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support pure video hls.
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commit
597993d414
6 changed files with 61 additions and 9 deletions
29
README.md
29
README.md
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@ -346,6 +346,7 @@ Remark:
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* v3.0, 2015-03-15, fork srs2 and start srs3. 3.0.0
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* v3.0, 2015-03-15, fork srs2 and start srs3. 3.0.0
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### SRS 2.0 history
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### SRS 2.0 history
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* v2.0, 2015-05-29, fix [#409](https://github.com/simple-rtmp-server/srs/issues/409) support pure video hls. 2.0.172.
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* v2.0, 2015-05-28, support [srs-dolphin][srs-dolphin], the multiple-process SRS.
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* v2.0, 2015-05-28, support [srs-dolphin][srs-dolphin], the multiple-process SRS.
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* v2.0, 2015-05-24, fix [#404](https://github.com/simple-rtmp-server/srs/issues/404) register handler then start http thread. 2.0.167.
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* v2.0, 2015-05-24, fix [#404](https://github.com/simple-rtmp-server/srs/issues/404) register handler then start http thread. 2.0.167.
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* v2.0, 2015-05-23, refine the thread, protocol, kbps code. 2.0.166
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* v2.0, 2015-05-23, refine the thread, protocol, kbps code. 2.0.166
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@ -735,7 +736,7 @@ The latency between encoder and player with realtime config(
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[CN](https://github.com/simple-rtmp-server/srs/wiki/v2_CN_LowLatency),
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[CN](https://github.com/simple-rtmp-server/srs/wiki/v2_CN_LowLatency),
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[EN](https://github.com/simple-rtmp-server/srs/wiki/v2_EN_LowLatency)
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[EN](https://github.com/simple-rtmp-server/srs/wiki/v2_EN_LowLatency)
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):
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):
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| Update | SRS | VP6 | H.264 | VP6+MP3 | H.264+MP3 |
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| Update | SRS | VP6 | H.264 | VP6+MP3 | H.264+MP3 |
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| ------------- | --------- | --------- | --------- | --------- | -------- |
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| ------------- | --------- | --------- | --------- | --------- | -------- |
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@ -747,6 +748,32 @@ We use FMLE as encoder for benchmark. The latency of server is 0.1s+,
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and the bottleneck is the encoder. For more information, read
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and the bottleneck is the encoder. For more information, read
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[bug #257](https://github.com/simple-rtmp-server/srs/issues/257#issuecomment-66864413).
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[bug #257](https://github.com/simple-rtmp-server/srs/issues/257#issuecomment-66864413).
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### HLS overhead
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About the HLS overhead of SRS, we compare the overhead to FLV by remux the HLS to FLV by ffmpeg.
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| Bitrate | Duration | FLV(KB) | HLS(KB) | Overhead |
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| ------- | -------- | ------- | -------- | --------- |
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| 275kbps | 600s | 11144 | 12756 | 14.46% |
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| 260kbps | 1860s | 59344 | 68004 | 14.59% |
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| 697kbps | 60s | 5116 | 5476 | 7.03% |
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| 565kbps | 453s | 31316 | 33544 | 7.11% |
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| 565kbps | 1813s | 125224 | 134140 | 7.12% |
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| 861kbps | 497s | 52316 | 54924 | 4.98% |
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| 857kbps | 1862s | 195008 | 204768 | 5.00% |
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| 1301kbps | 505s | 80320 | 83676 | 4.17% |
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| 1312kbps | 1915s | 306920 | 319680 | 4.15% |
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| 2707kbps | 600s | 198356 | 204560 | 3.12% |
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| 2814kbps | 1800s | 618456 | 637660 | 3.10% |
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| 2828kbps | 60s | 20716 | 21356 | 3.08% |
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| 2599kbps | 307s | 97580 | 100672 | 3.16% |
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| 2640kbps | 1283s | 413880 | 426912 | 3.14% |
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| 5254kbps | 71s | 45832 | 47056 | 2.67% |
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| 5147kbps | 370s | 195040 | 200280 | 2.68% |
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| 5158kbps | 1327s | 835664 | 858092 | 2.68% |
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The HLS overhead is calc by: (HLS - FLV) / FLV * 100%
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## Architecture
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## Architecture
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SRS always use the most simple architecture to support complex transaction.
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SRS always use the most simple architecture to support complex transaction.
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@ -608,7 +608,7 @@ vhost with-hls.srs.com {
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# when codec changed, write the PAT/PMT table, but maybe ok util next ts.
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# when codec changed, write the PAT/PMT table, but maybe ok util next ts.
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# so user can set the default codec for mp3.
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# so user can set the default codec for mp3.
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# the available audio codec:
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# the available audio codec:
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# aac, mp3
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# aac, mp3, an
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# default: aac
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# default: aac
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hls_acodec aac;
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hls_acodec aac;
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# the default video codec of hls.
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# the default video codec of hls.
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@ -420,6 +420,9 @@ int SrsHlsMuxer::segment_open(int64_t segment_start_dts)
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} else if (default_acodec_str == "aac") {
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} else if (default_acodec_str == "aac") {
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default_acodec = SrsCodecAudioAAC;
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default_acodec = SrsCodecAudioAAC;
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srs_info("hls: use default aac acodec");
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srs_info("hls: use default aac acodec");
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} else if (default_acodec_str == "an") {
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default_acodec = SrsCodecAudioDisabled;
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srs_info("hls: use default an acodec for pure video");
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} else {
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} else {
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srs_warn("hls: use aac for other codec=%s", default_acodec_str.c_str());
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srs_warn("hls: use aac for other codec=%s", default_acodec_str.c_str());
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}
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}
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@ -136,6 +136,9 @@ enum SrsCodecAudio
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// set to the max value to reserved, for array map.
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// set to the max value to reserved, for array map.
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SrsCodecAudioReserved1 = 16,
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SrsCodecAudioReserved1 = 16,
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// for user to disable audio, for example, use pure video hls.
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SrsCodecAudioDisabled = 17,
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SrsCodecAudioLinearPCMPlatformEndian = 0,
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SrsCodecAudioLinearPCMPlatformEndian = 0,
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SrsCodecAudioADPCM = 1,
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SrsCodecAudioADPCM = 1,
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SrsCodecAudioMP3 = 2,
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SrsCodecAudioMP3 = 2,
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@ -215,13 +215,13 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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#define ERROR_HTTP_DVR_CREATE_REQUEST 3053
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#define ERROR_HTTP_DVR_CREATE_REQUEST 3053
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#define ERROR_HTTP_DVR_NO_TAEGET 3054
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#define ERROR_HTTP_DVR_NO_TAEGET 3054
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#define ERROR_ADTS_ID_NOT_AAC 3055
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#define ERROR_ADTS_ID_NOT_AAC 3055
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// HDS error code
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#define ERROR_HDS_OPEN_F4M_FAILED 3056
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#define ERROR_HDS_OPEN_F4M_FAILED 3056
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#define ERROR_HDS_WRITE_F4M_FAILED 3057
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#define ERROR_HDS_WRITE_F4M_FAILED 3057
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#define ERROR_HDS_OPEN_BOOTSTRAP_FAILED 3058
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#define ERROR_HDS_OPEN_BOOTSTRAP_FAILED 3058
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#define ERROR_HDS_WRITE_BOOTSTRAP_FAILED 3059
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#define ERROR_HDS_WRITE_BOOTSTRAP_FAILED 3059
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#define ERROR_HDS_OPEN_FRAGMENT_FAILED 3060
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#define ERROR_HDS_OPEN_FRAGMENT_FAILED 3060
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#define ERROR_HDS_WRITE_FRAGMENT_FAILED 3061
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#define ERROR_HDS_WRITE_FRAGMENT_FAILED 3061
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#define ERROR_HLS_NO_STREAM 3062
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///////////////////////////////////////////////////////
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///////////////////////////////////////////////////////
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// HTTP/StreamCaster protocol error.
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// HTTP/StreamCaster protocol error.
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@ -302,10 +302,12 @@ int SrsTsContext::encode(SrsFileWriter* writer, SrsTsMessage* msg, SrsCodecVideo
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vs = SrsTsStreamVideoH264;
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vs = SrsTsStreamVideoH264;
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video_pid = TS_VIDEO_AVC_PID;
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video_pid = TS_VIDEO_AVC_PID;
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break;
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break;
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case SrsCodecVideoDisabled:
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vs = SrsTsStreamReserved;
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break;
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case SrsCodecVideoReserved:
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case SrsCodecVideoReserved:
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case SrsCodecVideoReserved1:
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case SrsCodecVideoReserved1:
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case SrsCodecVideoReserved2:
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case SrsCodecVideoReserved2:
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case SrsCodecVideoDisabled:
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case SrsCodecVideoSorensonH263:
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case SrsCodecVideoSorensonH263:
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case SrsCodecVideoScreenVideo:
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case SrsCodecVideoScreenVideo:
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case SrsCodecVideoOn2VP6:
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case SrsCodecVideoOn2VP6:
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@ -323,6 +325,9 @@ int SrsTsContext::encode(SrsFileWriter* writer, SrsTsMessage* msg, SrsCodecVideo
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as = SrsTsStreamAudioMp3;
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as = SrsTsStreamAudioMp3;
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audio_pid = TS_AUDIO_MP3_PID;
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audio_pid = TS_AUDIO_MP3_PID;
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break;
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break;
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case SrsCodecAudioDisabled:
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as = SrsTsStreamReserved;
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break;
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case SrsCodecAudioReserved1:
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case SrsCodecAudioReserved1:
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case SrsCodecAudioLinearPCMPlatformEndian:
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case SrsCodecAudioLinearPCMPlatformEndian:
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case SrsCodecAudioADPCM:
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case SrsCodecAudioADPCM:
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@ -340,6 +345,12 @@ int SrsTsContext::encode(SrsFileWriter* writer, SrsTsMessage* msg, SrsCodecVideo
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break;
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break;
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}
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}
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if (as == SrsTsStreamReserved && vs == SrsTsStreamReserved) {
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ret = ERROR_HLS_NO_STREAM;
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srs_error("hls: no video or audio stream, vcodec=%d, acodec=%d. ret=%d", vc, ac, ret);
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return ret;
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}
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// when any codec changed, write PAT/PMT table.
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// when any codec changed, write PAT/PMT table.
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if (vcodec != vc || acodec != ac) {
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if (vcodec != vc || acodec != ac) {
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vcodec = vc;
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vcodec = vc;
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@ -360,6 +371,12 @@ int SrsTsContext::encode(SrsFileWriter* writer, SrsTsMessage* msg, SrsCodecVideo
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int SrsTsContext::encode_pat_pmt(SrsFileWriter* writer, int16_t vpid, SrsTsStream vs, int16_t apid, SrsTsStream as)
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int SrsTsContext::encode_pat_pmt(SrsFileWriter* writer, int16_t vpid, SrsTsStream vs, int16_t apid, SrsTsStream as)
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{
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{
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int ret = ERROR_SUCCESS;
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int ret = ERROR_SUCCESS;
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if (vs != SrsTsStreamVideoH264 && as != SrsTsStreamAudioAAC && as != SrsTsStreamAudioMp3) {
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ret = ERROR_HLS_NO_STREAM;
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srs_error("hls: no pmt pcr pid, vs=%d, as=%d. ret=%d", vs, as, ret);
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return ret;
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}
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int16_t pmt_number = TS_PMT_NUMBER;
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int16_t pmt_number = TS_PMT_NUMBER;
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int16_t pmt_pid = TS_PMT_PID;
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int16_t pmt_pid = TS_PMT_PID;
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@ -754,15 +771,17 @@ SrsTsPacket* SrsTsPacket::create_pmt(SrsTsContext* context, int16_t pmt_number,
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pmt->last_section_number = 0;
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pmt->last_section_number = 0;
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pmt->program_info_length = 0;
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pmt->program_info_length = 0;
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// use audio to carray pcr by default.
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// for hls, there must be atleast one audio channel.
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pmt->PCR_PID = apid;
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pmt->infos.push_back(new SrsTsPayloadPMTESInfo(as, apid));
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// if h.264 specified, use video to carry pcr.
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// if h.264 specified, use video to carry pcr.
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if (vs == SrsTsStreamVideoH264) {
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if (vs == SrsTsStreamVideoH264) {
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pmt->PCR_PID = vpid;
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pmt->PCR_PID = vpid;
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pmt->infos.push_back(new SrsTsPayloadPMTESInfo(vs, vpid));
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pmt->infos.push_back(new SrsTsPayloadPMTESInfo(vs, vpid));
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} else if (as == SrsTsStreamAudioAAC || as == SrsTsStreamAudioMp3) {
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// use audio to carray pcr by default.
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// for hls, there must be atleast one audio channel.
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pmt->PCR_PID = apid;
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pmt->infos.push_back(new SrsTsPayloadPMTESInfo(as, apid));
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} else {
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srs_assert(false);
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}
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}
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pmt->CRC_32 = 0; // calc in encode.
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pmt->CRC_32 = 0; // calc in encode.
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