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fix #409: support pure video hls. 2.0.172.
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7 changed files with 35 additions and 9 deletions
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@ -342,6 +342,7 @@ Remark:
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## 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, 2015-05-24, 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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@ -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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# so user can set the default codec for mp3.
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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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hls_acodec aac;
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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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default_acodec = SrsCodecAudioAAC;
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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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srs_warn("hls: use aac for other codec=%s", default_acodec_str.c_str());
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}
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@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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// current release version
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#define VERSION_MAJOR 2
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#define VERSION_MINOR 0
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#define VERSION_REVISION 171
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#define VERSION_REVISION 172
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// server info.
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#define RTMP_SIG_SRS_KEY "SRS"
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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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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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SrsCodecAudioADPCM = 1,
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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_NO_TAEGET 3054
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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_WRITE_F4M_FAILED 3057
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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_OPEN_FRAGMENT_FAILED 3060
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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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// 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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video_pid = TS_VIDEO_AVC_PID;
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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 SrsCodecVideoReserved1:
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case SrsCodecVideoReserved2:
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case SrsCodecVideoDisabled:
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case SrsCodecVideoSorensonH263:
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case SrsCodecVideoScreenVideo:
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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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audio_pid = TS_AUDIO_MP3_PID;
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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 SrsCodecAudioLinearPCMPlatformEndian:
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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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}
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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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if (vcodec != vc || acodec != ac) {
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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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{
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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_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->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 (vs == SrsTsStreamVideoH264) {
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pmt->PCR_PID = 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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pmt->CRC_32 = 0; // calc in encode.
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