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for #304, rewrite hls/ts code, support h.264+mp3 for hls. 2.0.117.
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8 changed files with 54 additions and 10 deletions
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@ -487,6 +487,8 @@ Supported operating systems and hardware:
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[#301](https://github.com/winlinvip/simple-rtmp-server/issues/301).
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1. Support push MPEG-TS over UDP to SRS, read
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[#250](https://github.com/winlinvip/simple-rtmp-server/issues/250).
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1. Rewrite HLS(h.264+aac/mp3) streaming, read
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[#304](https://github.com/winlinvip/simple-rtmp-server/issues/304).
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1. [no-plan] Support <500ms latency, FRSC(Fast RTMP-compatible Stream Channel tech).
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1. [no-plan] Support RTMP 302 redirect [#92](https://github.com/winlinvip/simple-rtmp-server/issues/92).
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1. [no-plan] Support multiple processes, for both origin and edge
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@ -525,6 +527,7 @@ Supported operating systems and hardware:
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### SRS 2.0 history
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* v2.0, 2015-02-15, for [#304](https://github.com/winlinvip/simple-rtmp-server/issues/304), rewrite hls/ts code, support h.264+mp3 for hls. 2.0.117.
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* v2.0, 2015-02-12, for [#304](https://github.com/winlinvip/simple-rtmp-server/issues/304), use stringstream to generate m3u8, add hls_td_ratio. 2.0.116.
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* v2.0, 2015-02-11, dev code ZhouGuowen for 2.0.115.
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* v2.0, 2015-02-10, for [#311](https://github.com/winlinvip/simple-rtmp-server/issues/311), set pcr_base to dts. 2.0.114.
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@ -492,6 +492,13 @@ vhost with-hls.srs.com {
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# @remark the hls_mount must endswith .m3u8.
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# default: [vhost]/[app]/[stream].m3u8
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hls_mount [vhost]/[app]/[stream].m3u8;
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# the default audio codec of hls.
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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: aac, mp3
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# default: aac
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# TODO: FIXME: update wiki for it.
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hls_acodec aac;
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}
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}
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# the vhost with hls disabled.
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@ -1480,7 +1480,7 @@ int SrsConfig::check_config()
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for (int j = 0; j < (int)conf->directives.size(); j++) {
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string m = conf->at(j)->name.c_str();
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if (m != "enabled" && m != "hls_path" && m != "hls_fragment" && m != "hls_window" && m != "hls_on_error"
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&& m != "hls_storage" && m != "hls_mount" && m != "hls_td_ratio"
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&& m != "hls_storage" && m != "hls_mount" && m != "hls_td_ratio" && m != "hls_acodec"
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) {
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ret = ERROR_SYSTEM_CONFIG_INVALID;
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srs_error("unsupported vhost hls directive %s, ret=%d", m.c_str(), ret);
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@ -3332,6 +3332,23 @@ string SrsConfig::get_hls_mount(string vhost)
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return conf->arg0();
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}
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string SrsConfig::get_hls_acodec(string vhost)
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{
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SrsConfDirective* hls = get_hls(vhost);
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if (!hls) {
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return SRS_CONF_DEFAULT_HLS_ACODEC;
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}
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SrsConfDirective* conf = hls->get("hls_acodec");
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if (!conf) {
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return SRS_CONF_DEFAULT_HLS_ACODEC;
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}
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return conf->arg0();
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}
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SrsConfDirective* SrsConfig::get_dvr(string vhost)
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{
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SrsConfDirective* conf = get_vhost(vhost);
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@ -55,6 +55,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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#define SRS_CONF_DEFAULT_HLS_ON_ERROR SRS_CONF_DEFAULT_HLS_ON_ERROR_IGNORE
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#define SRS_CONF_DEFAULT_HLS_STORAGE "disk"
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#define SRS_CONF_DEFAULT_HLS_MOUNT "[vhost]/[app]/[stream].m3u8"
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#define SRS_CONF_DEFAULT_HLS_ACODEC "aac"
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#define SRS_CONF_DEFAULT_DVR_PATH "./objs/nginx/html"
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#define SRS_CONF_DEFAULT_DVR_PLAN_SESSION "session"
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#define SRS_CONF_DEFAULT_DVR_PLAN_SEGMENT "segment"
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@ -922,6 +923,10 @@ public:
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* get the HLS mount url for HTTP server.
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*/
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virtual std::string get_hls_mount(std::string vhost);
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/**
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* get the HLS default audio codec.
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*/
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virtual std::string get_hls_acodec(std::string vhost);
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// dvr section
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private:
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/**
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@ -131,14 +131,14 @@ string SrsHlsCacheWriter::cache()
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return data;
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}
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SrsHlsSegment::SrsHlsSegment(bool write_cache, bool write_file)
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SrsHlsSegment::SrsHlsSegment(bool write_cache, bool write_file, SrsCodecAudio ac)
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{
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duration = 0;
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sequence_no = 0;
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segment_start_dts = 0;
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is_sequence_header = false;
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writer = new SrsHlsCacheWriter(write_cache, write_file);
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muxer = new SrsTSMuxer(writer);
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muxer = new SrsTSMuxer(writer, ac);
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}
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SrsHlsSegment::~SrsHlsSegment()
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@ -244,8 +244,21 @@ int SrsHlsMuxer::segment_open(int64_t segment_start_dts)
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// when segment open, the current segment must be NULL.
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srs_assert(!current);
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// load the default acodec from config.
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SrsCodecAudio default_acodec = SrsCodecAudioAAC;
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std::string default_acodec_str = _srs_config->get_hls_acodec(req->vhost);
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if (default_acodec_str == "mp3") {
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default_acodec = SrsCodecAudioMP3;
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srs_info("hls: use default mp3 acodec");
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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 {
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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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// new segment.
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current = new SrsHlsSegment(should_write_cache, should_write_file);
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current = new SrsHlsSegment(should_write_cache, should_write_file, default_acodec);
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current->sequence_no = _sequence_no++;
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current->segment_start_dts = segment_start_dts;
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@ -144,7 +144,7 @@ public:
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// whether current segement is sequence header.
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bool is_sequence_header;
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public:
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SrsHlsSegment(bool write_cache, bool write_file);
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SrsHlsSegment(bool write_cache, bool write_file, SrsCodecAudio ac);
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virtual ~SrsHlsSegment();
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public:
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/**
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@ -2916,13 +2916,12 @@ int SrsTsPayloadPMT::psi_encode(SrsStream* stream)
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return ret;
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}
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SrsTSMuxer::SrsTSMuxer(SrsFileWriter* w)
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SrsTSMuxer::SrsTSMuxer(SrsFileWriter* w, SrsCodecAudio ac)
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{
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writer = w;
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context = NULL;
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// default to aac.
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acodec = SrsCodecAudioAAC;
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acodec = ac;
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// default to avc(h.264)
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vcodec = SrsCodecVideoAVC;
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}
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@ -3296,7 +3295,7 @@ int SrsTsEncoder::initialize(SrsFileWriter* fs)
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_fs = fs;
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srs_freep(muxer);
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muxer = new SrsTSMuxer(fs);
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muxer = new SrsTSMuxer(fs, SrsCodecAudioAAC);
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if ((ret = muxer->open("")) != ERROR_SUCCESS) {
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return ret;
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@ -1547,7 +1547,7 @@ private:
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SrsFileWriter* writer;
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std::string path;
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public:
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SrsTSMuxer(SrsFileWriter* w);
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SrsTSMuxer(SrsFileWriter* w, SrsCodecAudio ac);
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virtual ~SrsTSMuxer();
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public:
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/**
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