data->bs_num_env; e++) {
+        for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
+            memcpy(g_temp[h_SL + i], sbr->gain[e], m_max * sizeof(sbr->gain[0][0]));
+            memcpy(q_temp[h_SL + i], sbr->q_m[e],  m_max * sizeof(sbr->q_m[0][0]));
+        }
+    }
+
+    for (e = 0; e < ch_data->bs_num_env; e++) {
+        for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
+            SoftFloat g_filt_tab[48];
+            SoftFloat q_filt_tab[48];
+            SoftFloat *g_filt, *q_filt;
+
+            if (h_SL && e != e_a[0] && e != e_a[1]) {
+                g_filt = g_filt_tab;
+                q_filt = q_filt_tab;
+                for (m = 0; m < m_max; m++) {
+                    const int idx1 = i + h_SL;
+                    g_filt[m].mant = g_filt[m].exp = 0;
+                    q_filt[m].mant = q_filt[m].exp = 0;
+                    for (j = 0; j <= h_SL; j++) {
+                        g_filt[m] = av_add_sf(g_filt[m],
+                                        av_mul_sf(g_temp[idx1 - j][m],
+                                            h_smooth[j]));
+                        q_filt[m] = av_add_sf(q_filt[m],
+                                        av_mul_sf(q_temp[idx1 - j][m],
+                                            h_smooth[j]));
+                    }
+                }
+            } else {
+                g_filt = g_temp[i + h_SL];
+                q_filt = q_temp[i];
+            }
+
+            sbr->dsp.hf_g_filt(Y1[i] + kx, X_high + kx, g_filt, m_max,
+                               i + ENVELOPE_ADJUSTMENT_OFFSET);
+
+            if (e != e_a[0] && e != e_a[1]) {
+                sbr->dsp.hf_apply_noise[indexsine](Y1[i] + kx, sbr->s_m[e],
+                                                   q_filt, indexnoise,
+                                                   kx, m_max);
+            } else {
+                int idx = indexsine&1;
+                int A = (1-((indexsine+(kx & 1))&2));
+                int B = (A^(-idx)) + idx;
+                unsigned *out = &Y1[i][kx][idx];
+                int shift;
+                unsigned round;
+
+                SoftFloat *in  = sbr->s_m[e];
+                for (m = 0; m+1 < m_max; m+=2) {
+                    int shift2;
+                    shift = 22 - in[m  ].exp;
+                    shift2= 22 - in[m+1].exp;
+                    if (shift < 1 || shift2 < 1) {
+                        av_log(NULL, AV_LOG_ERROR, "Overflow in sbr_hf_assemble, shift=%d,%d\n", shift, shift2);
+                        return;
+                    }
+                    if (shift < 32) {
+                        round = 1 << (shift-1);
+                        out[2*m  ] += (int)(in[m  ].mant * A + round) >> shift;
+                    }
+
+                    if (shift2 < 32) {
+                        round = 1 << (shift2-1);
+                        out[2*m+2] += (int)(in[m+1].mant * B + round) >> shift2;
+                    }
+                }
+                if(m_max&1)
+                {
+                    shift = 22 - in[m  ].exp;
+                    if (shift < 1) {
+                        av_log(NULL, AV_LOG_ERROR, "Overflow in sbr_hf_assemble, shift=%d\n", shift);
+                        return;
+                    } else if (shift < 32) {
+                        round = 1 << (shift-1);
+                        out[2*m  ] += (int)(in[m  ].mant * A + round) >> shift;
+                    }
+                }
+            }
+            indexnoise = (indexnoise + m_max) & 0x1ff;
+            indexsine = (indexsine + 1) & 3;
+        }
+    }
+    ch_data->f_indexnoise = indexnoise;
+    ch_data->f_indexsine  = indexsine;
+}
+
+#include "aacsbr_template.c"

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacsbr_fixed_tablegen.h

@@ -0,0 +1,28 @@
+/*
+ * Header file for hardcoded AAC SBR windows
+ *
+ * Copyright (c) 2014 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AACSBR_FIXED_TABLEGEN_H
+#define AVCODEC_AACSBR_FIXED_TABLEGEN_H
+
+#include "aacsbr_tablegen_common.h"
+
+#endif /* AVCODEC_AACSBR_FIXED_TABLEGEN_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacsbr_tablegen.h

@@ -0,0 +1,28 @@
+/*
+ * Header file for hardcoded AAC SBR windows
+ *
+ * Copyright (c) 2014 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AACSBR_TABLEGEN_H
+#define AVCODEC_AACSBR_TABLEGEN_H
+
+#include "aacsbr_tablegen_common.h"
+
+#endif /* AVCODEC_AACSBR_TABLEGEN_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacsbr_tablegen_common.h

@@ -0,0 +1,114 @@
+/*
+ * Header file for hardcoded AAC SBR windows
+ *
+ * Copyright (c) 2014 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AACSBR_TABLEGEN_COMMON_H
+#define AVCODEC_AACSBR_TABLEGEN_COMMON_H
+#include "aac_defines.h"
+#include "libavutil/mem.h"
+
+///< window coefficients for analysis/synthesis QMF banks
+static DECLARE_ALIGNED(32, INTFLOAT, sbr_qmf_window_ds)[320];
+static DECLARE_ALIGNED(32, INTFLOAT, sbr_qmf_window_us)[640] = {
+    Q31( 0.0000000000f), Q31(-0.0005525286f), Q31(-0.0005617692f), Q31(-0.0004947518f),
+    Q31(-0.0004875227f), Q31(-0.0004893791f), Q31(-0.0005040714f), Q31(-0.0005226564f),
+    Q31(-0.0005466565f), Q31(-0.0005677802f), Q31(-0.0005870930f), Q31(-0.0006132747f),
+    Q31(-0.0006312493f), Q31(-0.0006540333f), Q31(-0.0006777690f), Q31(-0.0006941614f),
+    Q31(-0.0007157736f), Q31(-0.0007255043f), Q31(-0.0007440941f), Q31(-0.0007490598f),
+    Q31(-0.0007681371f), Q31(-0.0007724848f), Q31(-0.0007834332f), Q31(-0.0007779869f),
+    Q31(-0.0007803664f), Q31(-0.0007801449f), Q31(-0.0007757977f), Q31(-0.0007630793f),
+    Q31(-0.0007530001f), Q31(-0.0007319357f), Q31(-0.0007215391f), Q31(-0.0006917937f),
+    Q31(-0.0006650415f), Q31(-0.0006341594f), Q31(-0.0005946118f), Q31(-0.0005564576f),
+    Q31(-0.0005145572f), Q31(-0.0004606325f), Q31(-0.0004095121f), Q31(-0.0003501175f),
+    Q31(-0.0002896981f), Q31(-0.0002098337f), Q31(-0.0001446380f), Q31(-0.0000617334f),
+    Q31( 0.0000134949f), Q31( 0.0001094383f), Q31( 0.0002043017f), Q31( 0.0002949531f),
+    Q31( 0.0004026540f), Q31( 0.0005107388f), Q31( 0.0006239376f), Q31( 0.0007458025f),
+    Q31( 0.0008608443f), Q31( 0.0009885988f), Q31( 0.0011250155f), Q31( 0.0012577884f),
+    Q31( 0.0013902494f), Q31( 0.0015443219f), Q31( 0.0016868083f), Q31( 0.0018348265f),
+    Q31( 0.0019841140f), Q31( 0.0021461583f), Q31( 0.0023017254f), Q31( 0.0024625616f),
+    Q31( 0.0026201758f), Q31( 0.0027870464f), Q31( 0.0029469447f), Q31( 0.0031125420f),
+    Q31( 0.0032739613f), Q31( 0.0034418874f), Q31( 0.0036008268f), Q31( 0.0037603922f),
+    Q31( 0.0039207432f), Q31( 0.0040819753f), Q31( 0.0042264269f), Q31( 0.0043730719f),
+    Q31( 0.0045209852f), Q31( 0.0046606460f), Q31( 0.0047932560f), Q31( 0.0049137603f),
+    Q31( 0.0050393022f), Q31( 0.0051407353f), Q31( 0.0052461166f), Q31( 0.0053471681f),
+    Q31( 0.0054196775f), Q31( 0.0054876040f), Q31( 0.0055475714f), Q31( 0.0055938023f),
+    Q31( 0.0056220643f), Q31( 0.0056455196f), Q31( 0.0056389199f), Q31( 0.0056266114f),
+    Q31( 0.0055917128f), Q31( 0.0055404363f), Q31( 0.0054753783f), Q31( 0.0053838975f),
+    Q31( 0.0052715758f), Q31( 0.0051382275f), Q31( 0.0049839687f), Q31( 0.0048109469f),
+    Q31( 0.0046039530f), Q31( 0.0043801861f), Q31( 0.0041251642f), Q31( 0.0038456408f),
+    Q31( 0.0035401246f), Q31( 0.0032091885f), Q31( 0.0028446757f), Q31( 0.0024508540f),
+    Q31( 0.0020274176f), Q31( 0.0015784682f), Q31( 0.0010902329f), Q31( 0.0005832264f),
+    Q31( 0.0000276045f), Q31(-0.0005464280f), Q31(-0.0011568135f), Q31(-0.0018039472f),
+    Q31(-0.0024826723f), Q31(-0.0031933778f), Q31(-0.0039401124f), Q31(-0.0047222596f),
+    Q31(-0.0055337211f), Q31(-0.0063792293f), Q31(-0.0072615816f), Q31(-0.0081798233f),
+    Q31(-0.0091325329f), Q31(-0.0101150215f), Q31(-0.0111315548f), Q31(-0.0121849995f),
+    Q31( 0.0132718220f), Q31( 0.0143904666f), Q31( 0.0155405553f), Q31( 0.0167324712f),
+    Q31( 0.0179433381f), Q31( 0.0191872431f), Q31( 0.0204531793f), Q31( 0.0217467550f),
+    Q31( 0.0230680169f), Q31( 0.0244160992f), Q31( 0.0257875847f), Q31( 0.0271859429f),
+    Q31( 0.0286072173f), Q31( 0.0300502657f), Q31( 0.0315017608f), Q31( 0.0329754081f),
+    Q31( 0.0344620948f), Q31( 0.0359697560f), Q31( 0.0374812850f), Q31( 0.0390053679f),
+    Q31( 0.0405349170f), Q31( 0.0420649094f), Q31( 0.0436097542f), Q31( 0.0451488405f),
+    Q31( 0.0466843027f), Q31( 0.0482165720f), Q31( 0.0497385755f), Q31( 0.0512556155f),
+    Q31( 0.0527630746f), Q31( 0.0542452768f), Q31( 0.0557173648f), Q31( 0.0571616450f),
+    Q31( 0.0585915683f), Q31( 0.0599837480f), Q31( 0.0613455171f), Q31( 0.0626857808f),
+    Q31( 0.0639715898f), Q31( 0.0652247106f), Q31( 0.0664367512f), Q31( 0.0676075985f),
+    Q31( 0.0687043828f), Q31( 0.0697630244f), Q31( 0.0707628710f), Q31( 0.0717002673f),
+    Q31( 0.0725682583f), Q31( 0.0733620255f), Q31( 0.0741003642f), Q31( 0.0747452558f),
+    Q31( 0.0753137336f), Q31( 0.0758008358f), Q31( 0.0761992479f), Q31( 0.0764992170f),
+    Q31( 0.0767093490f), Q31( 0.0768173975f), Q31( 0.0768230011f), Q31( 0.0767204924f),
+    Q31( 0.0765050718f), Q31( 0.0761748321f), Q31( 0.0757305756f), Q31( 0.0751576255f),
+    Q31( 0.0744664394f), Q31( 0.0736406005f), Q31( 0.0726774642f), Q31( 0.0715826364f),
+    Q31( 0.0703533073f), Q31( 0.0689664013f), Q31( 0.0674525021f), Q31( 0.0657690668f),
+    Q31( 0.0639444805f), Q31( 0.0619602779f), Q31( 0.0598166570f), Q31( 0.0575152691f),
+    Q31( 0.0550460034f), Q31( 0.0524093821f), Q31( 0.0495978676f), Q31( 0.0466303305f),
+    Q31( 0.0434768782f), Q31( 0.0401458278f), Q31( 0.0366418116f), Q31( 0.0329583930f),
+    Q31( 0.0290824006f), Q31( 0.0250307561f), Q31( 0.0207997072f), Q31( 0.0163701258f),
+    Q31( 0.0117623832f), Q31( 0.0069636862f), Q31( 0.0019765601f), Q31(-0.0032086896f),
+    Q31(-0.0085711749f), Q31(-0.0141288827f), Q31(-0.0198834129f), Q31(-0.0258227288f),
+    Q31(-0.0319531274f), Q31(-0.0382776572f), Q31(-0.0447806821f), Q31(-0.0514804176f),
+    Q31(-0.0583705326f), Q31(-0.0654409853f), Q31(-0.0726943300f), Q31(-0.0801372934f),
+    Q31(-0.0877547536f), Q31(-0.0955533352f), Q31(-0.1035329531f), Q31(-0.1116826931f),
+    Q31(-0.1200077984f), Q31(-0.1285002850f), Q31(-0.1371551761f), Q31(-0.1459766491f),
+    Q31(-0.1549607071f), Q31(-0.1640958855f), Q31(-0.1733808172f), Q31(-0.1828172548f),
+    Q31(-0.1923966745f), Q31(-0.2021250176f), Q31(-0.2119735853f), Q31(-0.2219652696f),
+    Q31(-0.2320690870f), Q31(-0.2423016884f), Q31(-0.2526480309f), Q31(-0.2631053299f),
+    Q31(-0.2736634040f), Q31(-0.2843214189f), Q31(-0.2950716717f), Q31(-0.3059098575f),
+    Q31(-0.3168278913f), Q31(-0.3278113727f), Q31(-0.3388722693f), Q31(-0.3499914122f),
+    Q31( 0.3611589903f), Q31( 0.3723795546f), Q31( 0.3836350013f), Q31( 0.3949211761f),
+    Q31( 0.4062317676f), Q31( 0.4175696896f), Q31( 0.4289119920f), Q31( 0.4402553754f),
+    Q31( 0.4515996535f), Q31( 0.4629308085f), Q31( 0.4742453214f), Q31( 0.4855253091f),
+    Q31( 0.4967708254f), Q31( 0.5079817500f), Q31( 0.5191234970f), Q31( 0.5302240895f),
+    Q31( 0.5412553448f), Q31( 0.5522051258f), Q31( 0.5630789140f), Q31( 0.5738524131f),
+    Q31( 0.5845403235f), Q31( 0.5951123086f), Q31( 0.6055783538f), Q31( 0.6159109932f),
+    Q31( 0.6261242695f), Q31( 0.6361980107f), Q31( 0.6461269695f), Q31( 0.6559016302f),
+    Q31( 0.6655139880f), Q31( 0.6749663190f), Q31( 0.6842353293f), Q31( 0.6933282376f),
+    Q31( 0.7022388719f), Q31( 0.7109410426f), Q31( 0.7194462634f), Q31( 0.7277448900f),
+    Q31( 0.7358211758f), Q31( 0.7436827863f), Q31( 0.7513137456f), Q31( 0.7587080760f),
+    Q31( 0.7658674865f), Q31( 0.7727780881f), Q31( 0.7794287519f), Q31( 0.7858353120f),
+    Q31( 0.7919735841f), Q31( 0.7978466413f), Q31( 0.8034485751f), Q31( 0.8087695004f),
+    Q31( 0.8138191270f), Q31( 0.8185776004f), Q31( 0.8230419890f), Q31( 0.8272275347f),
+    Q31( 0.8311038457f), Q31( 0.8346937361f), Q31( 0.8379717337f), Q31( 0.8409541392f),
+    Q31( 0.8436238281f), Q31( 0.8459818469f), Q31( 0.8480315777f), Q31( 0.8497805198f),
+    Q31( 0.8511971524f), Q31( 0.8523047035f), Q31( 0.8531020949f), Q31( 0.8535720573f),
+    Q31( 0.8537385600f),
+};
+
+#endif /* AVCODEC_AACSBR_TABLEGEN_COMMON_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aacsbrdata.h

@@ -0,0 +1,535 @@
+/*
+ * AAC Spectral Band Replication decoding data
+ * Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC Spectral Band Replication decoding data
+ * @author Robert Swain ( rob opendot cl )
+ */
+
+#ifndef AVCODEC_AACSBRDATA_H
+#define AVCODEC_AACSBRDATA_H
+
+#include <stdint.h>
+#include "libavutil/mem.h"
+#include "aac_defines.h"
+
+///< Huffman tables for SBR
+
+static const uint8_t t_huffman_env_1_5dB_bits[121] = {
+    18, 18, 18, 18, 18, 18, 19, 19,
+    19, 19, 19, 19, 19, 19, 19, 19,
+    19, 19, 19, 19, 19, 19, 19, 19,
+    19, 19, 19, 19, 19, 19, 19, 19,
+    19, 19, 17, 18, 16, 17, 18, 17,
+    16, 16, 16, 16, 15, 14, 14, 13,
+    13, 12, 11, 10,  9,  8,  7,  6,
+     5,  4,  3,  2,  2,  3,  4,  5,
+     6,  7,  8,  9, 10, 12, 13, 14,
+    14, 15, 16, 17, 16, 19, 19, 19,
+    19, 19, 19, 19, 19, 19, 19, 19,
+    19, 19, 19, 19, 19, 19, 19, 19,
+    19, 19, 19, 19, 19, 19, 19, 19,
+    19, 19, 19, 19, 19, 19, 19, 19,
+    19, 19, 19, 19, 19, 19, 19, 19,
+    19,
+};
+
+static const uint32_t t_huffman_env_1_5dB_codes[121] = {
+    0x3ffd6, 0x3ffd7, 0x3ffd8, 0x3ffd9, 0x3ffda, 0x3ffdb, 0x7ffb8, 0x7ffb9,
+    0x7ffba, 0x7ffbb, 0x7ffbc, 0x7ffbd, 0x7ffbe, 0x7ffbf, 0x7ffc0, 0x7ffc1,
+    0x7ffc2, 0x7ffc3, 0x7ffc4, 0x7ffc5, 0x7ffc6, 0x7ffc7, 0x7ffc8, 0x7ffc9,
+    0x7ffca, 0x7ffcb, 0x7ffcc, 0x7ffcd, 0x7ffce, 0x7ffcf, 0x7ffd0, 0x7ffd1,
+    0x7ffd2, 0x7ffd3, 0x1ffe6, 0x3ffd4, 0x0fff0, 0x1ffe9, 0x3ffd5, 0x1ffe7,
+    0x0fff1, 0x0ffec, 0x0ffed, 0x0ffee, 0x07ff4, 0x03ff9, 0x03ff7, 0x01ffa,
+    0x01ff9, 0x00ffb, 0x007fc, 0x003fc, 0x001fd, 0x000fd, 0x0007d, 0x0003d,
+    0x0001d, 0x0000d, 0x00005, 0x00001, 0x00000, 0x00004, 0x0000c, 0x0001c,
+    0x0003c, 0x0007c, 0x000fc, 0x001fc, 0x003fd, 0x00ffa, 0x01ff8, 0x03ff6,
+    0x03ff8, 0x07ff5, 0x0ffef, 0x1ffe8, 0x0fff2, 0x7ffd4, 0x7ffd5, 0x7ffd6,
+    0x7ffd7, 0x7ffd8, 0x7ffd9, 0x7ffda, 0x7ffdb, 0x7ffdc, 0x7ffdd, 0x7ffde,
+    0x7ffdf, 0x7ffe0, 0x7ffe1, 0x7ffe2, 0x7ffe3, 0x7ffe4, 0x7ffe5, 0x7ffe6,
+    0x7ffe7, 0x7ffe8, 0x7ffe9, 0x7ffea, 0x7ffeb, 0x7ffec, 0x7ffed, 0x7ffee,
+    0x7ffef, 0x7fff0, 0x7fff1, 0x7fff2, 0x7fff3, 0x7fff4, 0x7fff5, 0x7fff6,
+    0x7fff7, 0x7fff8, 0x7fff9, 0x7fffa, 0x7fffb, 0x7fffc, 0x7fffd, 0x7fffe,
+    0x7ffff,
+};
+
+static const uint8_t f_huffman_env_1_5dB_bits[121] = {
+    19, 19, 20, 20, 20, 20, 20, 20,
+    20, 19, 20, 20, 20, 20, 19, 20,
+    19, 19, 20, 18, 20, 20, 20, 19,
+    20, 20, 20, 19, 20, 19, 18, 19,
+    18, 18, 17, 18, 17, 17, 17, 16,
+    16, 16, 15, 15, 14, 13, 13, 12,
+    12, 11, 10,  9,  9,  8,  7,  6,
+     5,  4,  3,  2,  2,  3,  4,  5,
+     6,  8,  8,  9, 10, 11, 11, 11,
+    12, 12, 13, 13, 14, 14, 16, 16,
+    17, 17, 18, 18, 18, 18, 18, 18,
+    18, 20, 19, 20, 20, 20, 20, 20,
+    20, 19, 20, 20, 20, 20, 19, 20,
+    18, 20, 20, 19, 19, 20, 20, 20,
+    20, 20, 20, 20, 20, 20, 20, 20,
+    20,
+};
+
+static const uint32_t f_huffman_env_1_5dB_codes[121] = {
+    0x7ffe7, 0x7ffe8, 0xfffd2, 0xfffd3, 0xfffd4, 0xfffd5, 0xfffd6, 0xfffd7,
+    0xfffd8, 0x7ffda, 0xfffd9, 0xfffda, 0xfffdb, 0xfffdc, 0x7ffdb, 0xfffdd,
+    0x7ffdc, 0x7ffdd, 0xfffde, 0x3ffe4, 0xfffdf, 0xfffe0, 0xfffe1, 0x7ffde,
+    0xfffe2, 0xfffe3, 0xfffe4, 0x7ffdf, 0xfffe5, 0x7ffe0, 0x3ffe8, 0x7ffe1,
+    0x3ffe0, 0x3ffe9, 0x1ffef, 0x3ffe5, 0x1ffec, 0x1ffed, 0x1ffee, 0x0fff4,
+    0x0fff3, 0x0fff0, 0x07ff7, 0x07ff6, 0x03ffa, 0x01ffa, 0x01ff9, 0x00ffa,
+    0x00ff8, 0x007f9, 0x003fb, 0x001fc, 0x001fa, 0x000fb, 0x0007c, 0x0003c,
+    0x0001c, 0x0000c, 0x00005, 0x00001, 0x00000, 0x00004, 0x0000d, 0x0001d,
+    0x0003d, 0x000fa, 0x000fc, 0x001fb, 0x003fa, 0x007f8, 0x007fa, 0x007fb,
+    0x00ff9, 0x00ffb, 0x01ff8, 0x01ffb, 0x03ff8, 0x03ff9, 0x0fff1, 0x0fff2,
+    0x1ffea, 0x1ffeb, 0x3ffe1, 0x3ffe2, 0x3ffea, 0x3ffe3, 0x3ffe6, 0x3ffe7,
+    0x3ffeb, 0xfffe6, 0x7ffe2, 0xfffe7, 0xfffe8, 0xfffe9, 0xfffea, 0xfffeb,
+    0xfffec, 0x7ffe3, 0xfffed, 0xfffee, 0xfffef, 0xffff0, 0x7ffe4, 0xffff1,
+    0x3ffec, 0xffff2, 0xffff3, 0x7ffe5, 0x7ffe6, 0xffff4, 0xffff5, 0xffff6,
+    0xffff7, 0xffff8, 0xffff9, 0xffffa, 0xffffb, 0xffffc, 0xffffd, 0xffffe,
+    0xfffff,
+};
+
+static const uint8_t t_huffman_env_bal_1_5dB_bits[49] = {
+    16, 16, 16, 16, 16, 16, 16, 16,
+    16, 16, 16, 16, 16, 16, 16, 16,
+    16, 16, 12, 11,  9,  7,  5,  3,
+     1,  2,  4,  6,  8, 11, 12, 15,
+    16, 16, 16, 16, 16, 16, 16, 17,
+    17, 17, 17, 17, 17, 17, 17, 17,
+    17,
+};
+
+static const uint32_t t_huffman_env_bal_1_5dB_codes[49] = {
+    0x0ffe4, 0x0ffe5, 0x0ffe6, 0x0ffe7, 0x0ffe8, 0x0ffe9, 0x0ffea, 0x0ffeb,
+    0x0ffec, 0x0ffed, 0x0ffee, 0x0ffef, 0x0fff0, 0x0fff1, 0x0fff2, 0x0fff3,
+    0x0fff4, 0x0ffe2, 0x00ffc, 0x007fc, 0x001fe, 0x0007e, 0x0001e, 0x00006,
+    0x00000, 0x00002, 0x0000e, 0x0003e, 0x000fe, 0x007fd, 0x00ffd, 0x07ff0,
+    0x0ffe3, 0x0fff5, 0x0fff6, 0x0fff7, 0x0fff8, 0x0fff9, 0x0fffa, 0x1fff6,
+    0x1fff7, 0x1fff8, 0x1fff9, 0x1fffa, 0x1fffb, 0x1fffc, 0x1fffd, 0x1fffe,
+    0x1ffff,
+};
+
+static const uint8_t f_huffman_env_bal_1_5dB_bits[49] = {
+    18, 18, 18, 18, 18, 18, 18, 18,
+    18, 18, 18, 18, 18, 18, 18, 16,
+    17, 14, 11, 11,  8,  7,  4,  2,
+     1,  3,  5,  6,  9, 11, 12, 15,
+    16, 18, 18, 18, 18, 18, 18, 18,
+    18, 18, 18, 18, 18, 18, 18, 19,
+    19,
+};
+
+static const uint32_t f_huffman_env_bal_1_5dB_codes[49] = {
+    0x3ffe2, 0x3ffe3, 0x3ffe4, 0x3ffe5, 0x3ffe6, 0x3ffe7, 0x3ffe8, 0x3ffe9,
+    0x3ffea, 0x3ffeb, 0x3ffec, 0x3ffed, 0x3ffee, 0x3ffef, 0x3fff0, 0x0fff7,
+    0x1fff0, 0x03ffc, 0x007fe, 0x007fc, 0x000fe, 0x0007e, 0x0000e, 0x00002,
+    0x00000, 0x00006, 0x0001e, 0x0003e, 0x001fe, 0x007fd, 0x00ffe, 0x07ffa,
+    0x0fff6, 0x3fff1, 0x3fff2, 0x3fff3, 0x3fff4, 0x3fff5, 0x3fff6, 0x3fff7,
+    0x3fff8, 0x3fff9, 0x3fffa, 0x3fffb, 0x3fffc, 0x3fffd, 0x3fffe, 0x7fffe,
+    0x7ffff,
+};
+
+static const uint8_t t_huffman_env_3_0dB_bits[63] = {
+    18, 18, 19, 19, 19, 19, 19, 19,
+    19, 19, 19, 19, 19, 19, 19, 19,
+    19, 17, 16, 16, 16, 14, 14, 14,
+    13, 12, 11,  8,  6,  4,  2,  1,
+     3,  5,  7,  9, 11, 13, 14, 14,
+    15, 16, 17, 18, 19, 19, 19, 19,
+    19, 19, 19, 19, 19, 19, 19, 19,
+    19, 19, 19, 19, 19, 19, 19,
+};
+
+static const uint32_t t_huffman_env_3_0dB_codes[63] = {
+    0x3ffed, 0x3ffee, 0x7ffde, 0x7ffdf, 0x7ffe0, 0x7ffe1, 0x7ffe2, 0x7ffe3,
+    0x7ffe4, 0x7ffe5, 0x7ffe6, 0x7ffe7, 0x7ffe8, 0x7ffe9, 0x7ffea, 0x7ffeb,
+    0x7ffec, 0x1fff4, 0x0fff7, 0x0fff9, 0x0fff8, 0x03ffb, 0x03ffa, 0x03ff8,
+    0x01ffa, 0x00ffc, 0x007fc, 0x000fe, 0x0003e, 0x0000e, 0x00002, 0x00000,
+    0x00006, 0x0001e, 0x0007e, 0x001fe, 0x007fd, 0x01ffb, 0x03ff9, 0x03ffc,
+    0x07ffa, 0x0fff6, 0x1fff5, 0x3ffec, 0x7ffed, 0x7ffee, 0x7ffef, 0x7fff0,
+    0x7fff1, 0x7fff2, 0x7fff3, 0x7fff4, 0x7fff5, 0x7fff6, 0x7fff7, 0x7fff8,
+    0x7fff9, 0x7fffa, 0x7fffb, 0x7fffc, 0x7fffd, 0x7fffe, 0x7ffff,
+};
+
+static const uint8_t f_huffman_env_3_0dB_bits[63] = {
+    20, 20, 20, 20, 20, 20, 20, 18,
+    19, 19, 19, 19, 18, 18, 20, 19,
+    17, 18, 17, 16, 16, 15, 14, 12,
+    11, 10,  9,  8,  6,  4,  2,  1,
+     3,  5,  8,  9, 10, 11, 12, 13,
+    14, 15, 15, 16, 16, 17, 17, 18,
+    18, 18, 20, 19, 19, 19, 20, 19,
+    19, 20, 20, 20, 20, 20, 20,
+};
+
+static const uint32_t f_huffman_env_3_0dB_codes[63] = {
+    0xffff0, 0xffff1, 0xffff2, 0xffff3, 0xffff4, 0xffff5, 0xffff6, 0x3fff3,
+    0x7fff5, 0x7ffee, 0x7ffef, 0x7fff6, 0x3fff4, 0x3fff2, 0xffff7, 0x7fff0,
+    0x1fff5, 0x3fff0, 0x1fff4, 0x0fff7, 0x0fff6, 0x07ff8, 0x03ffb, 0x00ffd,
+    0x007fd, 0x003fd, 0x001fd, 0x000fd, 0x0003e, 0x0000e, 0x00002, 0x00000,
+    0x00006, 0x0001e, 0x000fc, 0x001fc, 0x003fc, 0x007fc, 0x00ffc, 0x01ffc,
+    0x03ffa, 0x07ff9, 0x07ffa, 0x0fff8, 0x0fff9, 0x1fff6, 0x1fff7, 0x3fff5,
+    0x3fff6, 0x3fff1, 0xffff8, 0x7fff1, 0x7fff2, 0x7fff3, 0xffff9, 0x7fff7,
+    0x7fff4, 0xffffa, 0xffffb, 0xffffc, 0xffffd, 0xffffe, 0xfffff,
+};
+
+static const uint8_t t_huffman_env_bal_3_0dB_bits[25] = {
+    13, 13, 13, 13, 13, 13, 13, 12,
+     8,  7,  4,  3,  1,  2,  5,  6,
+     9, 13, 13, 13, 13, 13, 13, 14,
+    14,
+};
+
+static const uint16_t t_huffman_env_bal_3_0dB_codes[25] = {
+    0x1ff2, 0x1ff3, 0x1ff4, 0x1ff5, 0x1ff6, 0x1ff7, 0x1ff8, 0x0ff8,
+    0x00fe, 0x007e, 0x000e, 0x0006, 0x0000, 0x0002, 0x001e, 0x003e,
+    0x01fe, 0x1ff9, 0x1ffa, 0x1ffb, 0x1ffc, 0x1ffd, 0x1ffe, 0x3ffe,
+    0x3fff,
+};
+
+static const uint8_t f_huffman_env_bal_3_0dB_bits[25] = {
+    13, 13, 13, 13, 13, 14, 14, 11,
+     8,  7,  4,  2,  1,  3,  5,  6,
+     9, 12, 13, 14, 14, 14, 14, 14,
+    14,
+};
+
+static const uint16_t f_huffman_env_bal_3_0dB_codes[25] = {
+    0x1ff7, 0x1ff8, 0x1ff9, 0x1ffa, 0x1ffb, 0x3ff8, 0x3ff9, 0x07fc,
+    0x00fe, 0x007e, 0x000e, 0x0002, 0x0000, 0x0006, 0x001e, 0x003e,
+    0x01fe, 0x0ffa, 0x1ff6, 0x3ffa, 0x3ffb, 0x3ffc, 0x3ffd, 0x3ffe,
+    0x3fff,
+};
+
+static const uint8_t t_huffman_noise_3_0dB_bits[63] = {
+    13, 13, 13, 13, 13, 13, 13, 13,
+    13, 13, 13, 13, 13, 13, 13, 13,
+    13, 13, 13, 13, 13, 13, 13, 13,
+    13, 13, 11,  8,  6,  4,  3,  1,
+     2,  5,  8, 10, 13, 13, 13, 13,
+    13, 13, 13, 13, 13, 13, 13, 13,
+    13, 13, 13, 13, 13, 13, 13, 13,
+    13, 13, 13, 13, 13, 14, 14,
+};
+
+static const uint16_t t_huffman_noise_3_0dB_codes[63] = {
+    0x1fce, 0x1fcf, 0x1fd0, 0x1fd1, 0x1fd2, 0x1fd3, 0x1fd4, 0x1fd5,
+    0x1fd6, 0x1fd7, 0x1fd8, 0x1fd9, 0x1fda, 0x1fdb, 0x1fdc, 0x1fdd,
+    0x1fde, 0x1fdf, 0x1fe0, 0x1fe1, 0x1fe2, 0x1fe3, 0x1fe4, 0x1fe5,
+    0x1fe6, 0x1fe7, 0x07f2, 0x00fd, 0x003e, 0x000e, 0x0006, 0x0000,
+    0x0002, 0x001e, 0x00fc, 0x03f8, 0x1fcc, 0x1fe8, 0x1fe9, 0x1fea,
+    0x1feb, 0x1fec, 0x1fcd, 0x1fed, 0x1fee, 0x1fef, 0x1ff0, 0x1ff1,
+    0x1ff2, 0x1ff3, 0x1ff4, 0x1ff5, 0x1ff6, 0x1ff7, 0x1ff8, 0x1ff9,
+    0x1ffa, 0x1ffb, 0x1ffc, 0x1ffd, 0x1ffe, 0x3ffe, 0x3fff,
+};
+
+static const uint8_t t_huffman_noise_bal_3_0dB_bits[25] = {
+    8, 8, 8, 8, 8, 8, 8, 8,
+    8, 8, 5, 2, 1, 3, 6, 8,
+    8, 8, 8, 8, 8, 8, 8, 8,
+    8,
+};
+
+static const uint8_t t_huffman_noise_bal_3_0dB_codes[25] = {
+    0xec, 0xed, 0xee, 0xef, 0xf0, 0xf1, 0xf2, 0xf3,
+    0xf4, 0xf5, 0x1c, 0x02, 0x00, 0x06, 0x3a, 0xf6,
+    0xf7, 0xf8, 0xf9, 0xfa, 0xfb, 0xfc, 0xfd, 0xfe,
+    0xff,
+};
+
+static const int8_t sbr_offset[6][16] = {
+    {-8, -7, -6, -5, -4, -3, -2, -1,  0,  1,  2,  3,  4,  5,  6,  7}, //             fs_sbr  = 16000 Hz
+    {-5, -4, -3, -2, -1,  0,  1,  2,  3,  4,  5,  6,  7,  9, 11, 13}, //             fs_sbr  = 22050 Hz
+    {-5, -3, -2, -1,  0,  1,  2,  3,  4,  5,  6,  7,  9, 11, 13, 16}, //             fs_sbr  = 24000 Hz
+    {-6, -4, -2, -1,  0,  1,  2,  3,  4,  5,  6,  7,  9, 11, 13, 16}, //             fs_sbr  = 32000 Hz
+    {-4, -2, -1,  0,  1,  2,  3,  4,  5,  6,  7,  9, 11, 13, 16, 20}, // 44100 Hz <= fs_sbr <= 64000 Hz
+    {-2, -1,  0,  1,  2,  3,  4,  5,  6,  7,  9, 11, 13, 16, 20, 24}, // 64000 Hz <  fs_sbr
+};
+
+/* First eight entries repeated at end to simplify SIMD implementations. */
+const DECLARE_ALIGNED(16, INTFLOAT, AAC_RENAME(ff_sbr_noise_table))[][2] = {
+{Q31(-0.99948153278296f), Q31(-0.59483417516607f)}, {Q31( 0.97113454393991f), Q31(-0.67528515225647f)},
+{Q31( 0.14130051758487f), Q31(-0.95090983575689f)}, {Q31(-0.47005496701697f), Q31(-0.37340549728647f)},
+{Q31( 0.80705063769351f), Q31( 0.29653668284408f)}, {Q31(-0.38981478896926f), Q31( 0.89572605717087f)},
+{Q31(-0.01053049862020f), Q31(-0.66959058036166f)}, {Q31(-0.91266367957293f), Q31(-0.11522938140034f)},
+{Q31( 0.54840422910309f), Q31( 0.75221367176302f)}, {Q31( 0.40009252867955f), Q31(-0.98929400334421f)},
+{Q31(-0.99867974711855f), Q31(-0.88147068645358f)}, {Q31(-0.95531076805040f), Q31( 0.90908757154593f)},
+{Q31(-0.45725933317144f), Q31(-0.56716323646760f)}, {Q31(-0.72929675029275f), Q31(-0.98008272727324f)},
+{Q31( 0.75622801399036f), Q31( 0.20950329995549f)}, {Q31( 0.07069442601050f), Q31(-0.78247898470706f)},
+{Q31( 0.74496252926055f), Q31(-0.91169004445807f)}, {Q31(-0.96440182703856f), Q31(-0.94739918296622f)},
+{Q31( 0.30424629369539f), Q31(-0.49438267012479f)}, {Q31( 0.66565033746925f), Q31( 0.64652935542491f)},
+{Q31( 0.91697008020594f), Q31( 0.17514097332009f)}, {Q31(-0.70774918760427f), Q31( 0.52548653416543f)},
+{Q31(-0.70051415345560f), Q31(-0.45340028808763f)}, {Q31(-0.99496513054797f), Q31(-0.90071908066973f)},
+{Q31( 0.98164490790123f), Q31(-0.77463155528697f)}, {Q31(-0.54671580548181f), Q31(-0.02570928536004f)},
+{Q31(-0.01689629065389f), Q31( 0.00287506445732f)}, {Q31(-0.86110349531986f), Q31( 0.42548583726477f)},
+{Q31(-0.98892980586032f), Q31(-0.87881132267556f)}, {Q31( 0.51756627678691f), Q31( 0.66926784710139f)},
+{Q31(-0.99635026409640f), Q31(-0.58107730574765f)}, {Q31(-0.99969370862163f), Q31( 0.98369989360250f)},
+{Q31( 0.55266258627194f), Q31( 0.59449057465591f)}, {Q31( 0.34581177741673f), Q31( 0.94879421061866f)},
+{Q31( 0.62664209577999f), Q31(-0.74402970906471f)}, {Q31(-0.77149701404973f), Q31(-0.33883658042801f)},
+{Q31(-0.91592244254432f), Q31( 0.03687901376713f)}, {Q31(-0.76285492357887f), Q31(-0.91371867919124f)},
+{Q31( 0.79788337195331f), Q31(-0.93180971199849f)}, {Q31( 0.54473080610200f), Q31(-0.11919206037186f)},
+{Q31(-0.85639281671058f), Q31( 0.42429854760451f)}, {Q31(-0.92882402971423f), Q31( 0.27871809078609f)},
+{Q31(-0.11708371046774f), Q31(-0.99800843444966f)}, {Q31( 0.21356749817493f), Q31(-0.90716295627033f)},
+{Q31(-0.76191692573909f), Q31( 0.99768118356265f)}, {Q31( 0.98111043100884f), Q31(-0.95854459734407f)},
+{Q31(-0.85913269895572f), Q31( 0.95766566168880f)}, {Q31(-0.93307242253692f), Q31( 0.49431757696466f)},
+{Q31( 0.30485754879632f), Q31(-0.70540034357529f)}, {Q31( 0.85289650925190f), Q31( 0.46766131791044f)},
+{Q31( 0.91328082618125f), Q31(-0.99839597361769f)}, {Q31(-0.05890199924154f), Q31( 0.70741827819497f)},
+{Q31( 0.28398686150148f), Q31( 0.34633555702188f)}, {Q31( 0.95258164539612f), Q31(-0.54893416026939f)},
+{Q31(-0.78566324168507f), Q31(-0.75568541079691f)}, {Q31(-0.95789495447877f), Q31(-0.20423194696966f)},
+{Q31( 0.82411158711197f), Q31( 0.96654618432562f)}, {Q31(-0.65185446735885f), Q31(-0.88734990773289f)},
+{Q31(-0.93643603134666f), Q31( 0.99870790442385f)}, {Q31( 0.91427159529618f), Q31(-0.98290505544444f)},
+{Q31(-0.70395684036886f), Q31( 0.58796798221039f)}, {Q31( 0.00563771969365f), Q31( 0.61768196727244f)},
+{Q31( 0.89065051931895f), Q31( 0.52783352697585f)}, {Q31(-0.68683707712762f), Q31( 0.80806944710339f)},
+{Q31( 0.72165342518718f), Q31(-0.69259857349564f)}, {Q31(-0.62928247730667f), Q31( 0.13627037407335f)},
+{Q31( 0.29938434065514f), Q31(-0.46051329682246f)}, {Q31(-0.91781958879280f), Q31(-0.74012716684186f)},
+{Q31( 0.99298717043688f), Q31( 0.40816610075661f)}, {Q31( 0.82368298622748f), Q31(-0.74036047190173f)},
+{Q31(-0.98512833386833f), Q31(-0.99972330709594f)}, {Q31(-0.95915368242257f), Q31(-0.99237800466040f)},
+{Q31(-0.21411126572790f), Q31(-0.93424819052545f)}, {Q31(-0.68821476106884f), Q31(-0.26892306315457f)},
+{Q31( 0.91851997982317f), Q31( 0.09358228901785f)}, {Q31(-0.96062769559127f), Q31( 0.36099095133739f)},
+{Q31( 0.51646184922287f), Q31(-0.71373332873917f)}, {Q31( 0.61130721139669f), Q31( 0.46950141175917f)},
+{Q31( 0.47336129371299f), Q31(-0.27333178296162f)}, {Q31( 0.90998308703519f), Q31( 0.96715662938132f)},
+{Q31( 0.44844799194357f), Q31( 0.99211574628306f)}, {Q31( 0.66614891079092f), Q31( 0.96590176169121f)},
+{Q31( 0.74922239129237f), Q31(-0.89879858826087f)}, {Q31(-0.99571588506485f), Q31( 0.52785521494349f)},
+{Q31( 0.97401082477563f), Q31(-0.16855870075190f)}, {Q31( 0.72683747733879f), Q31(-0.48060774432251f)},
+{Q31( 0.95432193457128f), Q31( 0.68849603408441f)}, {Q31(-0.72962208425191f), Q31(-0.76608443420917f)},
+{Q31(-0.85359479233537f), Q31( 0.88738125901579f)}, {Q31(-0.81412430338535f), Q31(-0.97480768049637f)},
+{Q31(-0.87930772356786f), Q31( 0.74748307690436f)}, {Q31(-0.71573331064977f), Q31(-0.98570608178923f)},
+{Q31( 0.83524300028228f), Q31( 0.83702537075163f)}, {Q31(-0.48086065601423f), Q31(-0.98848504923531f)},
+{Q31( 0.97139128574778f), Q31( 0.80093621198236f)}, {Q31( 0.51992825347895f), Q31( 0.80247631400510f)},
+{Q31(-0.00848591195325f), Q31(-0.76670128000486f)}, {Q31(-0.70294374303036f), Q31( 0.55359910445577f)},
+{Q31(-0.95894428168140f), Q31(-0.43265504344783f)}, {Q31( 0.97079252950321f), Q31( 0.09325857238682f)},
+{Q31(-0.92404293670797f), Q31( 0.85507704027855f)}, {Q31(-0.69506469500450f), Q31( 0.98633412625459f)},
+{Q31( 0.26559203620024f), Q31( 0.73314307966524f)}, {Q31( 0.28038443336943f), Q31( 0.14537913654427f)},
+{Q31(-0.74138124825523f), Q31( 0.99310339807762f)}, {Q31(-0.01752795995444f), Q31(-0.82616635284178f)},
+{Q31(-0.55126773094930f), Q31(-0.98898543862153f)}, {Q31( 0.97960898850996f), Q31(-0.94021446752851f)},
+{Q31(-0.99196309146936f), Q31( 0.67019017358456f)}, {Q31(-0.67684928085260f), Q31( 0.12631491649378f)},
+{Q31( 0.09140039465500f), Q31(-0.20537731453108f)}, {Q31(-0.71658965751996f), Q31(-0.97788200391224f)},
+{Q31( 0.81014640078925f), Q31( 0.53722648362443f)}, {Q31( 0.40616991671205f), Q31(-0.26469008598449f)},
+{Q31(-0.67680188682972f), Q31( 0.94502052337695f)}, {Q31( 0.86849774348749f), Q31(-0.18333598647899f)},
+{Q31(-0.99500381284851f), Q31(-0.02634122068550f)}, {Q31( 0.84329189340667f), Q31( 0.10406957462213f)},
+{Q31(-0.09215968531446f), Q31( 0.69540012101253f)}, {Q31( 0.99956173327206f), Q31(-0.12358542001404f)},
+{Q31(-0.79732779473535f), Q31(-0.91582524736159f)}, {Q31( 0.96349973642406f), Q31( 0.96640458041000f)},
+{Q31(-0.79942778496547f), Q31( 0.64323902822857f)}, {Q31(-0.11566039853896f), Q31( 0.28587846253726f)},
+{Q31(-0.39922954514662f), Q31( 0.94129601616966f)}, {Q31( 0.99089197565987f), Q31(-0.92062625581587f)},
+{Q31( 0.28631285179909f), Q31(-0.91035047143603f)}, {Q31(-0.83302725605608f), Q31(-0.67330410892084f)},
+{Q31( 0.95404443402072f), Q31( 0.49162765398743f)}, {Q31(-0.06449863579434f), Q31( 0.03250560813135f)},
+{Q31(-0.99575054486311f), Q31( 0.42389784469507f)}, {Q31(-0.65501142790847f), Q31( 0.82546114655624f)},
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+{Q31( 0.99971463703691f), Q31(-0.83889954253462f)}, {Q31(-0.75376385639978f), Q31( 0.00814643438625f)},
+{Q31( 0.93887685615875f), Q31(-0.11284528204636f)}, {Q31( 0.85126435782309f), Q31( 0.52349251543547f)},
+{Q31( 0.39701421446381f), Q31( 0.81779634174316f)}, {Q31(-0.37024464187437f), Q31(-0.87071656222959f)},
+{Q31(-0.36024828242896f), Q31( 0.34655735648287f)}, {Q31(-0.93388812549209f), Q31(-0.84476541096429f)},
+{Q31(-0.65298804552119f), Q31(-0.18439575450921f)}, {Q31( 0.11960319006843f), Q31( 0.99899346780168f)},
+{Q31( 0.94292565553160f), Q31( 0.83163906518293f)}, {Q31( 0.75081145286948f), Q31(-0.35533223142265f)},
+{Q31( 0.56721979748394f), Q31(-0.24076836414499f)}, {Q31( 0.46857766746029f), Q31(-0.30140233457198f)},
+{Q31( 0.97312313923635f), Q31(-0.99548191630031f)}, {Q31(-0.38299976567017f), Q31( 0.98516909715427f)},
+{Q31( 0.41025800019463f), Q31( 0.02116736935734f)}, {Q31( 0.09638062008048f), Q31( 0.04411984381457f)},
+{Q31(-0.85283249275397f), Q31( 0.91475563922421f)}, {Q31( 0.88866808958124f), Q31(-0.99735267083226f)},
+{Q31(-0.48202429536989f), Q31(-0.96805608884164f)}, {Q31( 0.27572582416567f), Q31( 0.58634753335832f)},
+{Q31(-0.65889129659168f), Q31( 0.58835634138583f)}, {Q31( 0.98838086953732f), Q31( 0.99994349600236f)},
+{Q31(-0.20651349620689f), Q31( 0.54593044066355f)}, {Q31(-0.62126416356920f), Q31(-0.59893681700392f)},
+{Q31( 0.20320105410437f), Q31(-0.86879180355289f)}, {Q31(-0.97790548600584f), Q31( 0.96290806999242f)},
+{Q31( 0.11112534735126f), Q31( 0.21484763313301f)}, {Q31(-0.41368337314182f), Q31( 0.28216837680365f)},
+{Q31( 0.24133038992960f), Q31( 0.51294362630238f)}, {Q31(-0.66393410674885f), Q31(-0.08249679629081f)},
+{Q31(-0.53697829178752f), Q31(-0.97649903936228f)}, {Q31(-0.97224737889348f), Q31( 0.22081333579837f)},
+{Q31( 0.87392477144549f), Q31(-0.12796173740361f)}, {Q31( 0.19050361015753f), Q31( 0.01602615387195f)},
+{Q31(-0.46353441212724f), Q31(-0.95249041539006f)}, {Q31(-0.07064096339021f), Q31(-0.94479803205886f)},
+{Q31(-0.92444085484466f), Q31(-0.10457590187436f)}, {Q31(-0.83822593578728f), Q31(-0.01695043208885f)},
+{Q31( 0.75214681811150f), Q31(-0.99955681042665f)}, {Q31(-0.42102998829339f), Q31( 0.99720941999394f)},
+{Q31(-0.72094786237696f), Q31(-0.35008961934255f)}, {Q31( 0.78843311019251f), Q31( 0.52851398958271f)},
+{Q31( 0.97394027897442f), Q31(-0.26695944086561f)}, {Q31( 0.99206463477946f), Q31(-0.57010120849429f)},
+{Q31( 0.76789609461795f), Q31(-0.76519356730966f)}, {Q31(-0.82002421836409f), Q31(-0.73530179553767f)},
+{Q31( 0.81924990025724f), Q31( 0.99698425250579f)}, {Q31(-0.26719850873357f), Q31( 0.68903369776193f)},
+{Q31(-0.43311260380975f), Q31( 0.85321815947490f)}, {Q31( 0.99194979673836f), Q31( 0.91876249766422f)},
+{Q31(-0.80692001248487f), Q31(-0.32627540663214f)}, {Q31( 0.43080003649976f), Q31(-0.21919095636638f)},
+{Q31( 0.67709491937357f), Q31(-0.95478075822906f)}, {Q31( 0.56151770568316f), Q31(-0.70693811747778f)},
+{Q31( 0.10831862810749f), Q31(-0.08628837174592f)}, {Q31( 0.91229417540436f), Q31(-0.65987351408410f)},
+{Q31(-0.48972893932274f), Q31( 0.56289246362686f)}, {Q31(-0.89033658689697f), Q31(-0.71656563987082f)},
+{Q31( 0.65269447475094f), Q31( 0.65916004833932f)}, {Q31( 0.67439478141121f), Q31(-0.81684380846796f)},
+{Q31(-0.47770832416973f), Q31(-0.16789556203025f)}, {Q31(-0.99715979260878f), Q31(-0.93565784007648f)},
+{Q31(-0.90889593602546f), Q31( 0.62034397054380f)}, {Q31(-0.06618622548177f), Q31(-0.23812217221359f)},
+{Q31( 0.99430266919728f), Q31( 0.18812555317553f)}, {Q31( 0.97686402381843f), Q31(-0.28664534366620f)},
+{Q31( 0.94813650221268f), Q31(-0.97506640027128f)}, {Q31(-0.95434497492853f), Q31(-0.79607978501983f)},
+{Q31(-0.49104783137150f), Q31( 0.32895214359663f)}, {Q31( 0.99881175120751f), Q31( 0.88993983831354f)},
+{Q31( 0.50449166760303f), Q31(-0.85995072408434f)}, {Q31( 0.47162891065108f), Q31(-0.18680204049569f)},
+{Q31(-0.62081581361840f), Q31( 0.75000676218956f)}, {Q31(-0.43867015250812f), Q31( 0.99998069244322f)},
+{Q31( 0.98630563232075f), Q31(-0.53578899600662f)}, {Q31(-0.61510362277374f), Q31(-0.89515019899997f)},
+{Q31(-0.03841517601843f), Q31(-0.69888815681179f)}, {Q31(-0.30102157304644f), Q31(-0.07667808922205f)},
+{Q31( 0.41881284182683f), Q31( 0.02188098922282f)}, {Q31(-0.86135454941237f), Q31( 0.98947480909359f)},
+{Q31( 0.67226861393788f), Q31(-0.13494389011014f)}, {Q31(-0.70737398842068f), Q31(-0.76547349325992f)},
+{Q31( 0.94044946687963f), Q31( 0.09026201157416f)}, {Q31(-0.82386352534327f), Q31( 0.08924768823676f)},
+{Q31(-0.32070666698656f), Q31( 0.50143421908753f)}, {Q31( 0.57593163224487f), Q31(-0.98966422921509f)},
+{Q31(-0.36326018419965f), Q31( 0.07440243123228f)}, {Q31( 0.99979044674350f), Q31(-0.14130287347405f)},
+{Q31(-0.92366023326932f), Q31(-0.97979298068180f)}, {Q31(-0.44607178518598f), Q31(-0.54233252016394f)},
+{Q31( 0.44226800932956f), Q31( 0.71326756742752f)}, {Q31( 0.03671907158312f), Q31( 0.63606389366675f)},
+{Q31( 0.52175424682195f), Q31(-0.85396826735705f)}, {Q31(-0.94701139690956f), Q31(-0.01826348194255f)},
+{Q31(-0.98759606946049f), Q31( 0.82288714303073f)}, {Q31( 0.87434794743625f), Q31( 0.89399495655433f)},
+{Q31(-0.93412041758744f), Q31( 0.41374052024363f)}, {Q31( 0.96063943315511f), Q31( 0.93116709541280f)},
+{Q31( 0.97534253457837f), Q31( 0.86150930812689f)}, {Q31( 0.99642466504163f), Q31( 0.70190043427512f)},
+{Q31(-0.94705089665984f), Q31(-0.29580042814306f)}, {Q31( 0.91599807087376f), Q31(-0.98147830385781f)},
+// Start of duplicated table
+{Q31(-0.99948153278296f), Q31(-0.59483417516607f)}, {Q31( 0.97113454393991f), Q31(-0.67528515225647f)},
+{Q31( 0.14130051758487f), Q31(-0.95090983575689f)}, {Q31(-0.47005496701697f), Q31(-0.37340549728647f)},
+{Q31( 0.80705063769351f), Q31( 0.29653668284408f)}, {Q31(-0.38981478896926f), Q31( 0.89572605717087f)},
+{Q31(-0.01053049862020f), Q31(-0.66959058036166f)}, {Q31(-0.91266367957293f), Q31(-0.11522938140034f)},
+};
+
+#endif /* AVCODEC_AACSBRDATA_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aactab.h

@@ -0,0 +1,186 @@
+/*
+ * AAC data declarations
+ * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
+ * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC data declarations
+ * @author Oded Shimon  ( ods15 ods15 dyndns org )
+ * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
+ */
+
+#ifndef AVCODEC_AACTAB_H
+#define AVCODEC_AACTAB_H
+
+#include "libavutil/mem.h"
+#include "aac.h"
+
+#include <stdint.h>
+
+/* NOTE:
+ * Tables in this file are shared by the AAC decoders and encoder
+ */
+
+extern float ff_aac_pow2sf_tab[428];
+extern float ff_aac_pow34sf_tab[428];
+
+static inline void ff_aac_tableinit(void)
+{
+    int i;
+
+    /* 2^(i/16) for 0 <= i <= 15 */
+    static const float exp2_lut[] = {
+        1.00000000000000000000,
+        1.04427378242741384032,
+        1.09050773266525765921,
+        1.13878863475669165370,
+        1.18920711500272106672,
+        1.24185781207348404859,
+        1.29683955465100966593,
+        1.35425554693689272830,
+        1.41421356237309504880,
+        1.47682614593949931139,
+        1.54221082540794082361,
+        1.61049033194925430818,
+        1.68179283050742908606,
+        1.75625216037329948311,
+        1.83400808640934246349,
+        1.91520656139714729387,
+    };
+    float t1 = 8.8817841970012523233890533447265625e-16; // 2^(-50)
+    float t2 = 3.63797880709171295166015625e-12; // 2^(-38)
+    int t1_inc_cur, t2_inc_cur;
+    int t1_inc_prev = 0;
+    int t2_inc_prev = 8;
+
+    for (i = 0; i < 428; i++) {
+        t1_inc_cur = 4 * (i % 4);
+        t2_inc_cur = (8 + 3*i) % 16;
+        if (t1_inc_cur < t1_inc_prev)
+            t1 *= 2;
+        if (t2_inc_cur < t2_inc_prev)
+            t2 *= 2;
+        // A much more efficient and accurate way of doing:
+        // ff_aac_pow2sf_tab[i] = pow(2, (i - POW_SF2_ZERO) / 4.0);
+        // ff_aac_pow34sf_tab[i] = pow(ff_aac_pow2sf_tab[i], 3.0/4.0);
+        ff_aac_pow2sf_tab[i] = t1 * exp2_lut[t1_inc_cur];
+        ff_aac_pow34sf_tab[i] = t2 * exp2_lut[t2_inc_cur];
+        t1_inc_prev = t1_inc_cur;
+        t2_inc_prev = t2_inc_cur;
+    }
+}
+
+/* @name ltp_coef
+ * Table of the LTP coefficients
+ */
+static const INTFLOAT ltp_coef[8] = {
+    Q30(0.570829), Q30(0.696616), Q30(0.813004), Q30(0.911304),
+    Q30(0.984900), Q30(1.067894), Q30(1.194601), Q30(1.369533),
+};
+
+/* @name tns_tmp2_map
+ * Tables of the tmp2[] arrays of LPC coefficients used for TNS.
+ * The suffix _M_N[] indicate the values of coef_compress and coef_res
+ * respectively.
+ * @{
+ */
+static const INTFLOAT tns_tmp2_map_1_3[4] = {
+    Q31(0.00000000), Q31(-0.43388373),  Q31(0.64278758),  Q31(0.34202015),
+};
+
+static const INTFLOAT tns_tmp2_map_0_3[8] = {
+    Q31(0.00000000), Q31(-0.43388373), Q31(-0.78183150), Q31(-0.97492790),
+    Q31(0.98480773), Q31( 0.86602539), Q31( 0.64278758), Q31( 0.34202015),
+};
+
+static const INTFLOAT tns_tmp2_map_1_4[8] = {
+    Q31(0.00000000), Q31(-0.20791170), Q31(-0.40673664), Q31(-0.58778524),
+    Q31(0.67369562), Q31( 0.52643216), Q31( 0.36124167), Q31( 0.18374951),
+};
+
+static const INTFLOAT tns_tmp2_map_0_4[16] = {
+    Q31( 0.00000000), Q31(-0.20791170), Q31(-0.40673664), Q31(-0.58778524),
+    Q31(-0.74314481), Q31(-0.86602539), Q31(-0.95105654), Q31(-0.99452192),
+    Q31( 0.99573416), Q31( 0.96182561), Q31( 0.89516330), Q31( 0.79801720),
+    Q31( 0.67369562), Q31( 0.52643216), Q31( 0.36124167), Q31( 0.18374951),
+};
+
+static const INTFLOAT * const tns_tmp2_map[4] = {
+    tns_tmp2_map_0_3,
+    tns_tmp2_map_0_4,
+    tns_tmp2_map_1_3,
+    tns_tmp2_map_1_4
+};
+// @}
+
+/* @name window coefficients
+ * @{
+ */
+DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_long_1024)[1024];
+DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_short_128)[128];
+DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_long_960)[960];
+DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_short_120)[120];
+DECLARE_ALIGNED(32, extern int,    ff_aac_kbd_long_1024_fixed)[1024];
+DECLARE_ALIGNED(32, extern int,    ff_aac_kbd_long_512_fixed)[512];
+DECLARE_ALIGNED(32, extern int,    ff_aac_kbd_short_128_fixed)[128];
+DECLARE_ALIGNED(32, extern const float, ff_aac_eld_window_512)[1920];
+DECLARE_ALIGNED(32, extern const int,   ff_aac_eld_window_512_fixed)[1920];
+DECLARE_ALIGNED(32, extern const float, ff_aac_eld_window_480)[1800];
+DECLARE_ALIGNED(32, extern const int,   ff_aac_eld_window_480_fixed)[1800];
+// @}
+
+/* @name number of scalefactor window bands for long and short transform windows respectively
+ * @{
+ */
+extern const uint8_t ff_aac_num_swb_1024[];
+extern const uint8_t ff_aac_num_swb_960 [];
+extern const uint8_t ff_aac_num_swb_512 [];
+extern const uint8_t ff_aac_num_swb_480 [];
+extern const uint8_t ff_aac_num_swb_128 [];
+extern const uint8_t ff_aac_num_swb_120 [];
+// @}
+
+extern const uint8_t ff_aac_pred_sfb_max [];
+
+extern const uint32_t ff_aac_scalefactor_code[121];
+extern const uint8_t  ff_aac_scalefactor_bits[121];
+
+extern const uint16_t * const ff_aac_spectral_codes[11];
+extern const uint8_t  * const ff_aac_spectral_bits [11];
+extern const uint16_t  ff_aac_spectral_sizes[11];
+
+extern const float *ff_aac_codebook_vectors[];
+extern const float *ff_aac_codebook_vector_vals[];
+extern const uint16_t *ff_aac_codebook_vector_idx[];
+
+extern const uint16_t * const ff_swb_offset_1024[13];
+extern const uint16_t * const ff_swb_offset_960 [13];
+extern const uint16_t * const ff_swb_offset_512 [13];
+extern const uint16_t * const ff_swb_offset_480 [13];
+extern const uint16_t * const ff_swb_offset_128 [13];
+extern const uint16_t * const ff_swb_offset_120 [13];
+
+extern const uint8_t ff_tns_max_bands_1024[13];
+extern const uint8_t ff_tns_max_bands_512 [13];
+extern const uint8_t ff_tns_max_bands_480 [13];
+extern const uint8_t ff_tns_max_bands_128 [13];
+
+#endif /* AVCODEC_AACTAB_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aarch64/Makefile

@@ -0,0 +1,56 @@
+# subsystems
+OBJS-$(CONFIG_FFT)                      += aarch64/fft_init_aarch64.o
+OBJS-$(CONFIG_FMTCONVERT)               += aarch64/fmtconvert_init.o
+OBJS-$(CONFIG_H264CHROMA)               += aarch64/h264chroma_init_aarch64.o
+OBJS-$(CONFIG_H264DSP)                  += aarch64/h264dsp_init_aarch64.o
+OBJS-$(CONFIG_H264PRED)                 += aarch64/h264pred_init.o
+OBJS-$(CONFIG_H264QPEL)                 += aarch64/h264qpel_init_aarch64.o
+OBJS-$(CONFIG_HPELDSP)                  += aarch64/hpeldsp_init_aarch64.o
+OBJS-$(CONFIG_MPEGAUDIODSP)             += aarch64/mpegaudiodsp_init.o
+OBJS-$(CONFIG_NEON_CLOBBER_TEST)        += aarch64/neontest.o
+OBJS-$(CONFIG_VIDEODSP)                 += aarch64/videodsp_init.o
+
+# decoders/encoders
+OBJS-$(CONFIG_AAC_DECODER)              += aarch64/aacpsdsp_init_aarch64.o \
+                                           aarch64/sbrdsp_init_aarch64.o
+OBJS-$(CONFIG_DCA_DECODER)              += aarch64/synth_filter_init.o
+OBJS-$(CONFIG_RV40_DECODER)             += aarch64/rv40dsp_init_aarch64.o
+OBJS-$(CONFIG_VC1DSP)                   += aarch64/vc1dsp_init_aarch64.o
+OBJS-$(CONFIG_VORBIS_DECODER)           += aarch64/vorbisdsp_init.o
+OBJS-$(CONFIG_VP9_DECODER)              += aarch64/vp9dsp_init_10bpp_aarch64.o \
+                                           aarch64/vp9dsp_init_12bpp_aarch64.o \
+                                           aarch64/vp9dsp_init_aarch64.o
+
+# ARMv8 optimizations
+
+# subsystems
+ARMV8-OBJS-$(CONFIG_VIDEODSP)           += aarch64/videodsp.o
+
+# NEON optimizations
+
+# subsystems
+NEON-OBJS-$(CONFIG_AAC_DECODER)         += aarch64/sbrdsp_neon.o
+NEON-OBJS-$(CONFIG_FFT)                 += aarch64/fft_neon.o
+NEON-OBJS-$(CONFIG_FMTCONVERT)          += aarch64/fmtconvert_neon.o
+NEON-OBJS-$(CONFIG_H264CHROMA)          += aarch64/h264cmc_neon.o
+NEON-OBJS-$(CONFIG_H264DSP)             += aarch64/h264dsp_neon.o              \
+                                           aarch64/h264idct_neon.o
+NEON-OBJS-$(CONFIG_H264PRED)            += aarch64/h264pred_neon.o
+NEON-OBJS-$(CONFIG_H264QPEL)            += aarch64/h264qpel_neon.o             \
+                                           aarch64/hpeldsp_neon.o
+NEON-OBJS-$(CONFIG_HPELDSP)             += aarch64/hpeldsp_neon.o
+NEON-OBJS-$(CONFIG_IDCTDSP)             += aarch64/idctdsp_init_aarch64.o      \
+                                           aarch64/simple_idct_neon.o
+NEON-OBJS-$(CONFIG_MDCT)                += aarch64/mdct_neon.o
+NEON-OBJS-$(CONFIG_MPEGAUDIODSP)        += aarch64/mpegaudiodsp_neon.o
+
+# decoders/encoders
+NEON-OBJS-$(CONFIG_AAC_DECODER)         += aarch64/aacpsdsp_neon.o
+NEON-OBJS-$(CONFIG_DCA_DECODER)         += aarch64/synth_filter_neon.o
+NEON-OBJS-$(CONFIG_VORBIS_DECODER)      += aarch64/vorbisdsp_neon.o
+NEON-OBJS-$(CONFIG_VP9_DECODER)         += aarch64/vp9itxfm_16bpp_neon.o       \
+                                           aarch64/vp9itxfm_neon.o             \
+                                           aarch64/vp9lpf_16bpp_neon.o         \
+                                           aarch64/vp9lpf_neon.o               \
+                                           aarch64/vp9mc_16bpp_neon.o          \
+                                           aarch64/vp9mc_neon.o

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aarch64/aacpsdsp_init_aarch64.c

@@ -0,0 +1,48 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "config.h"
+
+#include "libavutil/aarch64/cpu.h"
+#include "libavcodec/aacpsdsp.h"
+
+void ff_ps_add_squares_neon(float *dst, const float (*src)[2], int n);
+void ff_ps_mul_pair_single_neon(float (*dst)[2], float (*src0)[2],
+                                float *src1, int n);
+void ff_ps_hybrid_analysis_neon(float (*out)[2], float (*in)[2],
+                                const float (*filter)[8][2],
+                                ptrdiff_t stride, int n);
+void ff_ps_stereo_interpolate_neon(float (*l)[2], float (*r)[2],
+                                   float h[2][4], float h_step[2][4],
+                                   int len);
+void ff_ps_stereo_interpolate_ipdopd_neon(float (*l)[2], float (*r)[2],
+                                          float h[2][4], float h_step[2][4],
+                                          int len);
+
+av_cold void ff_psdsp_init_aarch64(PSDSPContext *s)
+{
+    int cpu_flags = av_get_cpu_flags();
+
+    if (have_neon(cpu_flags)) {
+        s->add_squares           = ff_ps_add_squares_neon;
+        s->mul_pair_single       = ff_ps_mul_pair_single_neon;
+        s->hybrid_analysis       = ff_ps_hybrid_analysis_neon;
+        s->stereo_interpolate[0] = ff_ps_stereo_interpolate_neon;
+        s->stereo_interpolate[1] = ff_ps_stereo_interpolate_ipdopd_neon;
+    }
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aarch64/asm-offsets.h

@@ -0,0 +1,33 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AARCH64_ASM_OFFSETS_H
+#define AVCODEC_AARCH64_ASM_OFFSETS_H
+
+/* CeltIMDCTContext */
+#define CELT_EXPTAB                     0x20
+#define CELT_FFT_N                      0x00
+#define CELT_LEN2                       0x04
+#define CELT_LEN4                       (CELT_LEN2 + 0x4)   // loaded as pair
+#define CELT_TMP                        0x10
+#define CELT_TWIDDLE                    (CELT_TMP + 0x8)    // loaded as pair
+
+/* FFTContext */
+#define IMDCT_HALF                      0x48
+
+#endif /* AVCODEC_AARCH64_ASM_OFFSETS_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aarch64/cabac.h

@@ -0,0 +1,104 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AARCH64_CABAC_H
+#define AVCODEC_AARCH64_CABAC_H
+
+#include "config.h"
+#if HAVE_INLINE_ASM
+
+#include "libavutil/attributes.h"
+#include "libavutil/internal.h"
+#include "libavcodec/cabac.h"
+
+#define get_cabac_inline get_cabac_inline_aarch64
+static av_always_inline int get_cabac_inline_aarch64(CABACContext *c,
+                                                     uint8_t *const state)
+{
+    int bit;
+    void *reg_a, *reg_b, *reg_c, *tmp;
+
+    __asm__ volatile(
+        "ldrb       %w[bit]       , [%[state]]                  \n\t"
+        "add        %[r_b]        , %[tables]   , %[lps_off]    \n\t"
+        "mov        %w[tmp]       , %w[range]                   \n\t"
+        "and        %w[range]     , %w[range]   , #0xC0         \n\t"
+        "lsl        %w[r_c]       , %w[range]   , #1            \n\t"
+        "add        %[r_b]        , %[r_b]      , %w[bit], UXTW \n\t"
+        "ldrb       %w[range]     , [%[r_b], %w[r_c], SXTW]     \n\t"
+        "sub        %w[r_c]       , %w[tmp]     , %w[range]     \n\t"
+        "lsl        %w[tmp]       , %w[r_c]     , #17           \n\t"
+        "cmp        %w[tmp]       , %w[low]                     \n\t"
+        "csel       %w[tmp]       , %w[tmp]     , wzr      , cc \n\t"
+        "csel       %w[range]     , %w[r_c]     , %w[range], gt \n\t"
+        "cinv       %w[bit]       , %w[bit]     , cc            \n\t"
+        "sub        %w[low]       , %w[low]     , %w[tmp]       \n\t"
+        "add        %[r_b]        , %[tables]   , %[norm_off]   \n\t"
+        "add        %[r_a]        , %[tables]   , %[mlps_off]   \n\t"
+        "ldrb       %w[tmp]       , [%[r_b], %w[range], SXTW]   \n\t"
+        "ldrb       %w[r_a]       , [%[r_a], %w[bit], SXTW]     \n\t"
+        "lsl        %w[low]       , %w[low]     , %w[tmp]       \n\t"
+        "lsl        %w[range]     , %w[range]   , %w[tmp]       \n\t"
+        "uxth       %w[r_c]       , %w[low]                     \n\t"
+        "strb       %w[r_a]       , [%[state]]                  \n\t"
+        "cbnz       %w[r_c]       , 2f                          \n\t"
+        "ldr        %[r_c]        , [%[c], %[byte]]             \n\t"
+        "ldr        %[r_a]        , [%[c], %[end]]              \n\t"
+        "ldrh       %w[tmp]       , [%[r_c]]                    \n\t"
+        "cmp        %[r_c]        , %[r_a]                      \n\t"
+        "b.ge       1f                                          \n\t"
+        "add        %[r_a]        , %[r_c]      , #2            \n\t"
+        "str        %[r_a]        , [%[c], %[byte]]             \n\t"
+        "1:                                                     \n\t"
+        "sub        %w[r_c]       , %w[low]     , #1            \n\t"
+        "eor        %w[r_c]       , %w[r_c]     , %w[low]       \n\t"
+        "rev        %w[tmp]       , %w[tmp]                     \n\t"
+        "lsr        %w[r_c]       , %w[r_c]     , #15           \n\t"
+        "lsr        %w[tmp]       , %w[tmp]     , #15           \n\t"
+        "ldrb       %w[r_c]       , [%[r_b], %w[r_c], SXTW]     \n\t"
+        "mov        %w[r_b]       , #0xFFFF                     \n\t"
+        "mov        %w[r_a]       , #7                          \n\t"
+        "sub        %w[tmp]       , %w[tmp]     , %w[r_b]       \n\t"
+        "sub        %w[r_c]       , %w[r_a]     , %w[r_c]       \n\t"
+        "lsl        %w[tmp]       , %w[tmp]     , %w[r_c]       \n\t"
+        "add        %w[low]       , %w[low]     , %w[tmp]       \n\t"
+        "2:                                                     \n\t"
+        :    [bit]"=&r"(bit),
+             [low]"+&r"(c->low),
+           [range]"+&r"(c->range),
+             [r_a]"=&r"(reg_a),
+             [r_b]"=&r"(reg_b),
+             [r_c]"=&r"(reg_c),
+             [tmp]"=&r"(tmp)
+        :        [c]"r"(c),
+             [state]"r"(state),
+            [tables]"r"(ff_h264_cabac_tables),
+              [byte]"i"(offsetof(CABACContext, bytestream)),
+               [end]"i"(offsetof(CABACContext, bytestream_end)),
+          [norm_off]"I"(H264_NORM_SHIFT_OFFSET),
+           [lps_off]"I"(H264_LPS_RANGE_OFFSET),
+          [mlps_off]"I"(H264_MLPS_STATE_OFFSET + 128)
+        : "memory", "cc"
+        );
+
+    return bit & 1;
+}
+
+#endif /* HAVE_INLINE_ASM */
+
+#endif /* AVCODEC_AARCH64_CABAC_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aarch64/fft_init_aarch64.c

@@ -0,0 +1,50 @@
+/*
+ * Copyright (c) 2009 Mans Rullgard <mans@mansr.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "config.h"
+
+#include "libavutil/attributes.h"
+#include "libavutil/cpu.h"
+#include "libavutil/aarch64/cpu.h"
+
+#include "libavcodec/fft.h"
+
+void ff_fft_permute_neon(FFTContext *s, FFTComplex *z);
+void ff_fft_calc_neon(FFTContext *s, FFTComplex *z);
+
+void ff_imdct_calc_neon(FFTContext *s, FFTSample *output, const FFTSample *input);
+void ff_imdct_half_neon(FFTContext *s, FFTSample *output, const FFTSample *input);
+void ff_mdct_calc_neon(FFTContext *s, FFTSample *output, const FFTSample *input);
+
+av_cold void ff_fft_init_aarch64(FFTContext *s)
+{
+    int cpu_flags = av_get_cpu_flags();
+
+    if (have_neon(cpu_flags)) {
+        s->fft_permute  = ff_fft_permute_neon;
+        s->fft_calc     = ff_fft_calc_neon;
+#if CONFIG_MDCT
+        s->imdct_calc   = ff_imdct_calc_neon;
+        s->imdct_half   = ff_imdct_half_neon;
+        s->mdct_calc    = ff_mdct_calc_neon;
+        s->mdct_permutation = FF_MDCT_PERM_INTERLEAVE;
+#endif
+    }
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aarch64/fmtconvert_init.c

@@ -0,0 +1,43 @@
+/*
+ * ARM optimized Format Conversion Utils
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdint.h>
+
+#include "libavutil/attributes.h"
+#include "libavutil/aarch64/cpu.h"
+#include "libavcodec/avcodec.h"
+#include "libavcodec/fmtconvert.h"
+
+void ff_int32_to_float_fmul_array8_neon(FmtConvertContext *c, float *dst,
+                                        const int32_t *src, const float *mul,
+                                        int len);
+void ff_int32_to_float_fmul_scalar_neon(float *dst, const int32_t *src,
+                                        float mul, int len);
+
+av_cold void ff_fmt_convert_init_aarch64(FmtConvertContext *c,
+                                         AVCodecContext *avctx)
+{
+    int cpu_flags = av_get_cpu_flags();
+
+    if (have_neon(cpu_flags)) {
+        c->int32_to_float_fmul_array8 = ff_int32_to_float_fmul_array8_neon;
+        c->int32_to_float_fmul_scalar = ff_int32_to_float_fmul_scalar_neon;
+    }
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aarch64/h264chroma_init_aarch64.c

@@ -0,0 +1,59 @@
+/*
+ * ARM NEON optimised H.264 chroma functions
+ * Copyright (c) 2008 Mans Rullgard <mans@mansr.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdint.h>
+
+#include "libavutil/attributes.h"
+#include "libavutil/cpu.h"
+#include "libavutil/aarch64/cpu.h"
+#include "libavcodec/h264chroma.h"
+
+#include "config.h"
+
+void ff_put_h264_chroma_mc8_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride,
+                                 int h, int x, int y);
+void ff_put_h264_chroma_mc4_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride,
+                                 int h, int x, int y);
+void ff_put_h264_chroma_mc2_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride,
+                                 int h, int x, int y);
+
+void ff_avg_h264_chroma_mc8_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride,
+                                 int h, int x, int y);
+void ff_avg_h264_chroma_mc4_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride,
+                                 int h, int x, int y);
+void ff_avg_h264_chroma_mc2_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride,
+                                 int h, int x, int y);
+
+av_cold void ff_h264chroma_init_aarch64(H264ChromaContext *c, int bit_depth)
+{
+    const int high_bit_depth = bit_depth > 8;
+    int cpu_flags = av_get_cpu_flags();
+
+    if (have_neon(cpu_flags) && !high_bit_depth) {
+        c->put_h264_chroma_pixels_tab[0] = ff_put_h264_chroma_mc8_neon;
+        c->put_h264_chroma_pixels_tab[1] = ff_put_h264_chroma_mc4_neon;
+        c->put_h264_chroma_pixels_tab[2] = ff_put_h264_chroma_mc2_neon;
+
+        c->avg_h264_chroma_pixels_tab[0] = ff_avg_h264_chroma_mc8_neon;
+        c->avg_h264_chroma_pixels_tab[1] = ff_avg_h264_chroma_mc4_neon;
+        c->avg_h264_chroma_pixels_tab[2] = ff_avg_h264_chroma_mc2_neon;
+    }
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aarch64/h264dsp_init_aarch64.c

@@ -0,0 +1,102 @@
+/*
+ * Copyright (c) 2010 Mans Rullgard <mans@mansr.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdint.h>
+
+#include "libavutil/attributes.h"
+#include "libavutil/cpu.h"
+#include "libavutil/aarch64/cpu.h"
+#include "libavcodec/h264dsp.h"
+
+void ff_h264_v_loop_filter_luma_neon(uint8_t *pix, int stride, int alpha,
+                                     int beta, int8_t *tc0);
+void ff_h264_h_loop_filter_luma_neon(uint8_t *pix, int stride, int alpha,
+                                     int beta, int8_t *tc0);
+void ff_h264_v_loop_filter_chroma_neon(uint8_t *pix, int stride, int alpha,
+                                       int beta, int8_t *tc0);
+void ff_h264_h_loop_filter_chroma_neon(uint8_t *pix, int stride, int alpha,
+                                       int beta, int8_t *tc0);
+
+void ff_weight_h264_pixels_16_neon(uint8_t *dst, ptrdiff_t stride, int height,
+                                   int log2_den, int weight, int offset);
+void ff_weight_h264_pixels_8_neon(uint8_t *dst, ptrdiff_t stride, int height,
+                                  int log2_den, int weight, int offset);
+void ff_weight_h264_pixels_4_neon(uint8_t *dst, ptrdiff_t stride, int height,
+                                  int log2_den, int weight, int offset);
+
+void ff_biweight_h264_pixels_16_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride,
+                                     int height, int log2_den, int weightd,
+                                     int weights, int offset);
+void ff_biweight_h264_pixels_8_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride,
+                                    int height, int log2_den, int weightd,
+                                    int weights, int offset);
+void ff_biweight_h264_pixels_4_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride,
+                                    int height, int log2_den, int weightd,
+                                    int weights, int offset);
+
+void ff_h264_idct_add_neon(uint8_t *dst, int16_t *block, int stride);
+void ff_h264_idct_dc_add_neon(uint8_t *dst, int16_t *block, int stride);
+void ff_h264_idct_add16_neon(uint8_t *dst, const int *block_offset,
+                             int16_t *block, int stride,
+                             const uint8_t nnzc[6*8]);
+void ff_h264_idct_add16intra_neon(uint8_t *dst, const int *block_offset,
+                                  int16_t *block, int stride,
+                                  const uint8_t nnzc[6*8]);
+void ff_h264_idct_add8_neon(uint8_t **dest, const int *block_offset,
+                            int16_t *block, int stride,
+                            const uint8_t nnzc[6*8]);
+
+void ff_h264_idct8_add_neon(uint8_t *dst, int16_t *block, int stride);
+void ff_h264_idct8_dc_add_neon(uint8_t *dst, int16_t *block, int stride);
+void ff_h264_idct8_add4_neon(uint8_t *dst, const int *block_offset,
+                             int16_t *block, int stride,
+                             const uint8_t nnzc[6*8]);
+
+av_cold void ff_h264dsp_init_aarch64(H264DSPContext *c, const int bit_depth,
+                                     const int chroma_format_idc)
+{
+    int cpu_flags = av_get_cpu_flags();
+
+    if (have_neon(cpu_flags) && bit_depth == 8) {
+        c->h264_v_loop_filter_luma   = ff_h264_v_loop_filter_luma_neon;
+        c->h264_h_loop_filter_luma   = ff_h264_h_loop_filter_luma_neon;
+        c->h264_v_loop_filter_chroma = ff_h264_v_loop_filter_chroma_neon;
+        if (chroma_format_idc <= 1)
+        c->h264_h_loop_filter_chroma = ff_h264_h_loop_filter_chroma_neon;
+
+        c->weight_h264_pixels_tab[0] = ff_weight_h264_pixels_16_neon;
+        c->weight_h264_pixels_tab[1] = ff_weight_h264_pixels_8_neon;
+        c->weight_h264_pixels_tab[2] = ff_weight_h264_pixels_4_neon;
+
+        c->biweight_h264_pixels_tab[0] = ff_biweight_h264_pixels_16_neon;
+        c->biweight_h264_pixels_tab[1] = ff_biweight_h264_pixels_8_neon;
+        c->biweight_h264_pixels_tab[2] = ff_biweight_h264_pixels_4_neon;
+
+        c->h264_idct_add        = ff_h264_idct_add_neon;
+        c->h264_idct_dc_add     = ff_h264_idct_dc_add_neon;
+        c->h264_idct_add16      = ff_h264_idct_add16_neon;
+        c->h264_idct_add16intra = ff_h264_idct_add16intra_neon;
+        if (chroma_format_idc <= 1)
+            c->h264_idct_add8   = ff_h264_idct_add8_neon;
+        c->h264_idct8_add       = ff_h264_idct8_add_neon;
+        c->h264_idct8_dc_add    = ff_h264_idct8_dc_add_neon;
+        c->h264_idct8_add4      = ff_h264_idct8_add4_neon;
+    }
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aarch64/h264pred_init.c

@@ -0,0 +1,93 @@
+/*
+ * Copyright (c) 2009 Mans Rullgard <mans@mansr.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdint.h>
+
+#include "libavutil/attributes.h"
+#include "libavutil/aarch64/cpu.h"
+#include "libavcodec/avcodec.h"
+#include "libavcodec/h264pred.h"
+
+void ff_pred16x16_vert_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred16x16_hor_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred16x16_plane_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred16x16_dc_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred16x16_128_dc_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred16x16_left_dc_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred16x16_top_dc_neon(uint8_t *src, ptrdiff_t stride);
+
+void ff_pred8x8_vert_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred8x8_hor_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred8x8_plane_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred8x8_dc_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred8x8_128_dc_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred8x8_left_dc_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred8x8_top_dc_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred8x8_l0t_dc_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred8x8_0lt_dc_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred8x8_l00_dc_neon(uint8_t *src, ptrdiff_t stride);
+void ff_pred8x8_0l0_dc_neon(uint8_t *src, ptrdiff_t stride);
+
+static av_cold void h264_pred_init_neon(H264PredContext *h, int codec_id,
+                                        const int bit_depth,
+                                        const int chroma_format_idc)
+{
+    const int high_depth = bit_depth > 8;
+
+    if (high_depth)
+        return;
+
+    if (chroma_format_idc <= 1) {
+        h->pred8x8[VERT_PRED8x8     ] = ff_pred8x8_vert_neon;
+        h->pred8x8[HOR_PRED8x8      ] = ff_pred8x8_hor_neon;
+        if (codec_id != AV_CODEC_ID_VP7 && codec_id != AV_CODEC_ID_VP8)
+            h->pred8x8[PLANE_PRED8x8] = ff_pred8x8_plane_neon;
+        h->pred8x8[DC_128_PRED8x8   ] = ff_pred8x8_128_dc_neon;
+        if (codec_id != AV_CODEC_ID_RV40 && codec_id != AV_CODEC_ID_VP7 &&
+            codec_id != AV_CODEC_ID_VP8) {
+            h->pred8x8[DC_PRED8x8     ] = ff_pred8x8_dc_neon;
+            h->pred8x8[LEFT_DC_PRED8x8] = ff_pred8x8_left_dc_neon;
+            h->pred8x8[TOP_DC_PRED8x8 ] = ff_pred8x8_top_dc_neon;
+            h->pred8x8[ALZHEIMER_DC_L0T_PRED8x8] = ff_pred8x8_l0t_dc_neon;
+            h->pred8x8[ALZHEIMER_DC_0LT_PRED8x8] = ff_pred8x8_0lt_dc_neon;
+            h->pred8x8[ALZHEIMER_DC_L00_PRED8x8] = ff_pred8x8_l00_dc_neon;
+            h->pred8x8[ALZHEIMER_DC_0L0_PRED8x8] = ff_pred8x8_0l0_dc_neon;
+        }
+    }
+
+    h->pred16x16[DC_PRED8x8     ] = ff_pred16x16_dc_neon;
+    h->pred16x16[VERT_PRED8x8   ] = ff_pred16x16_vert_neon;
+    h->pred16x16[HOR_PRED8x8    ] = ff_pred16x16_hor_neon;
+    h->pred16x16[LEFT_DC_PRED8x8] = ff_pred16x16_left_dc_neon;
+    h->pred16x16[TOP_DC_PRED8x8 ] = ff_pred16x16_top_dc_neon;
+    h->pred16x16[DC_128_PRED8x8 ] = ff_pred16x16_128_dc_neon;
+    if (codec_id != AV_CODEC_ID_SVQ3 && codec_id != AV_CODEC_ID_RV40 &&
+        codec_id != AV_CODEC_ID_VP7 && codec_id != AV_CODEC_ID_VP8)
+        h->pred16x16[PLANE_PRED8x8  ] = ff_pred16x16_plane_neon;
+}
+
+av_cold void ff_h264_pred_init_aarch64(H264PredContext *h, int codec_id,
+                                       int bit_depth, const int chroma_format_idc)
+{
+    int cpu_flags = av_get_cpu_flags();
+
+    if (have_neon(cpu_flags))
+        h264_pred_init_neon(h, codec_id, bit_depth, chroma_format_idc);
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aarch64/h264qpel_init_aarch64.c

@@ -0,0 +1,172 @@
+/*
+ * ARM NEON optimised DSP functions
+ * Copyright (c) 2008 Mans Rullgard <mans@mansr.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdint.h>
+
+#include "config.h"
+#include "libavutil/attributes.h"
+#include "libavutil/cpu.h"
+#include "libavutil/aarch64/cpu.h"
+#include "libavcodec/h264qpel.h"
+
+void ff_put_h264_qpel16_mc00_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc10_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc20_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc30_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc01_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc11_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc21_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc31_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc02_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc12_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc22_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc32_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc03_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc13_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc23_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel16_mc33_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+
+void ff_put_h264_qpel8_mc00_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc10_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc20_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc30_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc01_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc11_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc21_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc31_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc02_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc12_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc22_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc32_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc03_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc13_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc23_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_put_h264_qpel8_mc33_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+
+void ff_avg_h264_qpel16_mc00_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc10_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc20_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc30_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc01_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc11_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc21_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc31_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc02_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc12_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc22_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc32_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc03_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc13_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc23_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel16_mc33_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+
+void ff_avg_h264_qpel8_mc00_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc10_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc20_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc30_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc01_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc11_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc21_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc31_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc02_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc12_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc22_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc32_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc03_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc13_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc23_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+void ff_avg_h264_qpel8_mc33_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
+
+av_cold void ff_h264qpel_init_aarch64(H264QpelContext *c, int bit_depth)
+{
+    const int high_bit_depth = bit_depth > 8;
+    int cpu_flags = av_get_cpu_flags();
+
+    if (have_neon(cpu_flags) && !high_bit_depth) {
+        c->put_h264_qpel_pixels_tab[0][ 0] = ff_put_h264_qpel16_mc00_neon;
+        c->put_h264_qpel_pixels_tab[0][ 1] = ff_put_h264_qpel16_mc10_neon;
+        c->put_h264_qpel_pixels_tab[0][ 2] = ff_put_h264_qpel16_mc20_neon;
+        c->put_h264_qpel_pixels_tab[0][ 3] = ff_put_h264_qpel16_mc30_neon;
+        c->put_h264_qpel_pixels_tab[0][ 4] = ff_put_h264_qpel16_mc01_neon;
+        c->put_h264_qpel_pixels_tab[0][ 5] = ff_put_h264_qpel16_mc11_neon;
+        c->put_h264_qpel_pixels_tab[0][ 6] = ff_put_h264_qpel16_mc21_neon;
+        c->put_h264_qpel_pixels_tab[0][ 7] = ff_put_h264_qpel16_mc31_neon;
+        c->put_h264_qpel_pixels_tab[0][ 8] = ff_put_h264_qpel16_mc02_neon;
+        c->put_h264_qpel_pixels_tab[0][ 9] = ff_put_h264_qpel16_mc12_neon;
+        c->put_h264_qpel_pixels_tab[0][10] = ff_put_h264_qpel16_mc22_neon;
+        c->put_h264_qpel_pixels_tab[0][11] = ff_put_h264_qpel16_mc32_neon;
+        c->put_h264_qpel_pixels_tab[0][12] = ff_put_h264_qpel16_mc03_neon;
+        c->put_h264_qpel_pixels_tab[0][13] = ff_put_h264_qpel16_mc13_neon;
+        c->put_h264_qpel_pixels_tab[0][14] = ff_put_h264_qpel16_mc23_neon;
+        c->put_h264_qpel_pixels_tab[0][15] = ff_put_h264_qpel16_mc33_neon;
+
+        c->put_h264_qpel_pixels_tab[1][ 0] = ff_put_h264_qpel8_mc00_neon;
+        c->put_h264_qpel_pixels_tab[1][ 1] = ff_put_h264_qpel8_mc10_neon;
+        c->put_h264_qpel_pixels_tab[1][ 2] = ff_put_h264_qpel8_mc20_neon;
+        c->put_h264_qpel_pixels_tab[1][ 3] = ff_put_h264_qpel8_mc30_neon;
+        c->put_h264_qpel_pixels_tab[1][ 4] = ff_put_h264_qpel8_mc01_neon;
+        c->put_h264_qpel_pixels_tab[1][ 5] = ff_put_h264_qpel8_mc11_neon;
+        c->put_h264_qpel_pixels_tab[1][ 6] = ff_put_h264_qpel8_mc21_neon;
+        c->put_h264_qpel_pixels_tab[1][ 7] = ff_put_h264_qpel8_mc31_neon;
+        c->put_h264_qpel_pixels_tab[1][ 8] = ff_put_h264_qpel8_mc02_neon;
+        c->put_h264_qpel_pixels_tab[1][ 9] = ff_put_h264_qpel8_mc12_neon;
+        c->put_h264_qpel_pixels_tab[1][10] = ff_put_h264_qpel8_mc22_neon;
+        c->put_h264_qpel_pixels_tab[1][11] = ff_put_h264_qpel8_mc32_neon;
+        c->put_h264_qpel_pixels_tab[1][12] = ff_put_h264_qpel8_mc03_neon;
+        c->put_h264_qpel_pixels_tab[1][13] = ff_put_h264_qpel8_mc13_neon;
+        c->put_h264_qpel_pixels_tab[1][14] = ff_put_h264_qpel8_mc23_neon;
+        c->put_h264_qpel_pixels_tab[1][15] = ff_put_h264_qpel8_mc33_neon;
+
+        c->avg_h264_qpel_pixels_tab[0][ 0] = ff_avg_h264_qpel16_mc00_neon;
+        c->avg_h264_qpel_pixels_tab[0][ 1] = ff_avg_h264_qpel16_mc10_neon;
+        c->avg_h264_qpel_pixels_tab[0][ 2] = ff_avg_h264_qpel16_mc20_neon;
+        c->avg_h264_qpel_pixels_tab[0][ 3] = ff_avg_h264_qpel16_mc30_neon;
+        c->avg_h264_qpel_pixels_tab[0][ 4] = ff_avg_h264_qpel16_mc01_neon;
+        c->avg_h264_qpel_pixels_tab[0][ 5] = ff_avg_h264_qpel16_mc11_neon;
+        c->avg_h264_qpel_pixels_tab[0][ 6] = ff_avg_h264_qpel16_mc21_neon;
+        c->avg_h264_qpel_pixels_tab[0][ 7] = ff_avg_h264_qpel16_mc31_neon;
+        c->avg_h264_qpel_pixels_tab[0][ 8] = ff_avg_h264_qpel16_mc02_neon;
+        c->avg_h264_qpel_pixels_tab[0][ 9] = ff_avg_h264_qpel16_mc12_neon;
+        c->avg_h264_qpel_pixels_tab[0][10] = ff_avg_h264_qpel16_mc22_neon;
+        c->avg_h264_qpel_pixels_tab[0][11] = ff_avg_h264_qpel16_mc32_neon;
+        c->avg_h264_qpel_pixels_tab[0][12] = ff_avg_h264_qpel16_mc03_neon;
+        c->avg_h264_qpel_pixels_tab[0][13] = ff_avg_h264_qpel16_mc13_neon;
+        c->avg_h264_qpel_pixels_tab[0][14] = ff_avg_h264_qpel16_mc23_neon;
+        c->avg_h264_qpel_pixels_tab[0][15] = ff_avg_h264_qpel16_mc33_neon;
+
+        c->avg_h264_qpel_pixels_tab[1][ 0] = ff_avg_h264_qpel8_mc00_neon;
+        c->avg_h264_qpel_pixels_tab[1][ 1] = ff_avg_h264_qpel8_mc10_neon;
+        c->avg_h264_qpel_pixels_tab[1][ 2] = ff_avg_h264_qpel8_mc20_neon;
+        c->avg_h264_qpel_pixels_tab[1][ 3] = ff_avg_h264_qpel8_mc30_neon;
+        c->avg_h264_qpel_pixels_tab[1][ 4] = ff_avg_h264_qpel8_mc01_neon;
+        c->avg_h264_qpel_pixels_tab[1][ 5] = ff_avg_h264_qpel8_mc11_neon;
+        c->avg_h264_qpel_pixels_tab[1][ 6] = ff_avg_h264_qpel8_mc21_neon;
+        c->avg_h264_qpel_pixels_tab[1][ 7] = ff_avg_h264_qpel8_mc31_neon;
+        c->avg_h264_qpel_pixels_tab[1][ 8] = ff_avg_h264_qpel8_mc02_neon;
+        c->avg_h264_qpel_pixels_tab[1][ 9] = ff_avg_h264_qpel8_mc12_neon;
+        c->avg_h264_qpel_pixels_tab[1][10] = ff_avg_h264_qpel8_mc22_neon;
+        c->avg_h264_qpel_pixels_tab[1][11] = ff_avg_h264_qpel8_mc32_neon;
+        c->avg_h264_qpel_pixels_tab[1][12] = ff_avg_h264_qpel8_mc03_neon;
+        c->avg_h264_qpel_pixels_tab[1][13] = ff_avg_h264_qpel8_mc13_neon;
+        c->avg_h264_qpel_pixels_tab[1][14] = ff_avg_h264_qpel8_mc23_neon;
+        c->avg_h264_qpel_pixels_tab[1][15] = ff_avg_h264_qpel8_mc33_neon;
+    }
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aarch64/hpeldsp_init_aarch64.c

@@ -0,0 +1,123 @@
+/*
+ * ARM NEON optimised DSP functions
+ * Copyright (c) 2008 Mans Rullgard <mans@mansr.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "config.h"
+
+#include "libavutil/attributes.h"
+#include "libavutil/cpu.h"
+#include "libavutil/aarch64/cpu.h"
+#include "libavcodec/hpeldsp.h"
+
+void     ff_put_pixels16_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+void  ff_put_pixels16_x2_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+void  ff_put_pixels16_y2_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+void ff_put_pixels16_xy2_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+void      ff_put_pixels8_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+void   ff_put_pixels8_x2_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+void   ff_put_pixels8_y2_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+void  ff_put_pixels8_xy2_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+
+void  ff_put_pixels16_x2_no_rnd_neon(uint8_t *block, const uint8_t *pixels,
+                                     ptrdiff_t line_size, int h);
+void  ff_put_pixels16_y2_no_rnd_neon(uint8_t *block, const uint8_t *pixels,
+                                     ptrdiff_t line_size, int h);
+void ff_put_pixels16_xy2_no_rnd_neon(uint8_t *block, const uint8_t *pixels,
+                                     ptrdiff_t line_size, int h);
+void   ff_put_pixels8_x2_no_rnd_neon(uint8_t *block, const uint8_t *pixels,
+                                     ptrdiff_t line_size, int h);
+void   ff_put_pixels8_y2_no_rnd_neon(uint8_t *block, const uint8_t *pixels,
+                                     ptrdiff_t line_size, int h);
+void  ff_put_pixels8_xy2_no_rnd_neon(uint8_t *block, const uint8_t *pixels,
+                                     ptrdiff_t line_size, int h);
+
+void     ff_avg_pixels16_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+void  ff_avg_pixels16_x2_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+void  ff_avg_pixels16_y2_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+void ff_avg_pixels16_xy2_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+void      ff_avg_pixels8_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+void   ff_avg_pixels8_x2_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+void   ff_avg_pixels8_y2_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+void  ff_avg_pixels8_xy2_neon(uint8_t *block, const uint8_t *pixels,
+                              ptrdiff_t line_size, int h);
+
+void  ff_avg_pixels16_x2_no_rnd_neon(uint8_t *block, const uint8_t *pixels,
+                                     ptrdiff_t line_size, int h);
+void  ff_avg_pixels16_y2_no_rnd_neon(uint8_t *block, const uint8_t *pixels,
+                                     ptrdiff_t line_size, int h);
+void ff_avg_pixels16_xy2_no_rnd_neon(uint8_t *block, const uint8_t *pixels,
+                                     ptrdiff_t line_size, int h);
+
+av_cold void ff_hpeldsp_init_aarch64(HpelDSPContext *c, int flags)
+{
+    int cpu_flags = av_get_cpu_flags();
+
+    if (have_neon(cpu_flags)) {
+        c->put_pixels_tab[0][0] = ff_put_pixels16_neon;
+        c->put_pixels_tab[0][1] = ff_put_pixels16_x2_neon;
+        c->put_pixels_tab[0][2] = ff_put_pixels16_y2_neon;
+        c->put_pixels_tab[0][3] = ff_put_pixels16_xy2_neon;
+        c->put_pixels_tab[1][0] = ff_put_pixels8_neon;
+        c->put_pixels_tab[1][1] = ff_put_pixels8_x2_neon;
+        c->put_pixels_tab[1][2] = ff_put_pixels8_y2_neon;
+        c->put_pixels_tab[1][3] = ff_put_pixels8_xy2_neon;
+
+        c->put_no_rnd_pixels_tab[0][0] = ff_put_pixels16_neon;
+        c->put_no_rnd_pixels_tab[0][1] = ff_put_pixels16_x2_no_rnd_neon;
+        c->put_no_rnd_pixels_tab[0][2] = ff_put_pixels16_y2_no_rnd_neon;
+        c->put_no_rnd_pixels_tab[0][3] = ff_put_pixels16_xy2_no_rnd_neon;
+        c->put_no_rnd_pixels_tab[1][0] = ff_put_pixels8_neon;
+        c->put_no_rnd_pixels_tab[1][1] = ff_put_pixels8_x2_no_rnd_neon;
+        c->put_no_rnd_pixels_tab[1][2] = ff_put_pixels8_y2_no_rnd_neon;
+        c->put_no_rnd_pixels_tab[1][3] = ff_put_pixels8_xy2_no_rnd_neon;
+
+        c->avg_pixels_tab[0][0] = ff_avg_pixels16_neon;
+        c->avg_pixels_tab[0][1] = ff_avg_pixels16_x2_neon;
+        c->avg_pixels_tab[0][2] = ff_avg_pixels16_y2_neon;
+        c->avg_pixels_tab[0][3] = ff_avg_pixels16_xy2_neon;
+        c->avg_pixels_tab[1][0] = ff_avg_pixels8_neon;
+        c->avg_pixels_tab[1][1] = ff_avg_pixels8_x2_neon;
+        c->avg_pixels_tab[1][2] = ff_avg_pixels8_y2_neon;
+        c->avg_pixels_tab[1][3] = ff_avg_pixels8_xy2_neon;
+
+        c->avg_no_rnd_pixels_tab[0] = ff_avg_pixels16_neon;
+        c->avg_no_rnd_pixels_tab[1] = ff_avg_pixels16_x2_no_rnd_neon;
+        c->avg_no_rnd_pixels_tab[2] = ff_avg_pixels16_y2_no_rnd_neon;
+        c->avg_no_rnd_pixels_tab[3] = ff_avg_pixels16_xy2_no_rnd_neon;
+    }
+}

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aarch64/idct.h

@@ -0,0 +1,28 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AARCH64_IDCT_H
+#define AVCODEC_AARCH64_IDCT_H
+
+#include <stdint.h>
+
+void ff_simple_idct_neon(int16_t *data);
+void ff_simple_idct_put_neon(uint8_t *dest, ptrdiff_t line_size, int16_t *data);
+void ff_simple_idct_add_neon(uint8_t *dest, ptrdiff_t line_size, int16_t *data);
+
+#endif /* AVCODEC_AARCH64_IDCT_H */

trunk/3rdparty/ffmpeg-4-fit/libavcodec/aarch64/idctdsp_init_aarch64.c

@@ -0,0 +1,41 @@
+/*
+ * ARM-NEON-optimized IDCT functions
+ * Copyright (c) 2008 Mans Rullgard <mans@mansr.com>
+ * Copyright (c) 2017 Matthieu Bouron <matthieu.bouron@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/attributes.h"
+#include "libavcodec/avcodec.h"
+#include "libavcodec/idctdsp.h"
+#include "idct.h"
+
+av_cold void ff_idctdsp_init_aarch64(IDCTDSPContext *c, AVCodecContext *avctx,
+                                     unsigned high_bit_depth)
+{
+    if (!avctx->lowres && !high_bit_depth) {
+        if (avctx->idct_algo == FF_IDCT_AUTO ||
+            avctx->idct_algo == FF_IDCT_SIMPLEAUTO ||
+            avctx->idct_algo == FF_IDCT_SIMPLENEON) {
+            c->idct_put  = ff_simple_idct_put_neon;
+            c->idct_add  = ff_simple_idct_add_neon;
+            c->idct      = ff_simple_idct_neon;
+            c->perm_type = FF_IDCT_PERM_PARTTRANS;
+        }
+    }
+}