for #301, hls support h.264+mp3, ok for vlc. 2.0.107

@@ -521,6 +521,7 @@ Supported operating systems and hardware:
 
 
 ### SRS 2.0 history
 ### SRS 2.0 history
 
 
+* v2.0, 2015-01-25, for [#301](https://github.com/winlinvip/simple-rtmp-server/issues/301), hls support h.264+mp3, ok for vlc. 2.0.107
 * v2.0, 2015-01-25, for [#301](https://github.com/winlinvip/simple-rtmp-server/issues/301), http ts stream support h.264+mp3. 2.0.106
 * v2.0, 2015-01-25, for [#301](https://github.com/winlinvip/simple-rtmp-server/issues/301), http ts stream support h.264+mp3. 2.0.106
 * v2.0, 2015-01-25, hotfix [#268](https://github.com/winlinvip/simple-rtmp-server/issues/268), refine the pcr start at 0, dts/pts plus delay. 2.0.105
 * v2.0, 2015-01-25, hotfix [#268](https://github.com/winlinvip/simple-rtmp-server/issues/268), refine the pcr start at 0, dts/pts plus delay. 2.0.105
 * v2.0, 2015-01-25, hotfix [#151](https://github.com/winlinvip/simple-rtmp-server/issues/151), refine pcr=dts-800ms and use dts/pts directly. 2.0.104
 * v2.0, 2015-01-25, hotfix [#151](https://github.com/winlinvip/simple-rtmp-server/issues/151), refine pcr=dts-800ms and use dts/pts directly. 2.0.104

@@ -92,6 +92,7 @@ SrsHlsMuxer::SrsHlsMuxer()
     hls_fragment = hls_window = 0;
     hls_fragment = hls_window = 0;
     _sequence_no = 0;
     _sequence_no = 0;
     current = NULL;
     current = NULL;
+    acodec = SrsCodecAudioReserved1;
 }
 }
 
 
 SrsHlsMuxer::~SrsHlsMuxer()
 SrsHlsMuxer::~SrsHlsMuxer()
@@ -170,6 +171,11 @@ int SrsHlsMuxer::segment_open(int64_t segment_start_dts)
     }
     }
     srs_info("open HLS muxer success. path=%s, tmp=%s", 
     srs_info("open HLS muxer success. path=%s, tmp=%s", 
         current->full_path.c_str(), tmp_file.c_str());
         current->full_path.c_str(), tmp_file.c_str());
+
+    // set the segment muxer audio codec.
+    if (acodec != SrsCodecAudioReserved1) {
+        current->muxer->update_acodec(acodec);
+    }
     
     
     return ret;
     return ret;
 }
 }
@@ -199,11 +205,12 @@ bool SrsHlsMuxer::is_segment_absolutely_overflow()
     return current->duration >= 2 * hls_fragment;
     return current->duration >= 2 * hls_fragment;
 }
 }
 
 
-int SrsHlsMuxer::update_acodec(SrsCodecAudio acodec)
+int SrsHlsMuxer::update_acodec(SrsCodecAudio ac)
 {
 {
     srs_assert(current);
     srs_assert(current);
     srs_assert(current->muxer);
     srs_assert(current->muxer);
-    return current->muxer->update_acodec(acodec);
+    acodec = ac;
+    return current->muxer->update_acodec(ac);
 }
 }
 
 
 int SrsHlsMuxer::flush_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab)
 int SrsHlsMuxer::flush_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab)
@@ -801,7 +808,7 @@ int SrsHls::on_audio(SrsSharedPtrMessage* __audio)
     }
     }
     
     
     // ignore sequence header
     // ignore sequence header
-    if (sample->aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
+    if (acodec == SrsCodecAudioAAC && sample->aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
         return hls_cache->on_sequence_header(muxer);
         return hls_cache->on_sequence_header(muxer);
     }
     }
     
     

@@ -118,6 +118,12 @@ private:
     * current writing segment.
     * current writing segment.
     */
     */
     SrsHlsSegment* current;
     SrsHlsSegment* current;
+    /**
+    * the current audio codec, when open new muxer,
+    * set the muxer audio codec.
+    * @see https://github.com/winlinvip/simple-rtmp-server/issues/301
+    */
+    SrsCodecAudio acodec;
 public:
 public:
     SrsHlsMuxer();
     SrsHlsMuxer();
     virtual ~SrsHlsMuxer();
     virtual ~SrsHlsMuxer();
@@ -144,7 +150,7 @@ public:
     */
     */
     virtual bool is_segment_absolutely_overflow();
     virtual bool is_segment_absolutely_overflow();
 public:
 public:
-    virtual int update_acodec(SrsCodecAudio acodec);
+    virtual int update_acodec(SrsCodecAudio ac);
     virtual int flush_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab);
     virtual int flush_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab);
     virtual int flush_video(SrsMpegtsFrame* af, SrsSimpleBuffer* ab, SrsMpegtsFrame* vf, SrsSimpleBuffer* vb);
     virtual int flush_video(SrsMpegtsFrame* af, SrsSimpleBuffer* ab, SrsMpegtsFrame* vf, SrsSimpleBuffer* vb);
     /**
     /**

@@ -352,6 +352,7 @@ int SrsGoHttpFileServer::serve_file(ISrsGoHttpResponseWriter* w, SrsHttpMessage*
         _mime[".zip"] = "application/zip";
         _mime[".zip"] = "application/zip";
         _mime[".rar"] = "application/x-rar-compressed";
         _mime[".rar"] = "application/x-rar-compressed";
         _mime[".xml"] = "text/xml;charset=utf-8";
         _mime[".xml"] = "text/xml;charset=utf-8";
+        _mime[".html"] = "text/html;charset=utf-8";
         _mime[".js"] = "text/javascript";
         _mime[".js"] = "text/javascript";
         _mime[".css"] = "text/css;charset=utf-8";
         _mime[".css"] = "text/css;charset=utf-8";
         _mime[".ico"] = "image/x-icon";
         _mime[".ico"] = "image/x-icon";

@@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 // current release version
 // current release version
 #define VERSION_MAJOR       2
 #define VERSION_MAJOR       2
 #define VERSION_MINOR       0
 #define VERSION_MINOR       0
-#define VERSION_REVISION    106
+#define VERSION_REVISION    107
 
 
 // server info.
 // server info.
 #define RTMP_SIG_SRS_KEY "SRS"
 #define RTMP_SIG_SRS_KEY "SRS"

@@ -430,14 +430,14 @@ int SrsTSMuxer::open(string _path)
     return ret;
     return ret;
 }
 }
 
 
-int SrsTSMuxer::update_acodec(SrsCodecAudio acodec)
+int SrsTSMuxer::update_acodec(SrsCodecAudio ac)
 {
 {
     int ret = ERROR_SUCCESS;
     int ret = ERROR_SUCCESS;
 
 
-    if (current == acodec) {
+    if (current == ac) {
         return ret;
         return ret;
     }
     }
-    current = acodec;
+    current = ac;
 
 
     return ret;
     return ret;
 }
 }

@@ -85,7 +85,7 @@ public:
     * @remark for audio aac codec, for example, SRS1, it's ok to write PSI when open ts.
     * @remark for audio aac codec, for example, SRS1, it's ok to write PSI when open ts.
     * @see https://github.com/winlinvip/simple-rtmp-server/issues/301
     * @see https://github.com/winlinvip/simple-rtmp-server/issues/301
     */
     */
-    virtual int update_acodec(SrsCodecAudio acodec);
+    virtual int update_acodec(SrsCodecAudio ac);
     /**
     /**
     * write an audio frame to ts, 
     * write an audio frame to ts, 
     * @remark write PSI first when not write yet.
     * @remark write PSI first when not write yet.