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200 lines
6 KiB
C
200 lines
6 KiB
C
/*
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* Opus decoder/demuxer common functions
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* Copyright (c) 2012 Andrew D'Addesio
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* Copyright (c) 2013-2014 Mozilla Corporation
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#ifndef AVCODEC_OPUS_H
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#define AVCODEC_OPUS_H
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#include <stdint.h>
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#include "libavutil/audio_fifo.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/frame.h"
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#include "libswresample/swresample.h"
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#include "avcodec.h"
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#include "opus_rc.h"
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#define MAX_FRAME_SIZE 1275
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#define MAX_FRAMES 48
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#define MAX_PACKET_DUR 5760
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#define CELT_SHORT_BLOCKSIZE 120
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#define CELT_OVERLAP CELT_SHORT_BLOCKSIZE
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#define CELT_MAX_LOG_BLOCKS 3
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#define CELT_MAX_FRAME_SIZE (CELT_SHORT_BLOCKSIZE * (1 << CELT_MAX_LOG_BLOCKS))
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#define CELT_MAX_BANDS 21
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#define SILK_HISTORY 322
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#define SILK_MAX_LPC 16
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#define ROUND_MULL(a,b,s) (((MUL64(a, b) >> ((s) - 1)) + 1) >> 1)
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#define ROUND_MUL16(a,b) ((MUL16(a, b) + 16384) >> 15)
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#define OPUS_TS_HEADER 0x7FE0 // 0x3ff (11 bits)
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#define OPUS_TS_MASK 0xFFE0 // top 11 bits
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static const uint8_t opus_default_extradata[30] = {
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'O', 'p', 'u', 's', 'H', 'e', 'a', 'd',
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1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
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0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
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};
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enum OpusMode {
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OPUS_MODE_SILK,
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OPUS_MODE_HYBRID,
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OPUS_MODE_CELT,
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OPUS_MODE_NB
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};
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enum OpusBandwidth {
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OPUS_BANDWIDTH_NARROWBAND,
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OPUS_BANDWIDTH_MEDIUMBAND,
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OPUS_BANDWIDTH_WIDEBAND,
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OPUS_BANDWIDTH_SUPERWIDEBAND,
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OPUS_BANDWIDTH_FULLBAND,
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OPUS_BANDWITH_NB
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};
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typedef struct SilkContext SilkContext;
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typedef struct CeltFrame CeltFrame;
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typedef struct OpusPacket {
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int packet_size; /**< packet size */
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int data_size; /**< size of the useful data -- packet size - padding */
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int code; /**< packet code: specifies the frame layout */
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int stereo; /**< whether this packet is mono or stereo */
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int vbr; /**< vbr flag */
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int config; /**< configuration: tells the audio mode,
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** bandwidth, and frame duration */
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int frame_count; /**< frame count */
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int frame_offset[MAX_FRAMES]; /**< frame offsets */
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int frame_size[MAX_FRAMES]; /**< frame sizes */
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int frame_duration; /**< frame duration, in samples @ 48kHz */
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enum OpusMode mode; /**< mode */
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enum OpusBandwidth bandwidth; /**< bandwidth */
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} OpusPacket;
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typedef struct OpusStreamContext {
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AVCodecContext *avctx;
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int output_channels;
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OpusRangeCoder rc;
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OpusRangeCoder redundancy_rc;
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SilkContext *silk;
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CeltFrame *celt;
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AVFloatDSPContext *fdsp;
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float silk_buf[2][960];
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float *silk_output[2];
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DECLARE_ALIGNED(32, float, celt_buf)[2][960];
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float *celt_output[2];
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DECLARE_ALIGNED(32, float, redundancy_buf)[2][960];
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float *redundancy_output[2];
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/* data buffers for the final output data */
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float *out[2];
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int out_size;
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float *out_dummy;
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int out_dummy_allocated_size;
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SwrContext *swr;
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AVAudioFifo *celt_delay;
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int silk_samplerate;
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/* number of samples we still want to get from the resampler */
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int delayed_samples;
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OpusPacket packet;
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int redundancy_idx;
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} OpusStreamContext;
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// a mapping between an opus stream and an output channel
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typedef struct ChannelMap {
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int stream_idx;
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int channel_idx;
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// when a single decoded channel is mapped to multiple output channels, we
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// write to the first output directly and copy from it to the others
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// this field is set to 1 for those copied output channels
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int copy;
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// this is the index of the output channel to copy from
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int copy_idx;
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// this channel is silent
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int silence;
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} ChannelMap;
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typedef struct OpusContext {
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AVClass *av_class;
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OpusStreamContext *streams;
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int apply_phase_inv;
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/* current output buffers for each streams */
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float **out;
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int *out_size;
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/* Buffers for synchronizing the streams when they have different
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* resampling delays */
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AVAudioFifo **sync_buffers;
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/* number of decoded samples for each stream */
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int *decoded_samples;
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int nb_streams;
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int nb_stereo_streams;
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AVFloatDSPContext *fdsp;
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int16_t gain_i;
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float gain;
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ChannelMap *channel_maps;
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} OpusContext;
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int ff_opus_parse_packet(OpusPacket *pkt, const uint8_t *buf, int buf_size,
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int self_delimited);
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int ff_opus_parse_extradata(AVCodecContext *avctx, OpusContext *s);
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int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels);
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void ff_silk_free(SilkContext **ps);
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void ff_silk_flush(SilkContext *s);
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/**
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* Decode the LP layer of one Opus frame (which may correspond to several SILK
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* frames).
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*/
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int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc,
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float *output[2],
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enum OpusBandwidth bandwidth, int coded_channels,
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int duration_ms);
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/* Encode or decode CELT bands */
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void ff_celt_quant_bands(CeltFrame *f, OpusRangeCoder *rc);
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/* Encode or decode CELT bitallocation */
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void ff_celt_bitalloc(CeltFrame *f, OpusRangeCoder *rc, int encode);
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#endif /* AVCODEC_OPUS_H */
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