external/srs
1
0
Fork 0
mirror of https://github.com/ossrs/srs.git synced 2025-02-13 03:41:55 +00:00
Code Issues Releases Wiki Activity

Merge branch 'srs.master'

Browse source
This commit is contained in:
winlin 2014-11-20 13:36:51 +08:00
commit 0ce0df3fb4
7 changed files with 301 additions and 4 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

1
README.md
View file

@ -482,6 +482,7 @@ Supported operating systems and hardware:
* 2013-10-17, Created.<br/>
## History
* v2.0, 2014-11-20, fix [#212](https://github.com/winlinvip/simple-rtmp-server/issues/212), support publish audio raw frames. 2.0.27
* v2.0, 2014-11-19, fix [#213](https://github.com/winlinvip/simple-rtmp-server/issues/213), support compile [srs-librtmp on windows](https://github.com/winlinvip/srs.librtmp), [bug #213](https://github.com/winlinvip/simple-rtmp-server/issues/213). 2.0.26
* v2.0, 2014-11-18, all wiki translated to English. 2.0.23.
* v2.0, 2014-11-15, fix [#204](https://github.com/winlinvip/simple-rtmp-server/issues/204), srs-librtmp drop duplicated sps/pps(sequence header). 2.0.22.

7
trunk/research/librtmp/Makefile
View file

@ -6,7 +6,8 @@ else
ST_ALL = objs/srs_flv_parser \
objs/srs_flv_injecter objs/srs_publish objs/srs_play \
objs/srs_ingest_flv objs/srs_ingest_rtmp objs/srs_detect_rtmp \
objs/srs_bandwidth_check objs/srs_h264_raw_publish
objs/srs_bandwidth_check objs/srs_h264_raw_publish \
objs/srs_audio_raw_publish
endif
.PHONY: default clean help ssl nossl
@ -24,6 +25,7 @@ help:
@echo " srs_flv_injecter inject keyframes information to metadata."
@echo " srs_publish publish program using srs-librtmp"
@echo " srs_h264_raw_publish publish raw h.264 stream to SSR by srs-librtmp"
@echo " srs_audio_raw_publish publish raw audio stream to SSR by srs-librtmp"
@echo " srs_play play program using srs-librtmp"
@echo " srs_ingest_flv ingest flv file and publish to RTMP server."
@echo " srs_ingest_rtmp ingest RTMP and publish to RTMP server."
@ -85,6 +87,9 @@ objs/srs_publish: srs_publish.c $(SRS_RESEARCH_DEPS) $(SRS_LIBRTMP_I) $(SRS_LIBR
objs/srs_h264_raw_publish: srs_h264_raw_publish.c $(SRS_RESEARCH_DEPS) $(SRS_LIBRTMP_I) $(SRS_LIBRTMP_L) $(SRS_LIBSSL_L)
$(GCC) srs_h264_raw_publish.c $(SRS_LIBRTMP_L) $(SRS_LIBSSL_L) $(EXTRA_CXX_FLAG) -o objs/srs_h264_raw_publish
objs/srs_audio_raw_publish: srs_audio_raw_publish.c $(SRS_RESEARCH_DEPS) $(SRS_LIBRTMP_I) $(SRS_LIBRTMP_L) $(SRS_LIBSSL_L)
$(GCC) srs_audio_raw_publish.c $(SRS_LIBRTMP_L) $(SRS_LIBSSL_L) $(EXTRA_CXX_FLAG) -o objs/srs_audio_raw_publish
objs/srs_play: srs_play.c $(SRS_RESEARCH_DEPS) $(SRS_LIBRTMP_I) $(SRS_LIBRTMP_L) $(SRS_LIBSSL_L)
$(GCC) srs_play.c $(SRS_LIBRTMP_L) $(SRS_LIBSSL_L) $(EXTRA_CXX_FLAG) -o objs/srs_play

190
trunk/research/librtmp/srs_audio_raw_publish.c Normal file
View file

@ -0,0 +1,190 @@
/*
The MIT License (MIT)
Copyright (c) 2013-2014 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
the Software without restriction, including without limitation the rights to
use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
the Software, and to permit persons to whom the Software is furnished to do so,
subject to the following conditions:
The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
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.
*/
/**
gcc srs_audio_raw_publish.c ../../objs/lib/srs_librtmp.a -g -O0 -lstdc++ -o srs_audio_raw_publish
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
// for open audio raw file.
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include "../../objs/include/srs_librtmp.h"
// https://github.com/winlinvip/simple-rtmp-server/issues/212#issuecomment-63648892
// allspace:
// Take this file as an example: https://github.com/allspace/files/blob/master/srs.pcm
// It's captured using SDK callback method. I have filtered out h264 video, so it's audio only now.
// For every frame, it's a 8 bytes vendor specific header, following 160 bytes audio frame.
// The header part can be ignored.
int read_audio_frame(char* audio_raw, int file_size, char** pp, char** pdata, int* psize)
{
char* p = *pp;
if (file_size - (p - audio_raw) < 168) {
srs_lib_trace("audio must be 160+8 bytes. left %d bytes.",
file_size - (p - audio_raw));
return - 1;
}
// ignore 8bytes vendor specific header.
p += 8;
// 160 bytes audio frame
*pdata = p;
*psize = 160;
// next frame.
*pp = p + *psize;
return 0;
}
int main(int argc, char** argv)
{
printf("publish raw audio as rtmp stream to server like FMLE/FFMPEG/Encoder\n");
printf("SRS(simple-rtmp-server) client librtmp library.\n");
printf("version: %d.%d.%d\n", srs_version_major(), srs_version_minor(), srs_version_revision());
if (argc <= 2) {
printf("Usage: %s <audio_raw_file> <rtmp_publish_url>\n", argv[0]);
printf(" audio_raw_file: the audio raw steam file.\n");
printf(" rtmp_publish_url: the rtmp publish url.\n");
printf("For example:\n");
printf(" %s ./audio.raw.pcm rtmp://127.0.0.1:1935/live/livestream\n", argv[0]);
printf("Where the file: http://winlinvip.github.io/srs.release/3rdparty/audio.raw.pcm\n");
printf("See: https://github.com/winlinvip/simple-rtmp-server/issues/212\n");
exit(-1);
}
const char* raw_file = argv[1];
const char* rtmp_url = argv[2];
srs_lib_trace("raw_file=%s, rtmp_url=%s", raw_file, rtmp_url);
// open file
int raw_fd = open(raw_file, O_RDONLY);
if (raw_fd < 0) {
srs_lib_trace("open audio raw file %s failed.", raw_fd);
goto rtmp_destroy;
}
off_t file_size = lseek(raw_fd, 0, SEEK_END);
if (file_size <= 0) {
srs_lib_trace("audio raw file %s empty.", raw_file);
goto rtmp_destroy;
}
srs_lib_trace("read entirely audio raw file, size=%dKB", (int)(file_size / 1024));
char* audio_raw = (char*)malloc(file_size);
if (!audio_raw) {
srs_lib_trace("alloc raw buffer failed for file %s.", raw_file);
goto rtmp_destroy;
}
lseek(raw_fd, 0, SEEK_SET);
ssize_t nb_read = 0;
if ((nb_read = read(raw_fd, audio_raw, file_size)) != file_size) {
srs_lib_trace("buffer %s failed, expect=%dKB, actual=%dKB.",
raw_file, (int)(file_size / 1024), (int)(nb_read / 1024));
goto rtmp_destroy;
}
// connect rtmp context
srs_rtmp_t rtmp = srs_rtmp_create(rtmp_url);
if (srs_simple_handshake(rtmp) != 0) {
srs_lib_trace("simple handshake failed.");
goto rtmp_destroy;
}
srs_lib_trace("simple handshake success");
if (srs_connect_app(rtmp) != 0) {
srs_lib_trace("connect vhost/app failed.");
goto rtmp_destroy;
}
srs_lib_trace("connect vhost/app success");
if (srs_publish_stream(rtmp) != 0) {
srs_lib_trace("publish stream failed.");
goto rtmp_destroy;
}
srs_lib_trace("publish stream success");
u_int32_t timestamp = 0;
u_int32_t time_delta = 17;
// @remark, to decode the file.
char* p = audio_raw;
for (;p < audio_raw + file_size;) {
// @remark, read a frame from file buffer.
char* data = NULL;
int size = 0;
if (read_audio_frame(audio_raw, file_size, &p, &data, &size) < 0) {
srs_lib_trace("read a frame from file buffer failed.");
goto rtmp_destroy;
}
// 0 = Linear PCM, platform endian
// 1 = ADPCM
// 2 = MP3
// 7 = G.711 A-law logarithmic PCM
// 8 = G.711 mu-law logarithmic PCM
// 10 = AAC
// 11 = Speex
char sound_format = 1;
// 3 = 44 kHz
char sound_rate = 3;
// 1 = 16-bit samples
char sound_size = 1;
// 1 = Stereo sound
char sound_type = 1;
timestamp += time_delta;
if (srs_audio_write_raw_frame(rtmp,
sound_format, sound_rate, sound_size, sound_type,
0, data, size, timestamp) != 0
) {
srs_lib_trace("send audio raw data failed.");
goto rtmp_destroy;
}
srs_lib_trace("sent packet: type=%s, time=%d, size=%d, codec=%d, rate=%d, sample=%d, channel=%d",
srs_type2string(SRS_RTMP_TYPE_AUDIO), timestamp, size, sound_format, sound_rate, sound_size,
sound_type);
// @remark, when use encode device, it not need to sleep.
usleep(1000 * time_delta);
}
rtmp_destroy:
srs_rtmp_destroy(rtmp);
close(raw_fd);
free(audio_raw);
return 0;
}

2
trunk/src/core/srs_core.hpp
View file

@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
// current release version
#define VERSION_MAJOR 2
#define VERSION_MINOR 0
#define VERSION_REVISION 26
#define VERSION_REVISION 27
// server info.
#define RTMP_SIG_SRS_KEY "SRS"
#define RTMP_SIG_SRS_ROLE "origin/edge server"

44
trunk/src/libs/srs_librtmp.cpp
View file

@ -1438,6 +1438,46 @@ char* srs_amf0_human_print(srs_amf0_t amf0, char** pdata, int* psize)
return any->human_print(pdata, psize);
}
/**
* write audio raw frame to SRS.
*/
int srs_audio_write_raw_frame(srs_rtmp_t rtmp,
char sound_format, char sound_rate, char sound_size, char sound_type,
char aac_packet_type, char* frame, int frame_size, u_int32_t timestamp
) {
int ret = ERROR_SUCCESS;
Context* context = (Context*)rtmp;
srs_assert(context);
// TODO: FIXME: for aac, must send the sequence header first.
// for audio frame, there is 1 or 2 bytes header:
// 1bytes, SoundFormat|SoundRate|SoundSize|SoundType
// 1bytes, AACPacketType for SoundFormat == 10
int size = frame_size + 1;
if (aac_packet_type == SrsCodecAudioAAC) {
size += 1;
}
char* data = new char[size];
char* p = data;
u_int8_t audio_header = sound_type & 0x01;
audio_header |= (sound_size << 1) & 0x02;
audio_header |= (sound_rate << 2) & 0x0c;
audio_header |= (sound_format << 4) & 0xf0;
*p++ = audio_header;
if (aac_packet_type == SrsCodecAudioAAC) {
*p++ = aac_packet_type;
}
memcpy(p, frame, frame_size);
return srs_write_packet(context, SRS_RTMP_TYPE_AUDIO, timestamp, data, size);
}
/**
* write h264 packet, with rtmp header.
* @param frame_type, SrsCodecVideoAVCFrameKeyFrame or SrsCodecVideoAVCFrameInterFrame.
@ -1458,7 +1498,6 @@ int __srs_write_h264_packet(Context* context,
// @see: E.4.3 Video Tags, video_file_format_spec_v10_1.pdf, page 78
int size = h264_raw_size + 5;
char* data = new char[size];
memcpy(data + 5, h264_raw_data, h264_raw_size);
char* p = data;
// @see: E.4.3 Video Tags, video_file_format_spec_v10_1.pdf, page 78
@ -1480,6 +1519,9 @@ int __srs_write_h264_packet(Context* context,
*p++ = pp[1];
*p++ = pp[0];
// h.264 raw data.
memcpy(p, h264_raw_data, h264_raw_size);
return srs_write_packet(context, SRS_RTMP_TYPE_VIDEO, timestamp, data, size);
}

60
trunk/src/libs/srs_librtmp.hpp
View file

@ -461,6 +461,64 @@ extern void srs_amf0_strict_array_append(srs_amf0_t amf0, srs_amf0_t value);
*/
extern char* srs_amf0_human_print(srs_amf0_t amf0, char** pdata, int* psize);
/*************************************************************
**************************************************************
* audio raw codec
**************************************************************
*************************************************************/
/**
* write an audio raw frame to srs.
* not similar to h.264 video, the audio never aggregated, always
* encoded one frame by one, so this api is used to write a frame.
*
* @param sound_format Format of SoundData. The following values are defined:
* 0 = Linear PCM, platform endian
* 1 = ADPCM
* 2 = MP3
* 3 = Linear PCM, little endian
* 4 = Nellymoser 16 kHz mono
* 5 = Nellymoser 8 kHz mono
* 6 = Nellymoser
* 7 = G.711 A-law logarithmic PCM
* 8 = G.711 mu-law logarithmic PCM
* 9 = reserved
* 10 = AAC
* 11 = Speex
* 14 = MP3 8 kHz
* 15 = Device-specific sound
* Formats 7, 8, 14, and 15 are reserved.
* AAC is supported in Flash Player 9,0,115,0 and higher.
* Speex is supported in Flash Player 10 and higher.
* @param sound_rate Sampling rate. The following values are defined:
* 0 = 5.5 kHz
* 1 = 11 kHz
* 2 = 22 kHz
* 3 = 44 kHz
* @param sound_size Size of each audio sample. This parameter only pertains to
* uncompressed formats. Compressed formats always decode
* to 16 bits internally.
* 0 = 8-bit samples
* 1 = 16-bit samples
* @param sound_type Mono or stereo sound
* 0 = Mono sound
* 1 = Stereo sound
* @param aac_packet_type The following values are defined:
* 0 = AAC sequence header
* 1 = AAC raw
* @param timestamp The timestamp of audio.
*
* @remark Ignore aac_packet_type if not aac(sound_format!=10).
*
* @see https://github.com/winlinvip/simple-rtmp-server/issues/212
* @see E.4.2.1 AUDIODATA of video_file_format_spec_v10_1.pdf
*
* @return 0, success; otherswise, failed.
*/
extern int srs_audio_write_raw_frame(srs_rtmp_t rtmp,
char sound_format, char sound_rate, char sound_size, char sound_type,
char aac_packet_type, char* frame, int frame_size, u_int32_t timestamp
);
/*************************************************************
**************************************************************
* h264 raw codec
@ -474,7 +532,7 @@ typedef int srs_h264_bool;
* each frame prefixed h.264 annexb header, by N[00] 00 00 01, where N>=0,
* for instance, frame = header(00 00 00 01) + payload(67 42 80 29 95 A0 14 01 6E 40)
* about annexb, @see H.264-AVC-ISO_IEC_14496-10.pdf, page 211.
* @paam frames_size the size of h264 raw data.
* @param frames_size the size of h264 raw data.
* assert frames_size > 0, at least has 1 bytes header.
* @param dts the dts of h.264 raw data.
* @param pts the pts of h.264 raw data.

1
trunk/src/srs/srs.upp
View file

@ -128,6 +128,7 @@ file
..\utest\srs_utest_reload.hpp,
..\utest\srs_utest_reload.cpp,
research readonly separator,
..\..\research\librtmp\srs_audio_raw_publish.c,
..\..\research\librtmp\srs_bandwidth_check.c,
..\..\research\librtmp\srs_detect_rtmp.c,
..\..\research\librtmp\srs_flv_injecter.c,