for #304, rewrite hls/ts code, support h.264+mp3 for hls. 2.0.117.

@@ -487,6 +487,8 @@ Supported operating systems and hardware:
 [#301](https://github.com/winlinvip/simple-rtmp-server/issues/301).
 [#301](https://github.com/winlinvip/simple-rtmp-server/issues/301).
 1. Support push MPEG-TS over UDP to SRS, read 
 1. Support push MPEG-TS over UDP to SRS, read 
 [#250](https://github.com/winlinvip/simple-rtmp-server/issues/250).
 [#250](https://github.com/winlinvip/simple-rtmp-server/issues/250).
+1. Rewrite HLS(h.264+aac/mp3) streaming, read
+[#304](https://github.com/winlinvip/simple-rtmp-server/issues/304).
 1. [no-plan] Support <500ms latency, FRSC(Fast RTMP-compatible Stream Channel tech).
 1. [no-plan] Support <500ms latency, FRSC(Fast RTMP-compatible Stream Channel tech).
 1. [no-plan] Support RTMP 302 redirect [#92](https://github.com/winlinvip/simple-rtmp-server/issues/92).
 1. [no-plan] Support RTMP 302 redirect [#92](https://github.com/winlinvip/simple-rtmp-server/issues/92).
 1. [no-plan] Support multiple processes, for both origin and edge
 1. [no-plan] Support multiple processes, for both origin and edge
@@ -525,6 +527,7 @@ Supported operating systems and hardware:
 
 
 ### SRS 2.0 history
 ### SRS 2.0 history
 
 
+* 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.
 * 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.
 * 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.
 * v2.0, 2015-02-11, dev code ZhouGuowen for 2.0.115.
 * v2.0, 2015-02-11, dev code ZhouGuowen for 2.0.115.
 * v2.0, 2015-02-10, for [#311](https://github.com/winlinvip/simple-rtmp-server/issues/311), set pcr_base to dts. 2.0.114.
 * v2.0, 2015-02-10, for [#311](https://github.com/winlinvip/simple-rtmp-server/issues/311), set pcr_base to dts. 2.0.114.

@@ -492,6 +492,13 @@ vhost with-hls.srs.com {
         # @remark the hls_mount must endswith .m3u8.
         # @remark the hls_mount must endswith .m3u8.
         # default: [vhost]/[app]/[stream].m3u8
         # default: [vhost]/[app]/[stream].m3u8
         hls_mount       [vhost]/[app]/[stream].m3u8;
         hls_mount       [vhost]/[app]/[stream].m3u8;
+        # the default audio codec of hls.
+        # when codec changed, write the PAT/PMT table, but maybe ok util next ts.
+        # so user can set the default codec for mp3.
+        # the available audio codec: aac, mp3
+        # default: aac
+        # TODO: FIXME: update wiki for it.
+        hls_acodec      aac;
     }
     }
 }
 }
 # the vhost with hls disabled.
 # the vhost with hls disabled.

@@ -1480,7 +1480,7 @@ int SrsConfig::check_config()
                 for (int j = 0; j < (int)conf->directives.size(); j++) {
                 for (int j = 0; j < (int)conf->directives.size(); j++) {
                     string m = conf->at(j)->name.c_str();
                     string m = conf->at(j)->name.c_str();
                     if (m != "enabled" && m != "hls_path" && m != "hls_fragment" && m != "hls_window" && m != "hls_on_error"
                     if (m != "enabled" && m != "hls_path" && m != "hls_fragment" && m != "hls_window" && m != "hls_on_error"
-                        && m != "hls_storage" && m != "hls_mount" && m != "hls_td_ratio"
+                        && m != "hls_storage" && m != "hls_mount" && m != "hls_td_ratio" && m != "hls_acodec"
                     ) {
                     ) {
                         ret = ERROR_SYSTEM_CONFIG_INVALID;
                         ret = ERROR_SYSTEM_CONFIG_INVALID;
                         srs_error("unsupported vhost hls directive %s, ret=%d", m.c_str(), ret);
                         srs_error("unsupported vhost hls directive %s, ret=%d", m.c_str(), ret);
@@ -3332,6 +3332,23 @@ string SrsConfig::get_hls_mount(string vhost)
     return conf->arg0();
     return conf->arg0();
 }
 }
 
 
+string SrsConfig::get_hls_acodec(string vhost)
+{
+    SrsConfDirective* hls = get_hls(vhost);
+    
+    if (!hls) {
+        return SRS_CONF_DEFAULT_HLS_ACODEC;
+    }
+    
+    SrsConfDirective* conf = hls->get("hls_acodec");
+    
+    if (!conf) {
+        return SRS_CONF_DEFAULT_HLS_ACODEC;
+    }
+
+    return conf->arg0();
+}
+
 SrsConfDirective* SrsConfig::get_dvr(string vhost)
 SrsConfDirective* SrsConfig::get_dvr(string vhost)
 {
 {
     SrsConfDirective* conf = get_vhost(vhost);
     SrsConfDirective* conf = get_vhost(vhost);

@@ -55,6 +55,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #define SRS_CONF_DEFAULT_HLS_ON_ERROR SRS_CONF_DEFAULT_HLS_ON_ERROR_IGNORE
 #define SRS_CONF_DEFAULT_HLS_ON_ERROR SRS_CONF_DEFAULT_HLS_ON_ERROR_IGNORE
 #define SRS_CONF_DEFAULT_HLS_STORAGE "disk"
 #define SRS_CONF_DEFAULT_HLS_STORAGE "disk"
 #define SRS_CONF_DEFAULT_HLS_MOUNT "[vhost]/[app]/[stream].m3u8"
 #define SRS_CONF_DEFAULT_HLS_MOUNT "[vhost]/[app]/[stream].m3u8"
+#define SRS_CONF_DEFAULT_HLS_ACODEC "aac"
 #define SRS_CONF_DEFAULT_DVR_PATH "./objs/nginx/html"
 #define SRS_CONF_DEFAULT_DVR_PATH "./objs/nginx/html"
 #define SRS_CONF_DEFAULT_DVR_PLAN_SESSION "session"
 #define SRS_CONF_DEFAULT_DVR_PLAN_SESSION "session"
 #define SRS_CONF_DEFAULT_DVR_PLAN_SEGMENT "segment"
 #define SRS_CONF_DEFAULT_DVR_PLAN_SEGMENT "segment"
@@ -922,6 +923,10 @@ public:
     * get the HLS mount url for HTTP server.
     * get the HLS mount url for HTTP server.
     */
     */
     virtual std::string         get_hls_mount(std::string vhost);
     virtual std::string         get_hls_mount(std::string vhost);
+    /**
+    * get the HLS default audio codec.
+    */
+    virtual std::string         get_hls_acodec(std::string vhost);
 // dvr section
 // dvr section
 private:
 private:
     /**
     /**

@@ -131,14 +131,14 @@ string SrsHlsCacheWriter::cache()
     return data;
     return data;
 }
 }
 
 
-SrsHlsSegment::SrsHlsSegment(bool write_cache, bool write_file)
+SrsHlsSegment::SrsHlsSegment(bool write_cache, bool write_file, SrsCodecAudio ac)
 {
 {
     duration = 0;
     duration = 0;
     sequence_no = 0;
     sequence_no = 0;
     segment_start_dts = 0;
     segment_start_dts = 0;
     is_sequence_header = false;
     is_sequence_header = false;
     writer = new SrsHlsCacheWriter(write_cache, write_file);
     writer = new SrsHlsCacheWriter(write_cache, write_file);
-    muxer = new SrsTSMuxer(writer);
+    muxer = new SrsTSMuxer(writer, ac);
 }
 }
 
 
 SrsHlsSegment::~SrsHlsSegment()
 SrsHlsSegment::~SrsHlsSegment()
@@ -243,9 +243,22 @@ int SrsHlsMuxer::segment_open(int64_t segment_start_dts)
     
     
     // when segment open, the current segment must be NULL.
     // when segment open, the current segment must be NULL.
     srs_assert(!current);
     srs_assert(!current);
+
+    // load the default acodec from config.
+    SrsCodecAudio default_acodec = SrsCodecAudioAAC;
+    std::string default_acodec_str = _srs_config->get_hls_acodec(req->vhost);
+    if (default_acodec_str == "mp3") {
+        default_acodec = SrsCodecAudioMP3;
+        srs_info("hls: use default mp3 acodec");
+    } else if (default_acodec_str == "aac") {
+        default_acodec = SrsCodecAudioAAC;
+        srs_info("hls: use default aac acodec");
+    } else {
+        srs_warn("hls: use aac for other codec=%s", default_acodec_str.c_str());
+    }
     
     
     // new segment.
     // new segment.
-    current = new SrsHlsSegment(should_write_cache, should_write_file);
+    current = new SrsHlsSegment(should_write_cache, should_write_file, default_acodec);
     current->sequence_no = _sequence_no++;
     current->sequence_no = _sequence_no++;
     current->segment_start_dts = segment_start_dts;
     current->segment_start_dts = segment_start_dts;
     
     

@@ -144,7 +144,7 @@ public:
     // whether current segement is sequence header.
     // whether current segement is sequence header.
     bool is_sequence_header;
     bool is_sequence_header;
 public:
 public:
-    SrsHlsSegment(bool write_cache, bool write_file);
+    SrsHlsSegment(bool write_cache, bool write_file, SrsCodecAudio ac);
     virtual ~SrsHlsSegment();
     virtual ~SrsHlsSegment();
 public:
 public:
     /**
     /**

@@ -2916,13 +2916,12 @@ int SrsTsPayloadPMT::psi_encode(SrsStream* stream)
     return ret;
     return ret;
 }
 }
 
 
-SrsTSMuxer::SrsTSMuxer(SrsFileWriter* w)
+SrsTSMuxer::SrsTSMuxer(SrsFileWriter* w, SrsCodecAudio ac)
 {
 {
     writer = w;
     writer = w;
     context = NULL;
     context = NULL;
 
 
-    // default to aac.
-    acodec = SrsCodecAudioAAC;
+    acodec = ac;
     // default to avc(h.264)
     // default to avc(h.264)
     vcodec = SrsCodecVideoAVC;
     vcodec = SrsCodecVideoAVC;
 }
 }
@@ -3296,7 +3295,7 @@ int SrsTsEncoder::initialize(SrsFileWriter* fs)
     _fs = fs;
     _fs = fs;
 
 
     srs_freep(muxer);
     srs_freep(muxer);
-    muxer = new SrsTSMuxer(fs);
+    muxer = new SrsTSMuxer(fs, SrsCodecAudioAAC);
 
 
     if ((ret = muxer->open("")) != ERROR_SUCCESS) {
     if ((ret = muxer->open("")) != ERROR_SUCCESS) {
         return ret;
         return ret;

@@ -1547,7 +1547,7 @@ private:
     SrsFileWriter* writer;
     SrsFileWriter* writer;
     std::string path;
     std::string path;
 public:
 public:
-    SrsTSMuxer(SrsFileWriter* w);
+    SrsTSMuxer(SrsFileWriter* w, SrsCodecAudio ac);
     virtual ~SrsTSMuxer();
     virtual ~SrsTSMuxer();
 public:
 public:
     /**
     /**