for #301, http ts stream support h.264+mp3. 2.0.106

@@ -521,6 +521,7 @@ Supported operating systems and hardware:
 
 ### SRS 2.0 history
 
+* 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 [#151](https://github.com/winlinvip/simple-rtmp-server/issues/151), refine pcr=dts-800ms and use dts/pts directly. 2.0.104
 * v2.0, 2015-01-23, hotfix [#151](https://github.com/winlinvip/simple-rtmp-server/issues/151), use absolutely overflow to make jwplayer happy. 2.0.103

@@ -199,6 +199,13 @@ bool SrsHlsMuxer::is_segment_absolutely_overflow()
     return current->duration >= 2 * hls_fragment;
 }
 
+int SrsHlsMuxer::update_acodec(SrsCodecAudio acodec)
+{
+    srs_assert(current);
+    srs_assert(current->muxer);
+    return current->muxer->update_acodec(acodec);
+}
+
 int SrsHlsMuxer::flush_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab)
 {
     int ret = ERROR_SUCCESS;
@@ -572,8 +579,6 @@ int SrsHlsCache::on_sequence_header(SrsHlsMuxer* muxer)
 int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t pts, SrsCodecSample* sample)
 {
     int ret = ERROR_SUCCESS;
-
-    audio_buffer_start_pts = pts;
     
     // write audio to cache.
     if ((ret = cache->cache_audio(codec, pts, sample)) != ERROR_SUCCESS) {
@@ -591,7 +596,7 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
     // in ms, audio delay to flush the audios.
     int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
     // flush if audio delay exceed
-    if (pts - audio_buffer_start_pts > audio_delay * 90) {
+    if (pts - cache->audio_buffer_start_pts > audio_delay * 90) {
         if ((ret = muxer->flush_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
             return ret;
         }
@@ -773,11 +778,25 @@ int SrsHls::on_audio(SrsSharedPtrMessage* __audio)
     
     sample->clear();
     if ((ret = codec->audio_aac_demux(audio->payload, audio->size, sample)) != ERROR_SUCCESS) {
-        srs_error("hls codec demux audio failed. ret=%d", ret);
-        return ret;
+        if (ret != ERROR_HLS_TRY_MP3) {
+            srs_error("hls aac demux audio failed. ret=%d", ret);
+            return ret;
+        }
+        if ((ret = codec->audio_mp3_demux(audio->payload, audio->size, sample)) != ERROR_SUCCESS) {
+            srs_error("hls mp3 demux audio failed. ret=%d", ret);
+            return ret;
+        }
     }
+    SrsCodecAudio acodec = (SrsCodecAudio)codec->audio_codec_id;
     
-    if (codec->audio_codec_id != SrsCodecAudioAAC) {
+    // ts support audio codec: aac/mp3
+    if (acodec != SrsCodecAudioAAC && acodec != SrsCodecAudioMP3) {
+        return ret;
+    }
+
+    // when codec changed, write new header.
+    if ((ret = muxer->update_acodec(acodec)) != ERROR_SUCCESS) {
+        srs_error("http: ts audio write header failed. ret=%d", ret);
         return ret;
     }
     

@@ -37,6 +37,8 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #include <string>
 #include <vector>
 
+#include <srs_kernel_codec.hpp>
+
 class SrsSharedPtrMessage;
 class SrsCodecSample;
 class SrsMpegtsFrame;
@@ -141,6 +143,8 @@ public:
     * @see https://github.com/winlinvip/simple-rtmp-server/issues/151#issuecomment-71155184
     */
     virtual bool is_segment_absolutely_overflow();
+public:
+    virtual int update_acodec(SrsCodecAudio acodec);
     virtual int flush_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab);
     virtual int flush_video(SrsMpegtsFrame* af, SrsSimpleBuffer* ab, SrsMpegtsFrame* vf, SrsSimpleBuffer* vb);
     /**
@@ -174,8 +178,6 @@ private:
 class SrsHlsCache
 {
 private:
-    // the audio cache buffer start pts, to flush audio if full.
-    int64_t audio_buffer_start_pts;
     SrsTsCache* cache;
 public:
     SrsHlsCache();

@@ -1349,12 +1349,17 @@ int SrsSource::on_audio(SrsCommonMessage* __audio)
         }
     }
 
-    // cache the sequence header if h264
-    // donot cache the sequence header to gop_cache, return here.
-    if (SrsFlvCodec::audio_is_sequence_header(msg.payload, msg.size)) {
+    // cache the sequence header of aac, or first packet of mp3.
+    // for example, the mp3 is used for hls to write the "right" audio codec.
+    bool is_aac_sequence_header = SrsFlvCodec::audio_is_sequence_header(msg.payload, msg.size);
+    if (is_aac_sequence_header || !cache_sh_audio) {
         srs_freep(cache_sh_audio);
         cache_sh_audio = msg.copy();
-        
+    }
+
+    // cache the sequence header if aac
+    // donot cache the sequence header to gop_cache, return here.
+    if (is_aac_sequence_header) {
         // parse detail audio codec
         SrsAvcAacCodec codec;
         SrsCodecSample sample;
@@ -1768,18 +1773,20 @@ int SrsSource::create_consumer(SrsConsumer*& consumer, bool ds, bool dm, bool dg
     srs_info("dispatch metadata success");
     
     // copy sequence header
+    // copy audio sequence first, for hls to fast parse the "right" audio codec.
+    // @see https://github.com/winlinvip/simple-rtmp-server/issues/301
+    if (ds && cache_sh_audio && (ret = consumer->enqueue(cache_sh_audio, atc, tba, tbv, ag)) != ERROR_SUCCESS) {
+        srs_error("dispatch audio sequence header failed. ret=%d", ret);
+        return ret;
+    }
+    srs_info("dispatch audio sequence header success");
+
     if (ds && cache_sh_video && (ret = consumer->enqueue(cache_sh_video, atc, tba, tbv, ag)) != ERROR_SUCCESS) {
         srs_error("dispatch video sequence header failed. ret=%d", ret);
         return ret;
     }
     srs_info("dispatch video sequence header success");
     
-    if (cache_sh_audio && (ret = consumer->enqueue(cache_sh_audio, atc, tba, tbv, ag)) != ERROR_SUCCESS) {
-        srs_error("dispatch audio sequence header failed. ret=%d", ret);
-        return ret;
-    }
-    srs_info("dispatch audio sequence header success");
-    
     // copy gop cache to client.
     if (dg && (ret = gop_cache->dump(consumer, atc, tba, tbv, ag)) != ERROR_SUCCESS) {
         return ret;

@@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 // current release version
 #define VERSION_MAJOR       2
 #define VERSION_MINOR       0
-#define VERSION_REVISION    105
+#define VERSION_REVISION    106
 
 // server info.
 #define RTMP_SIG_SRS_KEY "SRS"

@@ -60,6 +60,7 @@ void SrsCodecSample::clear()
     frame_type = SrsCodecVideoAVCFrameReserved;
     avc_packet_type = SrsCodecVideoAVCTypeReserved;
     
+    acodec = SrsCodecAudioReserved1;
     sound_rate = SrsCodecAudioSampleRateReserved;
     sound_size = SrsCodecAudioSampleSizeReserved;
     sound_type = SrsCodecAudioSoundTypeReserved;
@@ -91,10 +92,13 @@ SrsAvcAacCodec::SrsAvcAacCodec()
     duration                    = 0;
     NAL_unit_length             = 0;
     frame_rate                  = 0;
+
     video_data_rate             = 0;
     video_codec_id              = 0;
+
     audio_data_rate             = 0;
     audio_codec_id              = 0;
+
     avc_profile                 = 0;
     avc_level                   = 0;
     aac_profile                 = 0;
@@ -104,6 +108,7 @@ SrsAvcAacCodec::SrsAvcAacCodec()
     avc_extra_data              = NULL;
     aac_extra_size              = 0;
     aac_extra_data              = NULL;
+
     sequenceParameterSetLength  = 0;
     sequenceParameterSetNALUnit = NULL;
     pictureParameterSetLength   = 0;
@@ -129,7 +134,7 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
     sample->is_video = false;
     
     if (!data || size <= 0) {
-        srs_trace("no audio present, hls ignore it.");
+        srs_trace("no audio present, ignore it.");
         return ret;
     }
     
@@ -140,7 +145,7 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
     // audio decode
     if (!stream->require(1)) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls decode audio sound_format failed. ret=%d", ret);
+        srs_error("audio codec decode sound_format failed. ret=%d", ret);
         return ret;
     }
     
@@ -153,20 +158,27 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
     sound_format = (sound_format >> 4) & 0x0f;
     
     audio_codec_id = sound_format;
+    sample->acodec = (SrsCodecAudio)audio_codec_id;
+
     sample->sound_type = (SrsCodecAudioSoundType)sound_type;
     sample->sound_rate = (SrsCodecAudioSampleRate)sound_rate;
     sample->sound_size = (SrsCodecAudioSampleSize)sound_size;
+
+    // we support h.264+mp3 for hls.
+    if (audio_codec_id == SrsCodecAudioMP3) {
+        return ERROR_HLS_TRY_MP3;
+    }
     
     // only support aac
     if (audio_codec_id != SrsCodecAudioAAC) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls only support audio aac codec. actual=%d, ret=%d", audio_codec_id, ret);
+        srs_error("audio codec only support mp3/aac codec. actual=%d, ret=%d", audio_codec_id, ret);
         return ret;
     }
 
     if (!stream->require(1)) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls decode audio aac_packet_type failed. ret=%d", ret);
+        srs_error("audio codec decode aac_packet_type failed. ret=%d", ret);
         return ret;
     }
     
@@ -189,7 +201,7 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
         // channelConfiguration, aac_channels, 4bits
         if (!stream->require(2)) {
             ret = ERROR_HLS_DECODE_ERROR;
-            srs_error("hls decode audio aac sequence header failed. ret=%d", ret);
+            srs_error("audio codec decode aac sequence header failed. ret=%d", ret);
             return ret;
         }
         aac_profile = stream->read_1bytes();
@@ -201,7 +213,7 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
         
         if (aac_profile == 0 || aac_profile == 0x1f) {
             ret = ERROR_HLS_DECODE_ERROR;
-            srs_error("hls decode audio aac sequence header failed, "
+            srs_error("audio codec decode aac sequence header failed, "
                 "adts object=%d invalid. ret=%d", aac_profile, ret);
             return ret;
         }
@@ -221,14 +233,14 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
         // ensure the sequence header demuxed
         if (aac_extra_size <= 0 || !aac_extra_data) {
             ret = ERROR_HLS_DECODE_ERROR;
-            srs_error("hls decode audio aac failed, sequence header not found. ret=%d", ret);
+            srs_error("audio codec decode aac failed, sequence header not found. ret=%d", ret);
             return ret;
         }
         
         // Raw AAC frame data in UI8 []
         // 6.3 Raw Data, aac-iso-13818-7.pdf, page 28
         if ((ret = sample->add_sample_unit(stream->data() + stream->pos(), stream->size() - stream->pos())) != ERROR_SUCCESS) {
-            srs_error("hls add audio sample failed. ret=%d", ret);
+            srs_error("audio codec add sample failed. ret=%d", ret);
             return ret;
         }
     } else {
@@ -264,6 +276,31 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
     return ret;
 }
 
+int SrsAvcAacCodec::audio_mp3_demux(char* data, int size, SrsCodecSample* sample)
+{
+    int ret = ERROR_SUCCESS;
+
+    // we always decode aac then mp3.
+    srs_assert(sample->acodec == SrsCodecAudioMP3);
+    
+    // @see: E.4.2 Audio Tags, video_file_format_spec_v10_1.pdf, page 76
+    if (!data || size <= 1) {
+        srs_trace("no mp3 audio present, ignore it.");
+        return ret;
+    }
+
+    // mp3 payload.
+    if ((ret = sample->add_sample_unit(data + 1, size - 1)) != ERROR_SUCCESS) {
+        srs_error("audio codec add mp3 sample failed. ret=%d", ret);
+        return ret;
+    }
+    
+    srs_info("audio decoded, type=%d, codec=%d, asize=%d, rate=%d, format=%d, size=%d", 
+        sample->sound_type, audio_codec_id, sample->sound_size, sample->sound_rate, sample->acodec, size);
+    
+    return ret;
+}
+
 int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample)
 {
     int ret = ERROR_SUCCESS;
@@ -271,7 +308,7 @@ int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample
     sample->is_video = true;
     
     if (!data || size <= 0) {
-        srs_trace("no video present, hls ignore it.");
+        srs_trace("no video present, ignore it.");
         return ret;
     }
     
@@ -282,7 +319,7 @@ int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample
     // video decode
     if (!stream->require(1)) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls decode video frame_type failed. ret=%d", ret);
+        srs_error("video codec decode frame_type failed. ret=%d", ret);
         return ret;
     }
     
@@ -296,21 +333,21 @@ int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample
     // ignore info frame without error,
     // @see https://github.com/winlinvip/simple-rtmp-server/issues/288#issuecomment-69863909
     if (sample->frame_type == SrsCodecVideoAVCFrameVideoInfoFrame) {
-        srs_warn("hls igone the info frame, ret=%d", ret);
+        srs_warn("video codec igone the info frame, ret=%d", ret);
         return ret;
     }
     
     // only support h.264/avc
     if (codec_id != SrsCodecVideoAVC) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls only support video h.264/avc codec. actual=%d, ret=%d", codec_id, ret);
+        srs_error("video codec only support video h.264/avc codec. actual=%d, ret=%d", codec_id, ret);
         return ret;
     }
     video_codec_id = codec_id;
     
     if (!stream->require(4)) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls decode video avc_packet_type failed. ret=%d", ret);
+        srs_error("video codec decode avc_packet_type failed. ret=%d", ret);
         return ret;
     }
     int8_t avc_packet_type = stream->read_1bytes();
@@ -328,7 +365,7 @@ int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample
         // ensure the sequence header demuxed
         if (avc_extra_size <= 0 || !avc_extra_data) {
             ret = ERROR_HLS_DECODE_ERROR;
-            srs_error("hls decode video avc failed, sequence header not found. ret=%d", ret);
+            srs_error("avc decode failed, sequence header not found. ret=%d", ret);
             return ret;
         }
         
@@ -371,7 +408,7 @@ int SrsAvcAacCodec::avc_demux_sps_pps(SrsStream* stream)
     
     if (!stream->require(6)) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls decode video avc sequenc header failed. ret=%d", ret);
+        srs_error("avc decode sequenc header failed. ret=%d", ret);
         return ret;
     }
     //int8_t configurationVersion = stream->read_1bytes();
@@ -402,25 +439,25 @@ int SrsAvcAacCodec::avc_demux_sps_pps(SrsStream* stream)
     // 1 sps
     if (!stream->require(1)) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls decode video avc sequenc header sps failed. ret=%d", ret);
+        srs_error("avc decode sequenc header sps failed. ret=%d", ret);
         return ret;
     }
     int8_t numOfSequenceParameterSets = stream->read_1bytes();
     numOfSequenceParameterSets &= 0x1f;
     if (numOfSequenceParameterSets != 1) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls decode video avc sequenc header sps failed. ret=%d", ret);
+        srs_error("avc decode sequenc header sps failed. ret=%d", ret);
         return ret;
     }
     if (!stream->require(2)) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls decode video avc sequenc header sps size failed. ret=%d", ret);
+        srs_error("avc decode sequenc header sps size failed. ret=%d", ret);
         return ret;
     }
     sequenceParameterSetLength = stream->read_2bytes();
     if (!stream->require(sequenceParameterSetLength)) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls decode video avc sequenc header sps data failed. ret=%d", ret);
+        srs_error("avc decode sequenc header sps data failed. ret=%d", ret);
         return ret;
     }
     if (sequenceParameterSetLength > 0) {
@@ -432,25 +469,25 @@ int SrsAvcAacCodec::avc_demux_sps_pps(SrsStream* stream)
     // 1 pps
     if (!stream->require(1)) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls decode video avc sequenc header pps failed. ret=%d", ret);
+        srs_error("avc decode sequenc header pps failed. ret=%d", ret);
         return ret;
     }
     int8_t numOfPictureParameterSets = stream->read_1bytes();
     numOfPictureParameterSets &= 0x1f;
     if (numOfPictureParameterSets != 1) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls decode video avc sequenc header pps failed. ret=%d", ret);
+        srs_error("avc decode sequenc header pps failed. ret=%d", ret);
         return ret;
     }
     if (!stream->require(2)) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls decode video avc sequenc header pps size failed. ret=%d", ret);
+        srs_error("avc decode sequenc header pps size failed. ret=%d", ret);
         return ret;
     }
     pictureParameterSetLength = stream->read_2bytes();
     if (!stream->require(pictureParameterSetLength)) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("hls decode video avc sequenc header pps data failed. ret=%d", ret);
+        srs_error("avc decode sequenc header pps data failed. ret=%d", ret);
         return ret;
     }
     if (pictureParameterSetLength > 0) {
@@ -534,7 +571,7 @@ int SrsAvcAacCodec::avc_demux_ibmf_format(SrsStream* stream, SrsCodecSample* sam
         // unsigned int((NAL_unit_length+1)*8) NALUnitLength;
         if (!stream->require(NAL_unit_length + 1)) {
             ret = ERROR_HLS_DECODE_ERROR;
-            srs_error("hls decode video avc NALU size failed. ret=%d", ret);
+            srs_error("avc decode NALU size failed. ret=%d", ret);
             return ret;
         }
         int32_t NALUnitLength = 0;
@@ -557,12 +594,12 @@ int SrsAvcAacCodec::avc_demux_ibmf_format(SrsStream* stream, SrsCodecSample* sam
         // NALUnit
         if (!stream->require(NALUnitLength)) {
             ret = ERROR_HLS_DECODE_ERROR;
-            srs_error("hls decode video avc NALU data failed. ret=%d", ret);
+            srs_error("avc decode NALU data failed. ret=%d", ret);
             return ret;
         }
         // 7.3.1 NAL unit syntax, H.264-AVC-ISO_IEC_14496-10.pdf, page 44.
         if ((ret = sample->add_sample_unit(stream->data() + stream->pos(), NALUnitLength)) != ERROR_SUCCESS) {
-            srs_error("hls add video sample failed. ret=%d", ret);
+            srs_error("avc add video sample failed. ret=%d", ret);
             return ret;
         }
         stream->skip(NALUnitLength);

@@ -156,6 +156,8 @@ public:
     SrsCodecVideoAVCType avc_packet_type;
 public:
     // audio specified
+    SrsCodecAudio acodec;
+    // audio aac specified.
     SrsCodecAudioSampleRate sound_rate;
     SrsCodecAudioSampleSize sound_size;
     SrsCodecAudioSoundType sound_type;
@@ -271,6 +273,7 @@ public:
     * demux the aac raw to sample units.
     */
     virtual int audio_aac_demux(char* data, int size, SrsCodecSample* sample);
+    virtual int audio_mp3_demux(char* data, int size, SrsCodecSample* sample);
     /**
     * demux the video packet in h.264 codec.
     * the packet mux in FLV/RTMP format defined in flv specification.

@@ -202,6 +202,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #define ERROR_AAC_REQUIRED_ADTS             3046
 #define ERROR_AAC_ADTS_HEADER               3047
 #define ERROR_AAC_DATA_INVALID              3048
+#define ERROR_HLS_TRY_MP3                   3049
 
 ///////////////////////////////////////////////////////
 // HTTP/StreamCaster protocol error.

@@ -53,6 +53,7 @@ using namespace std;
 
 // ts aac stream id.
 #define TS_AUDIO_AAC 0xc0
+#define TS_AUDIO_MP3 0x04
 // ts avc stream id.
 #define TS_VIDEO_AVC 0xe0
 
@@ -118,11 +119,18 @@ u_int8_t mpegts_header[] = {
     // must generate header with/without video, @see:
     // https://github.com/winlinvip/simple-rtmp-server/issues/40
     0x1b, 0xe1, 0x00, 0xf0, 0x00, /* h264, pid=0x100=256 */
+};
+u_int8_t mpegts_header_aac[] = {
     0x0f, 0xe1, 0x01, 0xf0, 0x00, /* aac, pid=0x101=257 */
-    /*0x03, 0xe1, 0x01, 0xf0, 0x00,*/ /* mp3 */
     /* CRC */
     0x2f, 0x44, 0xb9, 0x9b, /* crc for aac */
-    /*0x4e, 0x59, 0x3d, 0x1e,*/ /* crc for mp3 */
+};
+u_int8_t mpegts_header_mp3[] = {
+    0x03, 0xe1, 0x01, 0xf0, 0x00, /* mp3 */
+    /* CRC */
+    0x4e, 0x59, 0x3d, 0x1e, /* crc for mp3 */
+};
+u_int8_t mpegts_header_padding[] = {
     /* stuffing 157 bytes */
     0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
     0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
@@ -147,7 +155,7 @@ u_int8_t mpegts_header[] = {
 class SrsMpegtsWriter
 {
 public:
-    static int write_header(SrsFileWriter* writer)
+    static int write_header(SrsFileWriter* writer, SrsCodecAudio acodec)
     {
         int ret = ERROR_SUCCESS;
         
@@ -157,6 +165,26 @@ public:
             return ret;
         }
 
+        if (acodec == SrsCodecAudioAAC) {
+            if ((ret = writer->write(mpegts_header_aac, sizeof(mpegts_header_aac), NULL)) != ERROR_SUCCESS) {
+                ret = ERROR_HLS_WRITE_FAILED;
+                srs_error("write ts file aac header failed. ret=%d", ret);
+                return ret;
+            }
+        } else {
+            if ((ret = writer->write(mpegts_header_mp3, sizeof(mpegts_header_mp3), NULL)) != ERROR_SUCCESS) {
+                ret = ERROR_HLS_WRITE_FAILED;
+                srs_error("write ts file mp3 header failed. ret=%d", ret);
+                return ret;
+            }
+        }
+        
+        if ((ret = writer->write(mpegts_header_padding, sizeof(mpegts_header_padding), NULL)) != ERROR_SUCCESS) {
+            ret = ERROR_HLS_WRITE_FAILED;
+            srs_error("write ts file padding header failed. ret=%d", ret);
+            return ret;
+        }
+
         return ret;
     }
     static int write_frame(SrsFileWriter* writer, SrsMpegtsFrame* frame, SrsSimpleBuffer* buffer)
@@ -375,6 +403,11 @@ SrsMpegtsFrame::SrsMpegtsFrame()
 SrsTSMuxer::SrsTSMuxer(SrsFileWriter* w)
 {
     writer = w;
+
+    // reserved is not written.
+    previous = SrsCodecAudioReserved1;
+    // current default to aac.
+    current = SrsCodecAudioAAC;
 }
 
 SrsTSMuxer::~SrsTSMuxer()
@@ -393,12 +426,19 @@ int SrsTSMuxer::open(string _path)
     if ((ret = writer->open(path)) != ERROR_SUCCESS) {
         return ret;
     }
+    
+    return ret;
+}
+
+int SrsTSMuxer::update_acodec(SrsCodecAudio acodec)
+{
+    int ret = ERROR_SUCCESS;
 
-    // write mpegts header
-    if ((ret = SrsMpegtsWriter::write_header(writer)) != ERROR_SUCCESS) {
+    if (current == acodec) {
         return ret;
     }
-    
+    current = acodec;
+
     return ret;
 }
 
@@ -406,6 +446,14 @@ int SrsTSMuxer::write_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab)
 {
     int ret = ERROR_SUCCESS;
     
+    // when acodec changed, write header.
+    if (current != previous) {
+        previous = current;
+        if ((ret = SrsMpegtsWriter::write_header(writer, previous)) != ERROR_SUCCESS) {
+            return ret;
+        }
+    }
+    
     if ((ret = SrsMpegtsWriter::write_frame(writer, af, ab)) != ERROR_SUCCESS) {
         return ret;
     }
@@ -417,6 +465,14 @@ int SrsTSMuxer::write_video(SrsMpegtsFrame* vf, SrsSimpleBuffer* vb)
 {
     int ret = ERROR_SUCCESS;
     
+    // when acodec changed, write header.
+    if (current != previous) {
+        previous = current;
+        if ((ret = SrsMpegtsWriter::write_header(writer, previous)) != ERROR_SUCCESS) {
+            return ret;
+        }
+    }
+    
     if ((ret = SrsMpegtsWriter::write_frame(writer, vf, vb)) != ERROR_SUCCESS) {
         return ret;
     }
@@ -501,6 +557,8 @@ SrsTsCache::SrsTsCache()
     
     af = new SrsMpegtsFrame();
     vf = new SrsMpegtsFrame();
+
+    audio_buffer_start_pts = 0;
 }
 
 SrsTsCache::~SrsTsCache()
@@ -520,23 +578,53 @@ SrsTsCache::~SrsTsCache()
 int SrsTsCache::cache_audio(SrsAvcAacCodec* codec, int64_t pts, SrsCodecSample* sample)
 {
     int ret = ERROR_SUCCESS;
-    
-    // start buffer, set the af
+
+    // @remark, always use the orignal pts.
     if (ab->length() == 0) {
-        pts = aac_jitter->on_buffer_start(pts, sample->sound_rate, codec->aac_sample_rate);
-        
-        af->dts = af->pts = pts;
-        af->pid = TS_AUDIO_PID;
-        af->sid = TS_AUDIO_AAC;
-    } else {
-        aac_jitter->on_buffer_continue();
+         audio_buffer_start_pts = pts;
     }
     
-    // write audio to cache.
-    if ((ret = do_cache_audio(codec, sample)) != ERROR_SUCCESS) {
+    // must be aac or mp3
+    SrsCodecAudio acodec = (SrsCodecAudio)codec->audio_codec_id;
+    srs_assert(acodec == SrsCodecAudioAAC || acodec == SrsCodecAudioMP3);
+    
+    // cache the aac audio.
+    if (codec->audio_codec_id == SrsCodecAudioAAC) {
+        // for aac audio, recalc the timestamp by aac jitter.
+        if (ab->length() == 0) {
+            pts = aac_jitter->on_buffer_start(pts, sample->sound_rate, codec->aac_sample_rate);
+        
+            af->dts = af->pts = pts;
+            af->pid = TS_AUDIO_PID;
+            af->sid = TS_AUDIO_AAC;
+        } else {
+            aac_jitter->on_buffer_continue();
+        }
+    
+        // write aac audio to cache.
+        if ((ret = do_cache_audio(codec, sample)) != ERROR_SUCCESS) {
+            return ret;
+        }
+
         return ret;
     }
     
+    // cache the mp3 audio.
+    if (codec->audio_codec_id == SrsCodecAudioMP3) {
+        // for mp3 audio, recalc the timestamp by mp3 jitter.
+        // TODO: FIXME: implements it.
+        af->dts = af->pts = pts;
+        af->pid = TS_AUDIO_PID;
+        af->sid = SrsCodecAudioMP3;
+        
+        // for mp3, directly write to cache.
+        // TODO: FIXME: implements it.
+        for (int i = 0; i < sample->nb_sample_units; i++) {
+            SrsCodecSampleUnit* sample_unit = &sample->sample_units[i];
+            ab->append(sample_unit->bytes, sample_unit->size);
+        }
+    }
+    
     return ret;
 }
     
@@ -784,16 +872,30 @@ int SrsTsEncoder::write_audio(int64_t timestamp, char* data, int size)
     
     sample->clear();
     if ((ret = codec->audio_aac_demux(data, size, sample)) != ERROR_SUCCESS) {
-        srs_error("http: ts codec demux audio failed. ret=%d", ret);
-        return ret;
+        if (ret != ERROR_HLS_TRY_MP3) {
+            srs_error("http: ts aac demux audio failed. ret=%d", ret);
+            return ret;
+        }
+        if ((ret = codec->audio_mp3_demux(data, size, sample)) != ERROR_SUCCESS) {
+            srs_error("http: ts mp3 demux audio failed. ret=%d", ret);
+            return ret;
+        }
     }
+    SrsCodecAudio acodec = (SrsCodecAudio)codec->audio_codec_id;
     
-    if (codec->audio_codec_id != SrsCodecAudioAAC) {
+    // ts support audio codec: aac/mp3
+    if (acodec != SrsCodecAudioAAC && acodec != SrsCodecAudioMP3) {
+        return ret;
+    }
+
+    // when codec changed, write new header.
+    if ((ret = muxer->update_acodec(acodec)) != ERROR_SUCCESS) {
+        srs_error("http: ts audio write header failed. ret=%d", ret);
         return ret;
     }
     
-    // ignore sequence header
-    if (sample->aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
+    // for aac: ignore sequence header
+    if (acodec == SrsCodecAudioAAC && sample->aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
         return ret;
     }
 
@@ -809,12 +911,15 @@ int SrsTsEncoder::write_audio(int64_t timestamp, char* data, int size)
     
     // flush if buffer exceed max size.
     if (cache->ab->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
-        if ((ret = muxer->write_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
-            return ret;
-        }
-    
-        // write success, clear and free the buffer
-        cache->ab->erase(cache->ab->length());
+        return flush_video();
+    }
+
+    // TODO: config it.
+    // in ms, audio delay to flush the audios.
+    int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
+    // flush if audio delay exceed
+    if (dts - cache->audio_buffer_start_pts > audio_delay * 90) {
+        return flush_audio();
     }
 
     return ret;
@@ -852,6 +957,27 @@ int SrsTsEncoder::write_video(int64_t timestamp, char* data, int size)
     if ((ret = cache->cache_video(codec, dts, sample)) != ERROR_SUCCESS) {
         return ret;
     }
+
+    return flush_video();
+}
+
+int SrsTsEncoder::flush_audio()
+{
+    int ret = ERROR_SUCCESS;
+
+    if ((ret = muxer->write_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
+        return ret;
+    }
+    
+    // write success, clear and free the buffer
+    cache->ab->erase(cache->ab->length());
+
+    return ret;
+}
+
+int SrsTsEncoder::flush_video()
+{
+    int ret = ERROR_SUCCESS;
     
     if ((ret = muxer->write_video(cache->vf, cache->vb)) != ERROR_SUCCESS) {
         return ret;

@@ -31,6 +31,8 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 
 #include <string>
 
+#include <srs_kernel_codec.hpp>
+
 class SrsTsCache;
 class SrsTSMuxer;
 class SrsFileWriter;
@@ -63,15 +65,40 @@ public:
 class SrsTSMuxer
 {
 private:
+    SrsCodecAudio previous;
+    SrsCodecAudio current;
+private:
     SrsFileWriter* writer;
     std::string path;
 public:
     SrsTSMuxer(SrsFileWriter* w);
     virtual ~SrsTSMuxer();
 public:
+    /**
+    * open the writer, donot write the PSI of ts.
+    */
     virtual int open(std::string _path);
+    /**
+    * when open ts, we donot write the header(PSI),
+    * for user may need to update the acodec to mp3 or others,
+    * so we use delay write PSI, when write audio or video.
+    * @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
+    */
+    virtual int update_acodec(SrsCodecAudio acodec);
+    /**
+    * write an audio frame to ts, 
+    * @remark write PSI first when not write yet.
+    */
     virtual int write_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab);
+    /**
+    * write a video frame to ts, 
+    * @remark write PSI first when not write yet.
+    */
     virtual int write_video(SrsMpegtsFrame* vf, SrsSimpleBuffer* vb);
+    /**
+    * close the writer.
+    */
     virtual void close();
 };
 
@@ -125,6 +152,10 @@ public:
     SrsSimpleBuffer* ab;
     SrsMpegtsFrame* vf;
     SrsSimpleBuffer* vb;
+public:
+    // the audio cache buffer start pts, to flush audio if full.
+    // @remark the pts is not the adjust one, it's the orignal pts.
+    int64_t audio_buffer_start_pts;
 protected:
     // time jitter for aac
     SrsTsAacJitter* aac_jitter;
@@ -172,6 +203,9 @@ public:
     */
     virtual int write_audio(int64_t timestamp, char* data, int size);
     virtual int write_video(int64_t timestamp, char* data, int size);
+private:
+    virtual int flush_audio();
+    virtual int flush_video();
 };
 
 #endif