external/srs
1
0
Fork 0
mirror of https://github.com/ossrs/srs.git synced 2025-03-09 15:49:59 +00:00
Code Issues Releases Wiki Activity

fix #212, support publish audio raw frames. 2.0.27

Browse source
This commit is contained in:
winlin 2014-11-20 13:36:14 +08:00
parent 856cc0a51d
commit a058eeeb20
7 changed files with 301 additions and 4 deletions
Download patch file Download diff file Expand all files Collapse all files

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,