diff --git a/trunk/src/app/srs_app_hls.cpp b/trunk/src/app/srs_app_hls.cpp
index 73cb1d57e..9e24f72e5 100644
--- a/trunk/src/app/srs_app_hls.cpp
+++ b/trunk/src/app/srs_app_hls.cpp
@@ -62,80 +62,14 @@ using namespace std;
 // reset the piece id when deviation overflow this.
 #define SRS_JUMP_WHEN_PIECE_DEVIATION 20
 
-SrsHlsCacheWriter::SrsHlsCacheWriter(bool write_cache, bool write_file)
-{
-    should_write_cache = write_cache;
-    should_write_file = write_file;
-}
-
-SrsHlsCacheWriter::~SrsHlsCacheWriter()
-{
-}
-
-int SrsHlsCacheWriter::open(string file)
-{
-    if (!should_write_file) {
-        return ERROR_SUCCESS;
-    }
-
-    return impl.open(file);
-}
-
-void SrsHlsCacheWriter::close()
-{
-    if (!should_write_file) {
-        return;
-    }
-
-    impl.close();
-}
-
-bool SrsHlsCacheWriter::is_open()
-{
-    if (!should_write_file) {
-        return true;
-    }
-
-    return impl.is_open();
-}
-
-int64_t SrsHlsCacheWriter::tellg()
-{
-    if (!should_write_file) {
-        return 0;
-    }
-
-    return impl.tellg();
-}
-
-int SrsHlsCacheWriter::write(void* buf, size_t count, ssize_t* pnwrite)
-{
-    if (should_write_cache) {
-        if (count > 0) {
-            data.append((char*)buf, count);
-        }
-    }
-
-    if (should_write_file) {
-        return impl.write(buf, count, pnwrite);
-    }
-
-    return ERROR_SUCCESS;
-}
-
-string SrsHlsCacheWriter::cache()
-{
-    return data;
-}
-
-SrsHlsSegment::SrsHlsSegment(SrsTsContext* c, bool write_cache, bool write_file, SrsCodecAudio ac, SrsCodecVideo vc)
+SrsHlsSegment::SrsHlsSegment(SrsTsContext* c, SrsCodecAudio ac, SrsCodecVideo vc)
 {
     duration = 0;
     sequence_no = 0;
     segment_start_dts = 0;
     is_sequence_header = false;
-    writer = new SrsHlsCacheWriter(write_cache, write_file);
-    muxer = new SrsTSMuxer(writer, c, ac, vc);
+    writer = new SrsFileWriter();
+    muxer = new SrsTsMuxer(writer, c, ac, vc);
 }
 
 SrsHlsSegment::~SrsHlsSegment()
@@ -290,8 +224,6 @@ SrsHlsMuxer::SrsHlsMuxer()
     _sequence_no = 0;
     current = NULL;
     acodec = SrsCodecAudioReserved1;
-    should_write_cache = false;
-    should_write_file = true;
     async = new SrsAsyncCallWorker();
     context = new SrsTsContext();
 }
@@ -313,28 +245,26 @@ SrsHlsMuxer::~SrsHlsMuxer()
 
 void SrsHlsMuxer::dispose()
 {
-    if (should_write_file) {
-        std::vector<SrsHlsSegment*>::iterator it;
-        for (it = segments.begin(); it != segments.end(); ++it) {
-            SrsHlsSegment* segment = *it;
-            if (unlink(segment->full_path.c_str()) < 0) {
-                srs_warn("dispose unlink path failed, file=%s.", segment->full_path.c_str());
-            }
-            srs_freep(segment);
+    std::vector<SrsHlsSegment*>::iterator it;
+    for (it = segments.begin(); it != segments.end(); ++it) {
+        SrsHlsSegment* segment = *it;
+        if (unlink(segment->full_path.c_str()) < 0) {
+            srs_warn("dispose unlink path failed, file=%s.", segment->full_path.c_str());
         }
-        segments.clear();
-        
-        if (current) {
-            std::string path = current->full_path + ".tmp";
-            if (unlink(path.c_str()) < 0) {
-                srs_warn("dispose unlink path failed, file=%s", path.c_str());
-            }
-            srs_freep(current);
-        }
-        
-        if (unlink(m3u8.c_str()) < 0) {
-            srs_warn("dispose unlink path failed. file=%s", m3u8.c_str());
+        srs_freep(segment);
+    }
+    segments.clear();
+    
+    if (current) {
+        std::string path = current->full_path + ".tmp";
+        if (unlink(path.c_str()) < 0) {
+            srs_warn("dispose unlink path failed, file=%s", path.c_str());
         }
+        srs_freep(current);
+    }
+    
+    if (unlink(m3u8.c_str()) < 0) {
+        srs_warn("dispose unlink path failed. file=%s", m3u8.c_str());
     }
     
     // TODO: FIXME: support hls dispose in HTTP cache.
@@ -407,13 +337,9 @@ int SrsHlsMuxer::update_config(SrsRequest* r, string entry_prefix,
     // when update config, reset the history target duration.
     max_td = (int)(fragment * _srs_config->get_hls_td_ratio(r->vhost));
     
-    // TODO: FIXME: refine better for SRS2 only support disk.
-    should_write_cache = false;
-    should_write_file = true;
-    
     // create m3u8 dir once.
     m3u8_dir = srs_path_dirname(m3u8);
-    if (should_write_file && (ret = srs_create_dir_recursively(m3u8_dir)) != ERROR_SUCCESS) {
+    if ((ret = srs_create_dir_recursively(m3u8_dir)) != ERROR_SUCCESS) {
         srs_error("create app dir %s failed. ret=%d", m3u8_dir.c_str(), ret);
         return ret;
     }
@@ -468,7 +394,7 @@ int SrsHlsMuxer::segment_open(int64_t segment_start_dts)
     }
     
     // new segment.
-    current = new SrsHlsSegment(context, should_write_cache, should_write_file, default_acodec, default_vcodec);
+    current = new SrsHlsSegment(context, default_acodec, default_vcodec);
     current->sequence_no = _sequence_no++;
     current->segment_start_dts = segment_start_dts;
     
@@ -540,7 +466,7 @@ int SrsHlsMuxer::segment_open(int64_t segment_start_dts)
     
     // create dir recursively for hls.
     std::string ts_dir = srs_path_dirname(current->full_path);
-    if (should_write_file && (ret = srs_create_dir_recursively(ts_dir)) != ERROR_SUCCESS) {
+    if ((ret = srs_create_dir_recursively(ts_dir)) != ERROR_SUCCESS) {
         srs_error("create app dir %s failed. ret=%d", ts_dir.c_str(), ret);
         return ret;
     }
@@ -735,7 +661,7 @@ int SrsHlsMuxer::segment_close(string log_desc)
 
         // rename from tmp to real path
         std::string tmp_file = full_path + ".tmp";
-        if (should_write_file && rename(tmp_file.c_str(), full_path.c_str()) < 0) {
+        if (rename(tmp_file.c_str(), full_path.c_str()) < 0) {
             ret = ERROR_HLS_WRITE_FAILED;
             srs_error("rename ts file failed, %s => %s. ret=%d", 
                 tmp_file.c_str(), full_path.c_str(), ret);
@@ -751,10 +677,8 @@ int SrsHlsMuxer::segment_close(string log_desc)
 
         // rename from tmp to real path
         std::string tmp_file = current->full_path + ".tmp";
-        if (should_write_file) {
-            if (unlink(tmp_file.c_str()) < 0) {
-                srs_warn("ignore unlink path failed, file=%s.", tmp_file.c_str());
-            }
+        if (unlink(tmp_file.c_str()) < 0) {
+            srs_warn("ignore unlink path failed, file=%s.", tmp_file.c_str());
         }
 
         srs_freep(current);
@@ -788,10 +712,8 @@ int SrsHlsMuxer::segment_close(string log_desc)
     for (int i = 0; i < (int)segment_to_remove.size(); i++) {
         SrsHlsSegment* segment = segment_to_remove[i];
         
-        if (hls_cleanup && should_write_file) {
-            if (unlink(segment->full_path.c_str()) < 0) {
-                srs_warn("cleanup unlink path failed, file=%s.", segment->full_path.c_str());
-            }
+        if (hls_cleanup && unlink(segment->full_path.c_str()) < 0) {
+            srs_warn("cleanup unlink path failed, file=%s.", segment->full_path.c_str());
         }
         
         srs_freep(segment);
@@ -818,7 +740,7 @@ int SrsHlsMuxer::refresh_m3u8()
     
     std::string temp_m3u8 = m3u8 + ".temp";
     if ((ret = _refresh_m3u8(temp_m3u8)) == ERROR_SUCCESS) {
-        if (should_write_file && rename(temp_m3u8.c_str(), m3u8.c_str()) < 0) {
+        if (rename(temp_m3u8.c_str(), m3u8.c_str()) < 0) {
             ret = ERROR_HLS_WRITE_FAILED;
             srs_error("rename m3u8 file failed. %s => %s, ret=%d", temp_m3u8.c_str(), m3u8.c_str(), ret);
         }
@@ -843,7 +765,7 @@ int SrsHlsMuxer::_refresh_m3u8(string m3u8_file)
         return ret;
     }
 
-    SrsHlsCacheWriter writer(should_write_cache, should_write_file);
+    SrsFileWriter writer;
     if ((ret = writer.open(m3u8_file)) != ERROR_SUCCESS) {
         srs_error("open m3u8 file %s failed. ret=%d", m3u8_file.c_str(), ret);
         return ret;
@@ -919,17 +841,54 @@ int SrsHlsMuxer::_refresh_m3u8(string m3u8_file)
     return ret;
 }
 
-SrsHlsCache::SrsHlsCache()
+SrsHlsController::SrsHlsController()
 {
-    cache = new SrsTsCache();
+    ts = new SrsTsCache();
+    muxer = new SrsHlsMuxer();
 }
 
-SrsHlsCache::~SrsHlsCache()
+SrsHlsController::~SrsHlsController()
 {
-    srs_freep(cache);
+    srs_freep(muxer);
+    srs_freep(ts);
 }
 
-int SrsHlsCache::on_publish(SrsHlsMuxer* muxer, SrsRequest* req, int64_t segment_start_dts)
+int SrsHlsController::initialize()
+{
+    return muxer->initialize();
+}
+
+void SrsHlsController::dispose()
+{
+    muxer->dispose();
+}
+
+int SrsHlsController::update_acodec(SrsCodecAudio ac)
+{
+    return muxer->update_acodec(ac);
+}
+
+int SrsHlsController::sequence_no()
+{
+    return muxer->sequence_no();
+}
+
+string SrsHlsController::ts_url()
+{
+    return muxer->ts_url();
+}
+
+double SrsHlsController::duration()
+{
+    return muxer->duration();
+}
+
+int SrsHlsController::deviation()
+{
+    return muxer->deviation();
+}
+
+int SrsHlsController::on_publish(SrsRequest* req, int64_t segment_start_dts)
 {
     int ret = ERROR_SUCCESS;
 
@@ -978,11 +937,11 @@ int SrsHlsCache::on_publish(SrsHlsMuxer* muxer, SrsRequest* req, int64_t segment
     return ret;
 }
 
-int SrsHlsCache::on_unpublish(SrsHlsMuxer* muxer)
+int SrsHlsController::on_unpublish()
 {
     int ret = ERROR_SUCCESS;
     
-    if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
+    if ((ret = muxer->flush_audio(ts)) != ERROR_SUCCESS) {
         srs_error("m3u8 muxer flush audio failed. ret=%d", ret);
         return ret;
     }
@@ -994,7 +953,7 @@ int SrsHlsCache::on_unpublish(SrsHlsMuxer* muxer)
     return ret;
 }
 
-int SrsHlsCache::on_sequence_header(SrsHlsMuxer* muxer)
+int SrsHlsController::on_sequence_header()
 {
     // TODO: support discontinuity for the same stream
     // currently we reap and insert discontinity when encoder republish,
@@ -1005,12 +964,12 @@ int SrsHlsCache::on_sequence_header(SrsHlsMuxer* muxer)
     return muxer->on_sequence_header();
 }
 
-int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t pts, SrsCodecSample* sample)
+int SrsHlsController::write_audio(SrsAudioFrame* frame, int64_t pts)
 {
     int ret = ERROR_SUCCESS;
     
     // write audio to cache.
-    if ((ret = cache->cache_audio(codec, pts, sample)) != ERROR_SUCCESS) {
+    if ((ret = ts->cache_audio(frame, pts)) != ERROR_SUCCESS) {
         return ret;
     }
     
@@ -1022,16 +981,16 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
     // @see https://github.com/ossrs/srs/issues/151
     // we use absolutely overflow of segment to make jwplayer/ffplay happy
     // @see https://github.com/ossrs/srs/issues/151#issuecomment-71155184
-    if (cache->audio && muxer->is_segment_absolutely_overflow()) {
+    if (ts->audio && muxer->is_segment_absolutely_overflow()) {
         srs_info("hls: absolute audio reap segment.");
-        if ((ret = reap_segment("audio", muxer, cache->audio->pts)) != ERROR_SUCCESS) {
+        if ((ret = reap_segment("audio", ts->audio->pts)) != ERROR_SUCCESS) {
             return ret;
         }
     }
     
     // for pure audio, aggregate some frame to one.
-    if (muxer->pure_audio() && cache->audio) {
-        if (pts - cache->audio->start_pts < SRS_CONSTS_HLS_PURE_AUDIO_AGGREGATE) {
+    if (muxer->pure_audio() &&ts->audio) {
+        if (pts - ts->audio->start_pts < SRS_CONSTS_HLS_PURE_AUDIO_AGGREGATE) {
             return ret;
         }
     }
@@ -1040,19 +999,19 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
     // it's ok for the hls overload, or maybe cause the audio corrupt,
     // which introduced by aggregate the audios to a big one.
     // @see https://github.com/ossrs/srs/issues/512
-    if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
+    if ((ret = muxer->flush_audio(ts)) != ERROR_SUCCESS) {
         return ret;
     }
     
     return ret;
 }
     
-int SrsHlsCache::write_video(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t dts, SrsCodecSample* sample)
+int SrsHlsController::write_video(SrsVideoFrame* frame, int64_t dts)
 {
     int ret = ERROR_SUCCESS;
     
     // write video to cache.
-    if ((ret = cache->cache_video(codec, dts, sample)) != ERROR_SUCCESS) {
+    if ((ret = ts->cache_video(frame, dts)) != ERROR_SUCCESS) {
         return ret;
     }
     
@@ -1061,16 +1020,16 @@ int SrsHlsCache::write_video(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
         // do reap ts if any of:
         //      a. wait keyframe and got keyframe.
         //      b. always reap when not wait keyframe.
-        if (!muxer->wait_keyframe() || sample->frame_type == SrsCodecVideoAVCFrameKeyFrame) {
+        if (!muxer->wait_keyframe() || frame->frame_type == SrsCodecVideoAVCFrameKeyFrame) {
             // reap the segment, which will also flush the video.
-            if ((ret = reap_segment("video", muxer, cache->video->dts)) != ERROR_SUCCESS) {
+            if ((ret = reap_segment("video", ts->video->dts)) != ERROR_SUCCESS) {
                 return ret;
             }
         }
     }
     
     // flush video when got one
-    if ((ret = muxer->flush_video(cache)) != ERROR_SUCCESS) {
+    if ((ret = muxer->flush_video(ts)) != ERROR_SUCCESS) {
         srs_error("m3u8 muxer flush video failed. ret=%d", ret);
         return ret;
     }
@@ -1078,7 +1037,7 @@ int SrsHlsCache::write_video(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
     return ret;
 }
 
-int SrsHlsCache::reap_segment(string log_desc, SrsHlsMuxer* muxer, int64_t segment_start_dts)
+int SrsHlsController::reap_segment(string log_desc, int64_t segment_start_dts)
 {
     int ret = ERROR_SUCCESS;
     
@@ -1098,7 +1057,7 @@ int SrsHlsCache::reap_segment(string log_desc, SrsHlsMuxer* muxer, int64_t segme
     }
     
     // segment open, flush video first.
-    if ((ret = muxer->flush_video(cache)) != ERROR_SUCCESS) {
+    if ((ret = muxer->flush_video(ts)) != ERROR_SUCCESS) {
         srs_error("m3u8 muxer flush video failed. ret=%d", ret);
         return ret;
     }
@@ -1106,7 +1065,7 @@ int SrsHlsCache::reap_segment(string log_desc, SrsHlsMuxer* muxer, int64_t segme
     // segment open, flush the audio.
     // @see: ngx_rtmp_hls_open_fragment
     /* start fragment with audio to make iPhone happy */
-    if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
+    if ((ret = muxer->flush_audio(ts)) != ERROR_SUCCESS) {
         srs_error("m3u8 muxer flush audio failed. ret=%d", ret);
         return ret;
     }
@@ -1125,9 +1084,7 @@ SrsHls::SrsHls()
     last_update_time = 0;
 
     jitter = new SrsRtmpJitter();
-    
-    muxer = new SrsHlsMuxer();
-    cache = new SrsHlsCache();
+    controller = new SrsHlsController();
 
     pprint = SrsPithyPrint::create_hls();
     stream_dts = 0;
@@ -1136,10 +1093,7 @@ SrsHls::SrsHls()
 SrsHls::~SrsHls()
 {
     srs_freep(jitter);
-    
-    srs_freep(muxer);
-    srs_freep(cache);
-    
+    srs_freep(controller);
     srs_freep(pprint);
 }
 
@@ -1149,7 +1103,7 @@ void SrsHls::dispose()
         on_unpublish();
     }
     
-    muxer->dispose();
+    controller->dispose();
 }
 
 int SrsHls::cycle()
@@ -1194,7 +1148,7 @@ int SrsHls::initialize(SrsOriginHub* h, SrsFormat* f, SrsRequest* r)
     req = r;
     format = f;
 
-    if ((ret = muxer->initialize()) != ERROR_SUCCESS) {
+    if ((ret = controller->initialize()) != ERROR_SUCCESS) {
         return ret;
     }
 
@@ -1217,7 +1171,7 @@ int SrsHls::on_publish()
         return ret;
     }
     
-    if ((ret = cache->on_publish(muxer, req, stream_dts)) != ERROR_SUCCESS) {
+    if ((ret = controller->on_publish(req, stream_dts)) != ERROR_SUCCESS) {
         return ret;
     }
     
@@ -1239,14 +1193,14 @@ void SrsHls::on_unpublish()
         return;
     }
 
-    if ((ret = cache->on_unpublish(muxer)) != ERROR_SUCCESS) {
+    if ((ret = controller->on_unpublish()) != ERROR_SUCCESS) {
         srs_error("ignore m3u8 muxer flush/close audio failed. ret=%d", ret);
     }
     
     enabled = false;
 }
 
-int SrsHls::on_audio(SrsSharedPtrMessage* shared_audio)
+int SrsHls::on_audio(SrsSharedPtrMessage* shared_audio, SrsFormat* format)
 {
     int ret = ERROR_SUCCESS;
     
@@ -1260,35 +1214,23 @@ int SrsHls::on_audio(SrsSharedPtrMessage* shared_audio)
     SrsSharedPtrMessage* audio = shared_audio->copy();
     SrsAutoFree(SrsSharedPtrMessage, audio);
     
-    sample->clear();
-    if ((ret = codec->audio_aac_demux(audio->payload, audio->size, sample)) != ERROR_SUCCESS) {
-        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;
-        }
-    }
-    srs_info("audio decoded, type=%d, codec=%d, cts=%d, size=%d, time=%"PRId64, 
-        sample->frame_type, codec->audio_codec_id, sample->cts, audio->size, audio->timestamp);
-    SrsCodecAudio acodec = (SrsCodecAudio)codec->audio_codec_id;
-    
     // ts support audio codec: aac/mp3
+    srs_assert(format->acodec);
+    SrsCodecAudio acodec = format->acodec->id;
     if (acodec != SrsCodecAudioAAC && acodec != SrsCodecAudioMP3) {
         return ret;
     }
 
     // when codec changed, write new header.
-    if ((ret = muxer->update_acodec(acodec)) != ERROR_SUCCESS) {
+    if ((ret = controller->update_acodec(acodec)) != ERROR_SUCCESS) {
         srs_error("http: ts audio write header failed. ret=%d", ret);
         return ret;
     }
     
     // ignore sequence header
-    if (acodec == SrsCodecAudioAAC && sample->aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
-        return cache->on_sequence_header(muxer);
+    srs_assert(format->audio);
+    if (acodec == SrsCodecAudioAAC && format->audio->aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
+        return controller->on_sequence_header();
     }
     
     // TODO: FIXME: config the jitter of HLS.
@@ -1303,7 +1245,7 @@ int SrsHls::on_audio(SrsSharedPtrMessage* shared_audio)
     // for pure audio, we need to update the stream dts also.
     stream_dts = dts;
     
-    if ((ret = cache->write_audio(codec, muxer, dts, sample)) != ERROR_SUCCESS) {
+    if ((ret = controller->write_audio(format->audio, dts)) != ERROR_SUCCESS) {
         srs_error("hls cache write audio failed. ret=%d", ret);
         return ret;
     }
@@ -1311,7 +1253,7 @@ int SrsHls::on_audio(SrsSharedPtrMessage* shared_audio)
     return ret;
 }
 
-int SrsHls::on_video(SrsSharedPtrMessage* shared_video, bool is_sps_pps)
+int SrsHls::on_video(SrsSharedPtrMessage* shared_video, SrsFormat* format)
 {
     int ret = ERROR_SUCCESS;
     
@@ -1325,34 +1267,21 @@ int SrsHls::on_video(SrsSharedPtrMessage* shared_video, bool is_sps_pps)
     SrsSharedPtrMessage* video = shared_video->copy();
     SrsAutoFree(SrsSharedPtrMessage, video);
     
-    // user can disable the sps parse to workaround when parse sps failed.
-    // @see https://github.com/ossrs/srs/issues/474
-    if (is_sps_pps) {
-        codec->avc_parse_sps = _srs_config->get_parse_sps(req->vhost);
-    }
-    
-    sample->clear();
-    if ((ret = codec->video_avc_demux(video->payload, video->size, sample)) != ERROR_SUCCESS) {
-        srs_error("hls codec demux video failed. ret=%d", ret);
-        return ret;
-    }
-    srs_info("video decoded, type=%d, codec=%d, avc=%d, cts=%d, size=%d, time=%"PRId64, 
-        sample->frame_type, codec->video_codec_id, sample->avc_packet_type, sample->cts, video->size, video->timestamp);
-    
     // ignore info frame,
     // @see https://github.com/ossrs/srs/issues/288#issuecomment-69863909
-    if (sample->frame_type == SrsCodecVideoAVCFrameVideoInfoFrame) {
+    srs_assert(format->video);
+    if (format->video->frame_type == SrsCodecVideoAVCFrameVideoInfoFrame) {
         return ret;
     }
     
-    if (codec->video_codec_id != SrsCodecVideoAVC) {
+    srs_assert(format->vcodec);
+    if (format->vcodec->id != SrsCodecVideoAVC) {
         return ret;
     }
     
     // ignore sequence header
-    if (sample->frame_type == SrsCodecVideoAVCFrameKeyFrame
-         && sample->avc_packet_type == SrsCodecVideoAVCTypeSequenceHeader) {
-        return cache->on_sequence_header(muxer);
+    if (format->video->avc_packet_type == SrsCodecVideoAVCTypeSequenceHeader) {
+        return controller->on_sequence_header();
     }
     
     // TODO: FIXME: config the jitter of HLS.
@@ -1363,7 +1292,7 @@ int SrsHls::on_video(SrsSharedPtrMessage* shared_video, bool is_sps_pps)
     
     int64_t dts = video->timestamp * 90;
     stream_dts = dts;
-    if ((ret = cache->write_video(codec, muxer, dts, sample)) != ERROR_SUCCESS) {
+    if ((ret = controller->write_video(format->video, dts)) != ERROR_SUCCESS) {
         srs_error("hls cache write video failed. ret=%d", ret);
         return ret;
     }
@@ -1378,15 +1307,16 @@ void SrsHls::hls_show_mux_log()
 {
     pprint->elapse();
 
-    // reportable
-    if (pprint->can_print()) {
-        // the run time is not equals to stream time,
-        // @see: https://github.com/ossrs/srs/issues/81#issuecomment-48100994
-        // it's ok.
-        srs_trace("-> "SRS_CONSTS_LOG_HLS" time=%"PRId64", stream dts=%"PRId64"(%"PRId64"ms), sno=%d, ts=%s, dur=%.2f, dva=%dp",
-            pprint->age(), stream_dts, stream_dts / 90, muxer->sequence_no(), muxer->ts_url().c_str(),
-            muxer->duration(), muxer->deviation());
+    if (!pprint->can_print()) {
+        return;
     }
+    
+    // the run time is not equals to stream time,
+    // @see: https://github.com/ossrs/srs/issues/81#issuecomment-48100994
+    // it's ok.
+    srs_trace("-> "SRS_CONSTS_LOG_HLS" time=%"PRId64", stream dts=%"PRId64"(%"PRId64"ms), sno=%d, ts=%s, dur=%.2f, dva=%dp",
+        pprint->age(), stream_dts, stream_dts / 90, controller->sequence_no(), controller->ts_url().c_str(),
+        controller->duration(), controller->deviation());
 }
 
 
diff --git a/trunk/src/app/srs_app_hls.hpp b/trunk/src/app/srs_app_hls.hpp
index ec50c1523..faf4af558 100644
--- a/trunk/src/app/srs_app_hls.hpp
+++ b/trunk/src/app/srs_app_hls.hpp
@@ -40,7 +40,7 @@ class SrsFormat;
 class SrsSharedPtrMessage;
 class SrsAmf0Object;
 class SrsRtmpJitter;
-class SrsTSMuxer;
+class SrsTsMuxer;
 class SrsRequest;
 class SrsPithyPrint;
 class SrsSource;
@@ -53,41 +53,6 @@ class SrsHlsSegment;
 class SrsTsCache;
 class SrsTsContext;
 
-/**
-* write to file and cache.
-*/
-class SrsHlsCacheWriter : public SrsFileWriter
-{
-private:
-    SrsFileWriter impl;
-    std::string data;
-    bool should_write_cache;
-    bool should_write_file;
-public:
-    SrsHlsCacheWriter(bool write_cache, bool write_file);
-    virtual ~SrsHlsCacheWriter();
-public:
-    /**
-    * open file writer, can open then close then open...
-    */
-    virtual int open(std::string file);
-    virtual void close();
-public:
-    virtual bool is_open();
-    virtual int64_t tellg();
-public:
-    /**
-    * write to file. 
-    * @param pnwrite the output nb_write, NULL to ignore.
-    */
-    virtual int write(void* buf, size_t count, ssize_t* pnwrite);
-public:
-    /**
-    * get the string cache.
-    */
-    virtual std::string cache();
-};
-
 /**
 * the wrapper of m3u8 segment from specification:
 *
@@ -106,14 +71,14 @@ public:
     // ts full file to write.
     std::string full_path;
     // the muxer to write ts.
-    SrsHlsCacheWriter* writer;
-    SrsTSMuxer* muxer;
+    SrsFileWriter* writer;
+    SrsTsMuxer* muxer;
     // current segment start dts for m3u8
     int64_t segment_start_dts;
     // whether current segement is sequence header.
     bool is_sequence_header;
 public:
-    SrsHlsSegment(SrsTsContext* c, bool write_cache, bool write_file, SrsCodecAudio ac, SrsCodecVideo vc);
+    SrsHlsSegment(SrsTsContext* c, SrsCodecAudio ac, SrsCodecVideo vc);
     virtual ~SrsHlsSegment();
 public:
     /**
@@ -200,10 +165,6 @@ private:
     int max_td;
     std::string m3u8;
     std::string m3u8_url;
-private:
-    // TODO: FIXME: remove it.
-    bool should_write_cache;
-    bool should_write_file;
 private:
     /**
     * m3u8 segments.
@@ -303,34 +264,46 @@ private:
 *   so we must gather audio frame together, and recalc the timestamp @see SrsTsAacJitter,
 *   we use a aac jitter to correct the audio pts.
 */
-class SrsHlsCache
+class SrsHlsController
 {
 private:
-    SrsTsCache* cache;
+    // The HLS muxer to reap ts and m3u8.
+    // The TS is cached to SrsTsCache then flush to ts segment.
+    SrsHlsMuxer* muxer;
+    // The TS cache
+    SrsTsCache* ts;
 public:
-    SrsHlsCache();
-    virtual ~SrsHlsCache();
+    SrsHlsController();
+    virtual ~SrsHlsController();
+public:
+    virtual int initialize();
+    virtual void dispose();
+    virtual int update_acodec(SrsCodecAudio ac);
+    virtual int sequence_no();
+    virtual std::string ts_url();
+    virtual double duration();
+    virtual int deviation();
 public:
     /**
     * when publish or unpublish stream.
     */
-    virtual int on_publish(SrsHlsMuxer* muxer, SrsRequest* req, int64_t segment_start_dts);
-    virtual int on_unpublish(SrsHlsMuxer* muxer);
+    virtual int on_publish(SrsRequest* req, int64_t segment_start_dts);
+    virtual int on_unpublish();
     /**
     * when get sequence header, 
     * must write a #EXT-X-DISCONTINUITY to m3u8.
     * @see: hls-m3u8-draft-pantos-http-live-streaming-12.txt
     * @see: 3.4.11.  EXT-X-DISCONTINUITY
     */
-    virtual int on_sequence_header(SrsHlsMuxer* muxer);
+    virtual int on_sequence_header();
     /**
     * write audio to cache, if need to flush, flush to muxer.
     */
-    virtual int write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t pts, SrsCodecSample* sample);
+    virtual int write_audio(SrsAudioFrame* frame, int64_t pts);
     /**
     * write video to muxer.
     */
-    virtual int write_video(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t dts, SrsCodecSample* sample);
+    virtual int write_video(SrsVideoFrame* frame, int64_t dts);
 private:
     /**
     * reopen the muxer for a new hls segment,
@@ -338,7 +311,7 @@ private:
     * then write the key frame to the new segment.
     * so, user must reap_segment then flush_video to hls muxer.
     */
-    virtual int reap_segment(std::string log_desc, SrsHlsMuxer* muxer, int64_t segment_start_dts);
+    virtual int reap_segment(std::string log_desc, int64_t segment_start_dts);
 };
 
 /**
@@ -348,8 +321,7 @@ private:
 class SrsHls
 {
 private:
-    SrsHlsMuxer* muxer;
-    SrsHlsCache* cache;
+    SrsHlsController* controller;
 private:
     SrsRequest* req;
     bool enabled;
@@ -400,13 +372,14 @@ public:
     * mux the audio packets to ts.
     * @param shared_audio, directly ptr, copy it if need to save it.
     */
-    virtual int on_audio(SrsSharedPtrMessage* shared_audio);
+    virtual int on_audio(SrsSharedPtrMessage* shared_audio, SrsFormat* format);
     /**
      * mux the video packets to ts.
      * @param shared_video, directly ptr, copy it if need to save it.
      * @param is_sps_pps whether the video is h.264 sps/pps.
      */
-    virtual int on_video(SrsSharedPtrMessage* shared_video, bool is_sps_pps);
+    // TODO: FIXME: Remove param is_sps_pps.
+    virtual int on_video(SrsSharedPtrMessage* shared_video, SrsFormat* format);
 private:
     virtual void hls_show_mux_log();
 };
diff --git a/trunk/src/app/srs_app_http_stream.cpp b/trunk/src/app/srs_app_http_stream.cpp
index d863b194b..ac4d59074 100755
--- a/trunk/src/app/srs_app_http_stream.cpp
+++ b/trunk/src/app/srs_app_http_stream.cpp
@@ -653,75 +653,6 @@ bool SrsLiveEntry::is_mp3()
     return _is_mp3;
 }
 
-SrsHlsM3u8Stream::SrsHlsM3u8Stream()
-{
-}
-
-SrsHlsM3u8Stream::~SrsHlsM3u8Stream()
-{
-}
-
-void SrsHlsM3u8Stream::set_m3u8(std::string v)
-{
-    m3u8 = v;
-}
-
-int SrsHlsM3u8Stream::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r)
-{
-    int ret = ERROR_SUCCESS;
-    
-    std::string data = m3u8;
-    
-    w->header()->set_content_length((int)data.length());
-    w->header()->set_content_type("application/x-mpegURL;charset=utf-8");
-
-    if ((ret = w->write((char*)data.data(), (int)data.length())) != ERROR_SUCCESS) {
-        if (!srs_is_client_gracefully_close(ret)) {
-            srs_error("send m3u8 failed. ret=%d", ret);
-        }
-        return ret;
-    }
-
-    return ret;
-}
-
-SrsHlsTsStream::SrsHlsTsStream()
-{
-}
-
-SrsHlsTsStream::~SrsHlsTsStream()
-{
-}
-
-void SrsHlsTsStream::set_ts(std::string v)
-{
-    ts = v;
-}
-
-int SrsHlsTsStream::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r)
-{
-    int ret = ERROR_SUCCESS;
-    
-    std::string data = ts;
-    
-    w->header()->set_content_length((int)data.length());
-    w->header()->set_content_type("video/MP2T");
-
-    if ((ret = w->write((char*)data.data(), (int)data.length())) != ERROR_SUCCESS) {
-        if (!srs_is_client_gracefully_close(ret)) {
-            srs_error("send ts failed. ret=%d", ret);
-        }
-        return ret;
-    }
-
-    return ret;
-}
-
-SrsHlsEntry::SrsHlsEntry()
-{
-    tmpl = NULL;
-}
-
 SrsHttpStreamServer::SrsHttpStreamServer(SrsServer* svr)
 {
     server = svr;
diff --git a/trunk/src/app/srs_app_http_stream.hpp b/trunk/src/app/srs_app_http_stream.hpp
index 23add75b6..0da29e907 100755
--- a/trunk/src/app/srs_app_http_stream.hpp
+++ b/trunk/src/app/srs_app_http_stream.hpp
@@ -262,58 +262,6 @@ public:
     bool is_aac();
 };
 
-/**
-* the m3u8 stream handler.
-*/
-class SrsHlsM3u8Stream : public ISrsHttpHandler
-{
-private:
-    std::string m3u8;
-public:
-    SrsHlsM3u8Stream();
-    virtual ~SrsHlsM3u8Stream();
-public:
-    virtual void set_m3u8(std::string v);
-public:
-    virtual int serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r);
-};
-
-/**
-* the ts stream handler.
-*/
-class SrsHlsTsStream : public ISrsHttpHandler
-{
-private:
-    std::string ts;
-public:
-    SrsHlsTsStream();
-    virtual ~SrsHlsTsStream();
-public:
-    virtual void set_ts(std::string v);
-public:
-    virtual int serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r);
-};
-
-/**
-* the srs hls entry.
-*/
-// TODO: FIXME: use hte hls template and entry.
-struct SrsHlsEntry
-{
-    // for template, the mount contains variables.
-    // for concrete stream, the mount is url to access.
-    std::string mount;
-    
-    // the template to create the entry
-    SrsHlsEntry* tmpl;
-
-    // key: the m3u8/ts file path.
-    // value: the http handler.
-    std::map<std::string, ISrsHttpHandler*> streams;
-    
-    SrsHlsEntry();
-};
-
 /**
 * the http stream server instance,
 * serve http stream, for example, flv/ts/mp3/aac live stream.
diff --git a/trunk/src/app/srs_app_rtsp.cpp b/trunk/src/app/srs_app_rtsp.cpp
index 6d3392ad0..7f6c0cd2d 100644
--- a/trunk/src/app/srs_app_rtsp.cpp
+++ b/trunk/src/app/srs_app_rtsp.cpp
@@ -44,6 +44,7 @@ using namespace std;
 #include <srs_app_pithy_print.hpp>
 #include <srs_app_rtmp_conn.hpp>
 #include <srs_protocol_utility.hpp>
+#include <srs_protocol_format.hpp>
 
 #ifdef SRS_AUTO_STREAM_CASTER
 
@@ -135,13 +136,13 @@ int SrsRtpConn::on_udp_packet(sockaddr_in* from, char* buf, int nb_buf)
 SrsRtspAudioCache::SrsRtspAudioCache()
 {
     dts = 0;
-    audio_samples = NULL;
+    audio = NULL;
     payload = NULL;
 }
 
 SrsRtspAudioCache::~SrsRtspAudioCache()
 {
-    srs_freep(audio_samples);
+    srs_freep(audio);
     srs_freep(payload);
 }
 
@@ -456,10 +457,10 @@ int SrsRtspConn::on_rtp_audio(SrsRtpPacket* pkt, int64_t dts)
 
     // cache current audio to kickoff.
     acache->dts = dts;
-    acache->audio_samples = pkt->audio_samples;
+    acache->audio = pkt->audio;
     acache->payload = pkt->payload;
 
-    pkt->audio_samples = NULL;
+    pkt->audio = NULL;
     pkt->payload = NULL;
 
     return ret;
@@ -474,11 +475,11 @@ int SrsRtspConn::kickoff_audio_cache(SrsRtpPacket* pkt, int64_t dts)
         return ret;
     }
 
-    if (dts - acache->dts > 0 && acache->audio_samples->nb_sample_units > 0) {
-        int64_t delta = (dts - acache->dts) / acache->audio_samples->nb_sample_units;
-        for (int i = 0; i < acache->audio_samples->nb_sample_units; i++) {
-            char* frame = acache->audio_samples->sample_units[i].bytes;
-            int nb_frame = acache->audio_samples->sample_units[i].size;
+    if (dts - acache->dts > 0 && acache->audio->nb_samples > 0) {
+        int64_t delta = (dts - acache->dts) / acache->audio->nb_samples;
+        for (int i = 0; i < acache->audio->nb_samples; i++) {
+            char* frame = acache->audio->samples[i].bytes;
+            int nb_frame = acache->audio->samples[i].size;
             int64_t timestamp = (acache->dts + delta * i) / 90;
             acodec->aac_packet_type = 1;
             if ((ret = write_audio_raw_frame(frame, nb_frame, acodec, (uint32_t)timestamp)) != ERROR_SUCCESS) {
@@ -488,7 +489,7 @@ int SrsRtspConn::kickoff_audio_cache(SrsRtpPacket* pkt, int64_t dts)
     }
 
     acache->dts = 0;
-    srs_freep(acache->audio_samples);
+    srs_freep(acache->audio);
     srs_freep(acache->payload);
 
     return ret;
@@ -510,13 +511,17 @@ int SrsRtspConn::write_sequence_header()
     if (true) {
         std::string sh = aac_specific_config;
 
-        SrsAvcAacCodec dec;
-        if ((ret = dec.audio_aac_sequence_header_demux((char*)sh.c_str(), (int)sh.length())) != ERROR_SUCCESS) {
+        SrsFormat* format = new SrsFormat();
+        SrsAutoFree(SrsFormat, format);
+        
+        if ((ret = format->on_aac_sequence_header((char*)sh.c_str(), (int)sh.length())) != ERROR_SUCCESS) {
             return ret;
         }
+        
+        SrsAudioCodec* dec = format->acodec;
 
         acodec->sound_format = SrsCodecAudioAAC;
-        acodec->sound_type = (dec.aac_channels == 2)? SrsCodecAudioSoundTypeStereo : SrsCodecAudioSoundTypeMono;
+        acodec->sound_type = (dec->aac_channels == 2)? SrsCodecAudioSoundTypeStereo : SrsCodecAudioSoundTypeMono;
         acodec->sound_size = SrsCodecAudioSampleSize16bit;
         acodec->aac_packet_type = 0;
 
@@ -526,7 +531,7 @@ int SrsRtspConn::write_sequence_header()
             16000, 12000, 11025,  8000,
             7350,     0,     0,    0
         };
-        switch (aac_sample_rates[dec.aac_sample_rate]) {
+        switch (aac_sample_rates[dec->aac_sample_rate]) {
             case 11025:
                 acodec->sound_rate = SrsCodecAudioSampleRate11025;
                 break;
diff --git a/trunk/src/app/srs_app_rtsp.hpp b/trunk/src/app/srs_app_rtsp.hpp
index 3ca34ad57..4a9d3314b 100644
--- a/trunk/src/app/srs_app_rtsp.hpp
+++ b/trunk/src/app/srs_app_rtsp.hpp
@@ -53,7 +53,7 @@ class SrsRawH264Stream;
 class SrsRawAacStream;
 struct SrsRawAacStreamCodec;
 class SrsSharedPtrMessage;
-class SrsCodecSample;
+class SrsAudioFrame;
 class SrsSimpleStream;
 class SrsPithyPrint;
 class SrsSimpleRtmpClient;
@@ -87,7 +87,7 @@ public:
 struct SrsRtspAudioCache
 {
     int64_t dts;
-    SrsCodecSample* audio_samples;
+    SrsAudioFrame* audio;
     SrsSimpleStream* payload;
 
     SrsRtspAudioCache();
diff --git a/trunk/src/app/srs_app_source.cpp b/trunk/src/app/srs_app_source.cpp
index 3b4b1fd91..8b50a510d 100755
--- a/trunk/src/app/srs_app_source.cpp
+++ b/trunk/src/app/srs_app_source.cpp
@@ -854,7 +854,7 @@ SrsOriginHub::SrsOriginHub()
     hds = new SrsHds();
 #endif
     ng_exec = new SrsNgExec();
-    format = new SrsFormat();
+    format = new SrsRtmpFormat();
     
     _srs_config->subscribe(this);
 }
@@ -971,7 +971,29 @@ int SrsOriginHub::on_audio(SrsSharedPtrMessage* shared_audio)
         return ret;
     }
     
-    if ((ret = hls->on_audio(msg)) != ERROR_SUCCESS) {
+    // cache the sequence header if aac
+    // donot cache the sequence header to gop_cache, return here.
+    if (format->is_aac_sequence_header()) {
+        srs_assert(format->acodec);
+        SrsAudioCodec* c = format->acodec;
+        
+        static int flv_sample_sizes[] = {8, 16, 0};
+        static int flv_sound_types[] = {1, 2, 0};
+        
+        // when got audio stream info.
+        SrsStatistic* stat = SrsStatistic::instance();
+        if ((ret = stat->on_audio_info(req, SrsCodecAudioAAC, c->sound_rate, c->sound_type, c->aac_object)) != ERROR_SUCCESS) {
+            return ret;
+        }
+        
+        srs_trace("%dB audio sh, codec(%d, profile=%s, %dchannels, %dkbps, %dHZ), flv(%dbits, %dchannels, %dHZ)",
+            msg->size, c->id, srs_codec_aac_object2str(c->aac_object).c_str(), c->aac_channels,
+            c->audio_data_rate / 1000, aac_sample_rates[c->aac_sample_rate],
+            flv_sample_sizes[c->sound_size], flv_sound_types[c->sound_type],
+            flv_sample_rates[c->sound_rate]);
+    }
+    
+    if ((ret = hls->on_audio(msg, format)) != ERROR_SUCCESS) {
         // apply the error strategy for hls.
         // @see https://github.com/ossrs/srs/issues/264
         std::string hls_error_strategy = _srs_config->get_hls_on_error(req->vhost);
@@ -1044,12 +1066,36 @@ int SrsOriginHub::on_video(SrsSharedPtrMessage* shared_video, bool is_sequence_h
     
     SrsSharedPtrMessage* msg = shared_video;
     
-    if ((ret = format->on_video(msg, is_sequence_header)) != ERROR_SUCCESS) {
+    // user can disable the sps parse to workaround when parse sps failed.
+    // @see https://github.com/ossrs/srs/issues/474
+    if (is_sequence_header) {
+        format->avc_parse_sps = _srs_config->get_parse_sps(req->vhost);
+    }
+    
+    if ((ret = format->on_video(msg)) != ERROR_SUCCESS) {
         srs_error("Codec parse video failed, ret=%d", ret);
         return ret;
     }
     
-    if ((ret = hls->on_video(msg, is_sequence_header)) != ERROR_SUCCESS) {
+    // cache the sequence header if h264
+    // donot cache the sequence header to gop_cache, return here.
+    if (format->is_avc_sequence_header()) {
+        SrsVideoCodec* c = format->vcodec;
+        srs_assert(c);
+        
+        // when got video stream info.
+        SrsStatistic* stat = SrsStatistic::instance();
+        if ((ret = stat->on_video_info(req, SrsCodecVideoAVC, c->avc_profile, c->avc_level, c->width, c->height)) != ERROR_SUCCESS) {
+            return ret;
+        }
+        
+        srs_trace("%dB video sh,  codec(%d, profile=%s, level=%s, %dx%d, %dkbps, %.1ffps, %.1fs)",
+            msg->size, c->id, srs_codec_avc_profile2str(c->avc_profile).c_str(),
+            srs_codec_avc_level2str(c->avc_level).c_str(), c->width, c->height,
+            c->video_data_rate / 1000, c->frame_rate, c->duration);
+    }
+    
+    if ((ret = hls->on_video(msg, format)) != ERROR_SUCCESS) {
         // apply the error strategy for hls.
         // @see https://github.com/ossrs/srs/issues/264
         std::string hls_error_strategy = _srs_config->get_hls_on_error(req->vhost);
@@ -1341,11 +1387,11 @@ int SrsOriginHub::on_reload_vhost_hls(string vhost)
     // when reload to start hls, hls will never get the sequence header in stream,
     // use the SrsSource.on_hls_start to push the sequence header to HLS.
     // TODO: maybe need to decode the metadata?
-    if (cache_sh_video && (ret = hls->on_video(cache_sh_video, true)) != ERROR_SUCCESS) {
+    if (cache_sh_video && (ret = hls->on_video(cache_sh_video, format)) != ERROR_SUCCESS) {
         srs_error("hls process video sequence header message failed. ret=%d", ret);
         return ret;
     }
-    if (cache_sh_audio && (ret = hls->on_audio(cache_sh_audio)) != ERROR_SUCCESS) {
+    if (cache_sh_audio && (ret = hls->on_audio(cache_sh_audio, format)) != ERROR_SUCCESS) {
         srs_error("hls process audio sequence header message failed. ret=%d", ret);
         return ret;
     }
@@ -2137,35 +2183,6 @@ int SrsSource::on_audio_imp(SrsSharedPtrMessage* msg)
         }
     }
     
-    // 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;
-        if ((ret = codec.audio_aac_demux(msg->payload, msg->size, &sample)) != ERROR_SUCCESS) {
-            srs_error("source codec demux audio failed. ret=%d", ret);
-            return ret;
-        }
-        
-        static int flv_sample_sizes[] = {8, 16, 0};
-        static int flv_sound_types[] = {1, 2, 0};
-        
-        // when got audio stream info.
-        SrsStatistic* stat = SrsStatistic::instance();
-        if ((ret = stat->on_audio_info(req, SrsCodecAudioAAC, sample.sound_rate, sample.sound_type, codec.aac_object)) != ERROR_SUCCESS) {
-            return ret;
-        }
-        
-        srs_trace("%dB audio sh, codec(%d, profile=%s, %dchannels, %dkbps, %dHZ), "
-            "flv(%dbits, %dchannels, %dHZ)",
-            msg->size, codec.audio_codec_id,
-            srs_codec_aac_object2str(codec.aac_object).c_str(), codec.aac_channels,
-            codec.audio_data_rate / 1000, aac_sample_rates[codec.aac_sample_rate],
-            flv_sample_sizes[sample.sound_size], flv_sound_types[sample.sound_type],
-            flv_sample_rates[sample.sound_rate]);
-    }
-    
     // copy to all consumer
     if (!drop_for_reduce) {
         for (int i = 0; i < (int)consumers.size(); i++) {
@@ -2296,31 +2313,6 @@ int SrsSource::on_video_imp(SrsSharedPtrMessage* msg)
     // donot cache the sequence header to gop_cache, return here.
     if (is_sequence_header) {
         meta->update_vsh(msg);
-        
-        // parse detail audio codec
-        SrsAvcAacCodec codec;
-        
-        // user can disable the sps parse to workaround when parse sps failed.
-        // @see https://github.com/ossrs/srs/issues/474
-        codec.avc_parse_sps = _srs_config->get_parse_sps(req->vhost);
-        
-        SrsCodecSample sample;
-        if ((ret = codec.video_avc_demux(msg->payload, msg->size, &sample)) != ERROR_SUCCESS) {
-            srs_error("source codec demux video failed. ret=%d", ret);
-            return ret;
-        }
-        
-        // when got video stream info.
-        SrsStatistic* stat = SrsStatistic::instance();
-        if ((ret = stat->on_video_info(req, SrsCodecVideoAVC, codec.avc_profile, codec.avc_level, codec.width, codec.height)) != ERROR_SUCCESS) {
-            return ret;
-        }
-        
-        srs_trace("%dB video sh,  codec(%d, profile=%s, level=%s, %dx%d, %dkbps, %dfps, %ds)",
-            msg->size, codec.video_codec_id,
-            srs_codec_avc_profile2str(codec.avc_profile).c_str(),
-            srs_codec_avc_level2str(codec.avc_level).c_str(), codec.width, codec.height,
-            codec.video_data_rate / 1000, codec.frame_rate, codec.duration);
     }
     
     // Copy to hub to all utilities.
diff --git a/trunk/src/app/srs_app_source.hpp b/trunk/src/app/srs_app_source.hpp
index 9d7d09678..4786337b2 100644
--- a/trunk/src/app/srs_app_source.hpp
+++ b/trunk/src/app/srs_app_source.hpp
@@ -38,7 +38,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #include <srs_app_reload.hpp>
 #include <srs_core_performance.hpp>
 
-class SrsFormat;
+class SrsRtmpFormat;
 class SrsConsumer;
 class SrsPlayEdge;
 class SrsPublishEdge;
@@ -423,7 +423,7 @@ private:
     bool is_active;
 private:
     // The format, codec information.
-    SrsFormat* format;
+    SrsRtmpFormat* format;
     // hls handler.
     SrsHls* hls;
     // The DASH encoder.
diff --git a/trunk/src/kernel/srs_kernel_aac.cpp b/trunk/src/kernel/srs_kernel_aac.cpp
index 87d061aee..530918013 100644
--- a/trunk/src/kernel/srs_kernel_aac.cpp
+++ b/trunk/src/kernel/srs_kernel_aac.cpp
@@ -93,7 +93,6 @@ int SrsAacEncoder::write_audio(int64_t timestamp, char* data, int size)
     // @see: E.4.2 Audio Tags, video_file_format_spec_v10_1.pdf, page 76
     int8_t sound_format = stream->read_1bytes();
     
-    // @see: SrsAvcAacCodec::audio_aac_demux
     //int8_t sound_type = sound_format & 0x01;
     //int8_t sound_size = (sound_format >> 1) & 0x01;
     //int8_t sound_rate = (sound_format >> 2) & 0x03;
diff --git a/trunk/src/kernel/srs_kernel_codec.cpp b/trunk/src/kernel/srs_kernel_codec.cpp
index 4cd4383a0..25dada8bd 100644
--- a/trunk/src/kernel/srs_kernel_codec.cpp
+++ b/trunk/src/kernel/srs_kernel_codec.cpp
@@ -343,8 +343,8 @@ string srs_codec_avc_level2str(SrsAvcLevel level)
 
 SrsSample::SrsSample()
 {
-    nb_unit = 0;
-    unit = NULL;
+    size = 0;
+    bytes = NULL;
 }
 
 SrsSample::~SrsSample()
@@ -361,43 +361,70 @@ SrsCodec::~SrsCodec()
 
 SrsAudioCodec::SrsAudioCodec()
 {
-    acodec = SrsCodecAudioForbidden;
+    id = SrsCodecAudioForbidden;
     sound_rate = SrsCodecAudioSampleRateForbidden;
     sound_size = SrsCodecAudioSampleSizeForbidden;
     sound_type = SrsCodecAudioSoundTypeForbidden;
-    aac_packet_type = SrsCodecAudioTypeForbidden;
+    
+    audio_data_rate = 0;
+    
+    aac_object = SrsAacObjectTypeForbidden;
+    aac_sample_rate = SRS_AAC_SAMPLE_RATE_UNSET; // sample rate ignored
+    aac_channels = 0;
+    aac_extra_size = 0;
+    aac_extra_data = NULL;
 }
 
 SrsAudioCodec::~SrsAudioCodec()
 {
 }
 
-SrsCodecFlvTag SrsAudioCodec::codec()
+bool SrsAudioCodec::is_aac_codec_ok()
 {
-    return SrsCodecFlvTagAudio;
+    return aac_extra_size > 0 && aac_extra_data;
 }
 
 SrsVideoCodec::SrsVideoCodec()
 {
-    frame_type = SrsCodecVideoAVCFrameForbidden;
-    avc_packet_type = SrsCodecVideoAVCTypeForbidden;
-    has_idr = has_aud = has_sps_pps = false;
-    first_nalu_type = SrsAvcNaluTypeForbidden;
+    id = SrsCodecVideoForbidden;
+    video_data_rate = 0;
+    frame_rate = duration = 0;
+    
+    width = 0;
+    height = 0;
+    
+    avc_extra_size = 0;
+    avc_extra_data = NULL;
+    
+    NAL_unit_length = 0;
+    avc_profile = SrsAvcProfileReserved;
+    avc_level = SrsAvcLevelReserved;
+    sequenceParameterSetLength = 0;
+    sequenceParameterSetNALUnit = NULL;
+    pictureParameterSetLength = 0;
+    pictureParameterSetNALUnit = NULL;
+    
+    payload_format = SrsAvcPayloadFormatGuess;
 }
 
 SrsVideoCodec::~SrsVideoCodec()
 {
+    srs_freepa(avc_extra_data);
+    srs_freepa(sequenceParameterSetNALUnit);
+    srs_freepa(pictureParameterSetNALUnit);
 }
 
-SrsCodecFlvTag SrsVideoCodec::codec()
+bool SrsVideoCodec::is_avc_codec_ok()
 {
-    return SrsCodecFlvTagVideo;
+    return avc_extra_size > 0 && avc_extra_data;
 }
 
 SrsFrame::SrsFrame()
 {
     codec = NULL;
     nb_samples = 0;
+    dts = 0;
+    cts = 0;
 }
 
 SrsFrame::~SrsFrame()
@@ -405,348 +432,249 @@ SrsFrame::~SrsFrame()
     srs_freep(codec);
 }
 
-SrsCodecSample::SrsCodecSample()
+int SrsFrame::initialize(SrsCodec* c)
 {
-    clear();
-}
-
-SrsCodecSample::~SrsCodecSample()
-{
-}
-
-void SrsCodecSample::clear()
-{
-    is_video = false;
-    nb_sample_units = 0;
-
+    codec = c;
+    nb_samples = 0;
+    dts = 0;
     cts = 0;
-    frame_type = SrsCodecVideoAVCFrameReserved;
-    avc_packet_type = SrsCodecVideoAVCTypeReserved;
-    has_sps_pps = has_aud = has_idr = false;
-    first_nalu_type = SrsAvcNaluTypeReserved;
-    
-    acodec = SrsCodecAudioReserved1;
-    sound_rate = SrsCodecAudioSampleRateReserved;
-    sound_size = SrsCodecAudioSampleSizeReserved;
-    sound_type = SrsCodecAudioSoundTypeReserved;
-    aac_packet_type = SrsCodecAudioTypeReserved;
+    return ERROR_SUCCESS;
 }
 
-int SrsCodecSample::add_sample_unit(char* bytes, int size)
+int SrsFrame::add_sample(char* bytes, int size)
 {
     int ret = ERROR_SUCCESS;
     
-    if (nb_sample_units >= SRS_MAX_CODEC_SAMPLE) {
+    if (nb_samples >= SRS_MAX_CODEC_SAMPLE) {
         ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("avc exceed samples count, max=%d, video=%d. ret=%d", SRS_MAX_CODEC_SAMPLE, is_video, ret);
+        srs_error("Frame samples overflow, max=%d. ret=%d", SRS_MAX_CODEC_SAMPLE, ret);
         return ret;
     }
     
-    SrsCodecSampleUnit* sample_unit = &sample_units[nb_sample_units++];
-    sample_unit->bytes = bytes;
-    sample_unit->size = size;
+    SrsSample* sample = &samples[nb_samples++];
+    sample->bytes = bytes;
+    sample->size = size;
+    
+    return ret;
+}
+
+SrsAudioFrame::SrsAudioFrame()
+{
+    aac_packet_type = SrsCodecAudioTypeForbidden;
+}
+
+SrsAudioFrame::~SrsAudioFrame()
+{
+}
+
+SrsAudioCodec* SrsAudioFrame::acodec()
+{
+    return (SrsAudioCodec*)codec;
+}
+
+SrsVideoFrame::SrsVideoFrame()
+{
+    frame_type = SrsCodecVideoAVCFrameForbidden;
+    avc_packet_type = SrsCodecVideoAVCTypeForbidden;
+    has_idr = has_aud = has_sps_pps = false;
+    first_nalu_type = SrsAvcNaluTypeForbidden;
+}
+
+SrsVideoFrame::~SrsVideoFrame()
+{
+}
+
+int SrsVideoFrame::add_sample(char* bytes, int size)
+{
+    int ret = ERROR_SUCCESS;
+    
+    if ((ret = SrsFrame::add_sample(bytes, size)) != ERROR_SUCCESS) {
+        return ret;
+    }
     
     // for video, parse the nalu type, set the IDR flag.
-    if (is_video) {
-        SrsAvcNaluType nal_unit_type = (SrsAvcNaluType)(bytes[0] & 0x1f);
-        
-        if (nal_unit_type == SrsAvcNaluTypeIDR) {
-            has_idr = true;
-        } else if (nal_unit_type == SrsAvcNaluTypeSPS || nal_unit_type == SrsAvcNaluTypePPS) {
-            has_sps_pps = true;
-        } else if (nal_unit_type == SrsAvcNaluTypeAccessUnitDelimiter) {
-            has_aud = true;
-        }
+    SrsAvcNaluType nal_unit_type = (SrsAvcNaluType)(bytes[0] & 0x1f);
     
-        if (first_nalu_type == SrsAvcNaluTypeReserved) {
-            first_nalu_type = nal_unit_type;
-        }
+    if (nal_unit_type == SrsAvcNaluTypeIDR) {
+        has_idr = true;
+    } else if (nal_unit_type == SrsAvcNaluTypeSPS || nal_unit_type == SrsAvcNaluTypePPS) {
+        has_sps_pps = true;
+    } else if (nal_unit_type == SrsAvcNaluTypeAccessUnitDelimiter) {
+        has_aud = true;
+    }
+    
+    if (first_nalu_type == SrsAvcNaluTypeReserved) {
+        first_nalu_type = nal_unit_type;
     }
     
     return ret;
 }
 
-SrsAvcAacCodec::SrsAvcAacCodec()
+SrsVideoCodec* SrsVideoFrame::vcodec()
 {
+    return (SrsVideoCodec*)codec;
+}
+
+SrsFormat::SrsFormat()
+{
+    acodec = NULL;
+    vcodec = NULL;
+    audio = NULL;
+    video = NULL;
+    buffer = new SrsBuffer();
     avc_parse_sps = true;
-    
-    width = 0;
-    height = 0;
-    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 = SrsAvcProfileReserved;
-    avc_level = SrsAvcLevelReserved;
-    aac_object = SrsAacObjectTypeReserved;
-    aac_sample_rate = SRS_AAC_SAMPLE_RATE_UNSET; // sample rate ignored
-    aac_channels = 0;
-    avc_extra_size = 0;
-    avc_extra_data = NULL;
-    aac_extra_size = 0;
-    aac_extra_data = NULL;
-
-    sequenceParameterSetLength = 0;
-    sequenceParameterSetNALUnit = NULL;
-    pictureParameterSetLength = 0;
-    pictureParameterSetNALUnit = NULL;
-
-    payload_format = SrsAvcPayloadFormatGuess;
-    stream = new SrsBuffer();
 }
 
-SrsAvcAacCodec::~SrsAvcAacCodec()
+SrsFormat::~SrsFormat()
 {
-    srs_freepa(avc_extra_data);
-    srs_freepa(aac_extra_data);
-
-    srs_freep(stream);
-    srs_freepa(sequenceParameterSetNALUnit);
-    srs_freepa(pictureParameterSetNALUnit);
+    srs_freep(audio);
+    srs_freep(video);
+    srs_freep(acodec);
+    srs_freep(vcodec);
+    srs_freep(buffer);
 }
 
-bool SrsAvcAacCodec::is_avc_codec_ok()
+int SrsFormat::initialize()
 {
-    return avc_extra_size > 0 && avc_extra_data;
+    return ERROR_SUCCESS;
 }
 
-bool SrsAvcAacCodec::is_aac_codec_ok()
-{
-    return aac_extra_size > 0 && aac_extra_data;
-}
-
-int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample)
+int SrsFormat::on_audio(int64_t timestamp, char* data, int size)
 {
     int ret = ERROR_SUCCESS;
     
-    sample->is_video = false;
-    
     if (!data || size <= 0) {
         srs_trace("no audio present, ignore it.");
         return ret;
     }
     
-    if ((ret = stream->initialize(data, size)) != ERROR_SUCCESS) {
+    if ((ret = buffer->initialize(data, size)) != ERROR_SUCCESS) {
         return ret;
     }
-
+    
     // audio decode
-    if (!stream->require(1)) {
+    if (!buffer->require(1)) {
         ret = ERROR_HLS_DECODE_ERROR;
         srs_error("aac decode sound_format failed. ret=%d", ret);
         return ret;
     }
     
     // @see: E.4.2 Audio Tags, video_file_format_spec_v10_1.pdf, page 76
-    int8_t sound_format = stream->read_1bytes();
+    SrsCodecAudio codec = (SrsCodecAudio)((buffer->read_1bytes() >> 4) & 0x0f);
     
-    int8_t sound_type = sound_format & 0x01;
-    int8_t sound_size = (sound_format >> 1) & 0x01;
-    int8_t sound_rate = (sound_format >> 2) & 0x03;
-    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("aac 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("aac decode aac_packet_type failed. ret=%d", ret);
+    if (codec != SrsCodecAudioMP3 && codec != SrsCodecAudioAAC) {
         return ret;
     }
     
-    int8_t aac_packet_type = stream->read_1bytes();
-    sample->aac_packet_type = (SrsCodecAudioType)aac_packet_type;
+    if (!acodec) {
+        acodec = new SrsAudioCodec();
+    }
+    if (!audio) {
+        audio = new SrsAudioFrame();
+    }
     
-    if (aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
-        // AudioSpecificConfig
-        // 1.6.2.1 AudioSpecificConfig, in ISO_IEC_14496-3-AAC-2001.pdf, page 33.
-        aac_extra_size = stream->size() - stream->pos();
-        if (aac_extra_size > 0) {
-            srs_freepa(aac_extra_data);
-            aac_extra_data = new char[aac_extra_size];
-            memcpy(aac_extra_data, stream->data() + stream->pos(), aac_extra_size);
-
-            // demux the sequence header.
-            if ((ret = audio_aac_sequence_header_demux(aac_extra_data, aac_extra_size)) != ERROR_SUCCESS) {
-                return ret;
-            }
-        }
-    } else if (aac_packet_type == SrsCodecAudioTypeRawData) {
-        // ensure the sequence header demuxed
-        if (!is_aac_codec_ok()) {
-            srs_warn("aac ignore type=%d for no sequence header. ret=%d", aac_packet_type, 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("aac add sample failed. ret=%d", ret);
-            return ret;
-        }
+    if ((ret = audio->initialize(acodec)) != ERROR_SUCCESS) {
+        return ret;
+    }
+    
+    buffer->skip(-1 * buffer->pos());
+    if (codec == SrsCodecAudioMP3) {
+        return audio_mp3_demux(buffer, timestamp);
+    } else if (codec == SrsCodecAudioAAC) {
+        return audio_aac_demux(buffer, timestamp);
     } else {
-        // ignored.
+        return ret;
     }
-    
-    // reset the sample rate by sequence header
-    if (aac_sample_rate != SRS_AAC_SAMPLE_RATE_UNSET) {
-        static int aac_sample_rates[] = {
-            96000, 88200, 64000, 48000,
-            44100, 32000, 24000, 22050,
-            16000, 12000, 11025,  8000,
-            7350,     0,     0,    0
-        };
-        switch (aac_sample_rates[aac_sample_rate]) {
-            case 11025:
-                sample->sound_rate = SrsCodecAudioSampleRate11025;
-                break;
-            case 22050:
-                sample->sound_rate = SrsCodecAudioSampleRate22050;
-                break;
-            case 44100:
-                sample->sound_rate = SrsCodecAudioSampleRate44100;
-                break;
-            default:
-                break;
-        };
-    }
-    
-    srs_info("aac decoded, type=%d, codec=%d, asize=%d, rate=%d, format=%d, size=%d",
-        sound_type, audio_codec_id, sound_size, sound_rate, sound_format, size);
-    
-    return ret;
 }
 
-int SrsAvcAacCodec::audio_mp3_demux(char* data, int size, SrsCodecSample* sample)
+int SrsFormat::on_video(int64_t timestamp, char* data, int size)
 {
     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::audio_aac_sequence_header_demux(char* data, int size)
-{
-    int ret = ERROR_SUCCESS;
-    
-    if ((ret = stream->initialize(data, size)) != ERROR_SUCCESS) {
-        return ret;
-    }
-        
-    // only need to decode the first 2bytes:
-    //      audioObjectType, aac_profile, 5bits.
-    //      samplingFrequencyIndex, aac_sample_rate, 4bits.
-    //      channelConfiguration, aac_channels, 4bits
-    if (!stream->require(2)) {
-        ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("audio codec decode aac sequence header failed. ret=%d", ret);
-        return ret;
-    }
-    uint8_t profile_ObjectType = stream->read_1bytes();
-    uint8_t samplingFrequencyIndex = stream->read_1bytes();
-        
-    aac_channels = (samplingFrequencyIndex >> 3) & 0x0f;
-    samplingFrequencyIndex = ((profile_ObjectType << 1) & 0x0e) | ((samplingFrequencyIndex >> 7) & 0x01);
-    profile_ObjectType = (profile_ObjectType >> 3) & 0x1f;
-
-    // set the aac sample rate.
-    aac_sample_rate = samplingFrequencyIndex;
-
-    // convert the object type in sequence header to aac profile of ADTS.
-    aac_object = (SrsAacObjectType)profile_ObjectType;
-    if (aac_object == SrsAacObjectTypeReserved) {
-        ret = ERROR_HLS_DECODE_ERROR;
-        srs_error("audio codec decode aac sequence header failed, "
-            "adts object=%d invalid. ret=%d", profile_ObjectType, ret);
-        return ret;
-    }
-        
-    // TODO: FIXME: to support aac he/he-v2, see: ngx_rtmp_codec_parse_aac_header
-    // @see: https://github.com/winlinvip/nginx-rtmp-module/commit/3a5f9eea78fc8d11e8be922aea9ac349b9dcbfc2
-    // 
-    // donot force to LC, @see: https://github.com/ossrs/srs/issues/81
-    // the source will print the sequence header info.
-    //if (aac_profile > 3) {
-        // Mark all extended profiles as LC
-        // to make Android as happy as possible.
-        // @see: ngx_rtmp_hls_parse_aac_header
-        //aac_profile = 1;
-    //}
-
-    return ret;
-}
-
-int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample)
-{
-    int ret = ERROR_SUCCESS;
-    
-    sample->is_video = true;
     
     if (!data || size <= 0) {
         srs_trace("no video present, ignore it.");
         return ret;
     }
     
-    if ((ret = stream->initialize(data, size)) != ERROR_SUCCESS) {
+    if ((ret = buffer->initialize(data, size)) != ERROR_SUCCESS) {
         return ret;
     }
-
+    
     // video decode
-    if (!stream->require(1)) {
+    if (!buffer->require(1)) {
         ret = ERROR_HLS_DECODE_ERROR;
         srs_error("avc decode frame_type failed. ret=%d", ret);
         return ret;
     }
     
+    // @see: E.4.3 Video Tags, video_file_format_spec_v10_1.pdf, page 78
+    int8_t frame_type = buffer->read_1bytes();
+    SrsCodecVideo codec_id = (SrsCodecVideo)(frame_type & 0x0f);
+    
+    // TODO: Support other codecs.
+    if (codec_id != SrsCodecVideoAVC) {
+        return ret;
+    }
+    
+    if (!vcodec) {
+        vcodec = new SrsVideoCodec();
+    }
+    if (!video) {
+        video = new SrsVideoFrame();
+    }
+    
+    if ((ret = video->initialize(vcodec)) != ERROR_SUCCESS) {
+        return ret;
+    }
+    
+    buffer->skip(-1 * buffer->pos());
+    return video_avc_demux(buffer, timestamp);
+}
+
+int SrsFormat::on_aac_sequence_header(char* data, int size)
+{
+    int ret = ERROR_SUCCESS;
+    
+    if (!acodec) {
+        acodec = new SrsAudioCodec();
+    }
+    if (!audio) {
+        audio = new SrsAudioFrame();
+    }
+    
+    if ((ret = audio->initialize(acodec)) != ERROR_SUCCESS) {
+        return ret;
+    }
+    
+    return audio_aac_sequence_header_demux(data, size);
+}
+
+bool SrsFormat::is_aac_sequence_header()
+{
+    return acodec && acodec->id == SrsCodecAudioAAC
+    && audio && audio->aac_packet_type == SrsCodecAudioTypeSequenceHeader;
+}
+
+bool SrsFormat::is_avc_sequence_header()
+{
+    return vcodec && vcodec->id == SrsCodecVideoAVC
+    && video && video->avc_packet_type == SrsCodecVideoAVCTypeSequenceHeader;
+}
+
+int SrsFormat::video_avc_demux(SrsBuffer* stream, int64_t timestamp)
+{
+    int ret = ERROR_SUCCESS;
+    
     // @see: E.4.3 Video Tags, video_file_format_spec_v10_1.pdf, page 78
     int8_t frame_type = stream->read_1bytes();
-    int8_t codec_id = frame_type & 0x0f;
+    SrsCodecVideo codec_id = (SrsCodecVideo)(frame_type & 0x0f);
     frame_type = (frame_type >> 4) & 0x0f;
     
-    sample->frame_type = (SrsCodecVideoAVCFrame)frame_type;
+    video->frame_type = (SrsCodecVideoAVCFrame)frame_type;
     
     // ignore info frame without error,
     // @see https://github.com/ossrs/srs/issues/288#issuecomment-69863909
-    if (sample->frame_type == SrsCodecVideoAVCFrameVideoInfoFrame) {
+    if (video->frame_type == SrsCodecVideoAVCFrameVideoInfoFrame) {
         srs_warn("avc igone the info frame, ret=%d", ret);
         return ret;
     }
@@ -757,7 +685,7 @@ int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample
         srs_error("avc only support video h.264/avc codec. actual=%d, ret=%d", codec_id, ret);
         return ret;
     }
-    video_codec_id = codec_id;
+    vcodec->id = codec_id;
     
     if (!stream->require(4)) {
         ret = ERROR_HLS_DECODE_ERROR;
@@ -768,15 +696,16 @@ int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample
     int32_t composition_time = stream->read_3bytes();
     
     // pts = dts + cts.
-    sample->cts = composition_time;
-    sample->avc_packet_type = (SrsCodecVideoAVCType)avc_packet_type;
+    video->dts = timestamp;
+    video->cts = composition_time;
+    video->avc_packet_type = (SrsCodecVideoAVCType)avc_packet_type;
     
     if (avc_packet_type == SrsCodecVideoAVCTypeSequenceHeader) {
         if ((ret = avc_demux_sps_pps(stream)) != ERROR_SUCCESS) {
             return ret;
         }
     } else if (avc_packet_type == SrsCodecVideoAVCTypeNALU){
-        if ((ret = video_nalu_demux(stream, sample)) != ERROR_SUCCESS) {
+        if ((ret = video_nalu_demux(stream)) != ERROR_SUCCESS) {
             return ret;
         }
     } else {
@@ -784,84 +713,22 @@ int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample
     }
     
     srs_info("avc decoded, type=%d, codec=%d, avc=%d, cts=%d, size=%d",
-        frame_type, video_codec_id, avc_packet_type, composition_time, size);
+             frame_type, video_codec_id, avc_packet_type, composition_time, size);
     
     return ret;
 }
 
-int SrsAvcAacCodec::video_nalu_demux(SrsBuffer* stream, SrsCodecSample* sample)
-{
-    int ret = ERROR_SUCCESS;
-    
-    // ensure the sequence header demuxed
-    if (!is_avc_codec_ok()) {
-        srs_warn("avc ignore type=%d for no sequence header. ret=%d", SrsCodecVideoAVCTypeNALU, ret);
-        return ret;
-    }
-    
-    // guess for the first time.
-    if (payload_format == SrsAvcPayloadFormatGuess) {
-        // One or more NALUs (Full frames are required)
-        // try  "AnnexB" from ISO_IEC_14496-10-AVC-2003.pdf, page 211.
-        if ((ret = avc_demux_annexb_format(stream, sample)) != ERROR_SUCCESS) {
-            // stop try when system error.
-            if (ret != ERROR_HLS_AVC_TRY_OTHERS) {
-                srs_error("avc demux for annexb failed. ret=%d", ret);
-                return ret;
-            }
-            
-            // try "ISO Base Media File Format" from ISO_IEC_14496-15-AVC-format-2012.pdf, page 20
-            if ((ret = avc_demux_ibmf_format(stream, sample)) != ERROR_SUCCESS) {
-                return ret;
-            } else {
-                payload_format = SrsAvcPayloadFormatIbmf;
-                srs_info("hls guess avc payload is ibmf format.");
-            }
-        } else {
-            payload_format = SrsAvcPayloadFormatAnnexb;
-            srs_info("hls guess avc payload is annexb format.");
-        }
-    } else if (payload_format == SrsAvcPayloadFormatIbmf) {
-        // try "ISO Base Media File Format" from ISO_IEC_14496-15-AVC-format-2012.pdf, page 20
-        if ((ret = avc_demux_ibmf_format(stream, sample)) != ERROR_SUCCESS) {
-            return ret;
-        }
-        srs_info("hls decode avc payload in ibmf format.");
-    } else {
-        // One or more NALUs (Full frames are required)
-        // try  "AnnexB" from ISO_IEC_14496-10-AVC-2003.pdf, page 211.
-        if ((ret = avc_demux_annexb_format(stream, sample)) != ERROR_SUCCESS) {
-            // ok, we guess out the payload is annexb, but maybe changed to ibmf.
-            if (ret != ERROR_HLS_AVC_TRY_OTHERS) {
-                srs_error("avc demux for annexb failed. ret=%d", ret);
-                return ret;
-            }
-            
-            // try "ISO Base Media File Format" from ISO_IEC_14496-15-AVC-format-2012.pdf, page 20
-            if ((ret = avc_demux_ibmf_format(stream, sample)) != ERROR_SUCCESS) {
-                return ret;
-            } else {
-                payload_format = SrsAvcPayloadFormatIbmf;
-                srs_warn("hls avc payload change from annexb to ibmf format.");
-            }
-        }
-        srs_info("hls decode avc payload in annexb format.");
-    }
-    
-    return ret;
-}
-
-int SrsAvcAacCodec::avc_demux_sps_pps(SrsBuffer* stream)
+int SrsFormat::avc_demux_sps_pps(SrsBuffer* stream)
 {
     int ret = ERROR_SUCCESS;
     
     // AVCDecoderConfigurationRecord
     // 5.2.4.1.1 Syntax, ISO_IEC_14496-15-AVC-format-2012.pdf, page 16
-    avc_extra_size = stream->size() - stream->pos();
-    if (avc_extra_size > 0) {
-        srs_freepa(avc_extra_data);
-        avc_extra_data = new char[avc_extra_size];
-        memcpy(avc_extra_data, stream->data() + stream->pos(), avc_extra_size);
+    vcodec->avc_extra_size = stream->size() - stream->pos();
+    if (vcodec->avc_extra_size > 0) {
+        srs_freepa(vcodec->avc_extra_data);
+        vcodec->avc_extra_data = new char[vcodec->avc_extra_size];
+        memcpy(vcodec->avc_extra_data, stream->data() + stream->pos(), vcodec->avc_extra_size);
     }
     
     if (!stream->require(6)) {
@@ -872,23 +739,23 @@ int SrsAvcAacCodec::avc_demux_sps_pps(SrsBuffer* stream)
     //int8_t configurationVersion = stream->read_1bytes();
     stream->read_1bytes();
     //int8_t AVCProfileIndication = stream->read_1bytes();
-    avc_profile = (SrsAvcProfile)stream->read_1bytes();
+    vcodec->avc_profile = (SrsAvcProfile)stream->read_1bytes();
     //int8_t profile_compatibility = stream->read_1bytes();
     stream->read_1bytes();
     //int8_t AVCLevelIndication = stream->read_1bytes();
-    avc_level = (SrsAvcLevel)stream->read_1bytes();
+    vcodec->avc_level = (SrsAvcLevel)stream->read_1bytes();
     
     // parse the NALU size.
     int8_t lengthSizeMinusOne = stream->read_1bytes();
     lengthSizeMinusOne &= 0x03;
-    NAL_unit_length = lengthSizeMinusOne;
+    vcodec->NAL_unit_length = lengthSizeMinusOne;
     
     // 5.3.4.2.1 Syntax, ISO_IEC_14496-15-AVC-format-2012.pdf, page 16
     // 5.2.4.1 AVC decoder configuration record
     // 5.2.4.1.2 Semantics
     // The value of this field shall be one of 0, 1, or 3 corresponding to a
     // length encoded with 1, 2, or 4 bytes, respectively.
-    if (NAL_unit_length == 2) {
+    if (vcodec->NAL_unit_length == 2) {
         ret = ERROR_HLS_DECODE_ERROR;
         srs_error("sps lengthSizeMinusOne should never be 2. ret=%d", ret);
         return ret;
@@ -913,16 +780,16 @@ int SrsAvcAacCodec::avc_demux_sps_pps(SrsBuffer* stream)
         srs_error("avc decode sequenc header sps size failed. ret=%d", ret);
         return ret;
     }
-    sequenceParameterSetLength = stream->read_2bytes();
-    if (!stream->require(sequenceParameterSetLength)) {
+    vcodec->sequenceParameterSetLength = stream->read_2bytes();
+    if (!stream->require(vcodec->sequenceParameterSetLength)) {
         ret = ERROR_HLS_DECODE_ERROR;
         srs_error("avc decode sequenc header sps data failed. ret=%d", ret);
         return ret;
     }
-    if (sequenceParameterSetLength > 0) {
-        srs_freepa(sequenceParameterSetNALUnit);
-        sequenceParameterSetNALUnit = new char[sequenceParameterSetLength];
-        stream->read_bytes(sequenceParameterSetNALUnit, sequenceParameterSetLength);
+    if (vcodec->sequenceParameterSetLength > 0) {
+        srs_freepa(vcodec->sequenceParameterSetNALUnit);
+        vcodec->sequenceParameterSetNALUnit = new char[vcodec->sequenceParameterSetLength];
+        stream->read_bytes(vcodec->sequenceParameterSetNALUnit, vcodec->sequenceParameterSetLength);
     }
     // 1 pps
     if (!stream->require(1)) {
@@ -942,31 +809,31 @@ int SrsAvcAacCodec::avc_demux_sps_pps(SrsBuffer* stream)
         srs_error("avc decode sequenc header pps size failed. ret=%d", ret);
         return ret;
     }
-    pictureParameterSetLength = stream->read_2bytes();
-    if (!stream->require(pictureParameterSetLength)) {
+    vcodec->pictureParameterSetLength = stream->read_2bytes();
+    if (!stream->require(vcodec->pictureParameterSetLength)) {
         ret = ERROR_HLS_DECODE_ERROR;
         srs_error("avc decode sequenc header pps data failed. ret=%d", ret);
         return ret;
     }
-    if (pictureParameterSetLength > 0) {
-        srs_freepa(pictureParameterSetNALUnit);
-        pictureParameterSetNALUnit = new char[pictureParameterSetLength];
-        stream->read_bytes(pictureParameterSetNALUnit, pictureParameterSetLength);
+    if (vcodec->pictureParameterSetLength > 0) {
+        srs_freepa(vcodec->pictureParameterSetNALUnit);
+        vcodec->pictureParameterSetNALUnit = new char[vcodec->pictureParameterSetLength];
+        stream->read_bytes(vcodec->pictureParameterSetNALUnit, vcodec->pictureParameterSetLength);
     }
     
     return avc_demux_sps();
 }
 
-int SrsAvcAacCodec::avc_demux_sps()
+int SrsFormat::avc_demux_sps()
 {
     int ret = ERROR_SUCCESS;
     
-    if (!sequenceParameterSetLength) {
+    if (!vcodec->sequenceParameterSetLength) {
         return ret;
     }
     
     SrsBuffer stream;
-    if ((ret = stream.initialize(sequenceParameterSetNALUnit, sequenceParameterSetLength)) != ERROR_SUCCESS) {
+    if ((ret = stream.initialize(vcodec->sequenceParameterSetNALUnit, vcodec->sequenceParameterSetLength)) != ERROR_SUCCESS) {
         return ret;
     }
     
@@ -1008,7 +875,7 @@ int SrsAvcAacCodec::avc_demux_sps()
     
     // decode the rbsp from sps.
     // rbsp[ i ] a raw byte sequence payload is specified as an ordered sequence of bytes.
-    int8_t* rbsp = new int8_t[sequenceParameterSetLength];
+    int8_t* rbsp = new int8_t[vcodec->sequenceParameterSetLength];
     SrsAutoFreeA(int8_t, rbsp);
     
     int nb_rbsp = 0;
@@ -1034,7 +901,7 @@ int SrsAvcAacCodec::avc_demux_sps()
 }
 
 
-int SrsAvcAacCodec::avc_demux_sps_rbsp(char* rbsp, int nb_rbsp)
+int SrsFormat::avc_demux_sps_rbsp(char* rbsp, int nb_rbsp)
 {
     int ret = ERROR_SUCCESS;
     
@@ -1098,7 +965,7 @@ int SrsAvcAacCodec::avc_demux_sps_rbsp(char* rbsp, int nb_rbsp)
     if (profile_idc == 100 || profile_idc == 110 || profile_idc == 122 || profile_idc == 244
         || profile_idc == 44 || profile_idc == 83 || profile_idc == 86 || profile_idc == 118
         || profile_idc == 128
-    ) {
+        ) {
         if ((ret = srs_avc_nalu_read_uev(&bs, chroma_format_idc)) != ERROR_SUCCESS) {
             return ret;
         }
@@ -1207,13 +1074,75 @@ int SrsAvcAacCodec::avc_demux_sps_rbsp(char* rbsp, int nb_rbsp)
         return ret;
     }
     
-    width = (int)(pic_width_in_mbs_minus1 + 1) * 16;
-    height = (int)(pic_height_in_map_units_minus1 + 1) * 16;
+    vcodec->width = (int)(pic_width_in_mbs_minus1 + 1) * 16;
+    vcodec->height = (int)(pic_height_in_map_units_minus1 + 1) * 16;
     
     return ret;
 }
 
-int SrsAvcAacCodec::avc_demux_annexb_format(SrsBuffer* stream, SrsCodecSample* sample)
+int SrsFormat::video_nalu_demux(SrsBuffer* stream)
+{
+    int ret = ERROR_SUCCESS;
+    
+    // ensure the sequence header demuxed
+    if (!vcodec->is_avc_codec_ok()) {
+        srs_warn("avc ignore type=%d for no sequence header. ret=%d", SrsCodecVideoAVCTypeNALU, ret);
+        return ret;
+    }
+    
+    // guess for the first time.
+    if (vcodec->payload_format == SrsAvcPayloadFormatGuess) {
+        // One or more NALUs (Full frames are required)
+        // try  "AnnexB" from ISO_IEC_14496-10-AVC-2003.pdf, page 211.
+        if ((ret = avc_demux_annexb_format(stream)) != ERROR_SUCCESS) {
+            // stop try when system error.
+            if (ret != ERROR_HLS_AVC_TRY_OTHERS) {
+                srs_error("avc demux for annexb failed. ret=%d", ret);
+                return ret;
+            }
+            
+            // try "ISO Base Media File Format" from ISO_IEC_14496-15-AVC-format-2012.pdf, page 20
+            if ((ret = avc_demux_ibmf_format(stream)) != ERROR_SUCCESS) {
+                return ret;
+            } else {
+                vcodec->payload_format = SrsAvcPayloadFormatIbmf;
+                srs_info("hls guess avc payload is ibmf format.");
+            }
+        } else {
+            vcodec->payload_format = SrsAvcPayloadFormatAnnexb;
+            srs_info("hls guess avc payload is annexb format.");
+        }
+    } else if (vcodec->payload_format == SrsAvcPayloadFormatIbmf) {
+        // try "ISO Base Media File Format" from ISO_IEC_14496-15-AVC-format-2012.pdf, page 20
+        if ((ret = avc_demux_ibmf_format(stream)) != ERROR_SUCCESS) {
+            return ret;
+        }
+        srs_info("hls decode avc payload in ibmf format.");
+    } else {
+        // One or more NALUs (Full frames are required)
+        // try  "AnnexB" from ISO_IEC_14496-10-AVC-2003.pdf, page 211.
+        if ((ret = avc_demux_annexb_format(stream)) != ERROR_SUCCESS) {
+            // ok, we guess out the payload is annexb, but maybe changed to ibmf.
+            if (ret != ERROR_HLS_AVC_TRY_OTHERS) {
+                srs_error("avc demux for annexb failed. ret=%d", ret);
+                return ret;
+            }
+            
+            // try "ISO Base Media File Format" from ISO_IEC_14496-15-AVC-format-2012.pdf, page 20
+            if ((ret = avc_demux_ibmf_format(stream)) != ERROR_SUCCESS) {
+                return ret;
+            } else {
+                vcodec->payload_format = SrsAvcPayloadFormatIbmf;
+                srs_warn("hls avc payload change from annexb to ibmf format.");
+            }
+        }
+        srs_info("hls decode avc payload in annexb format.");
+    }
+    
+    return ret;
+}
+
+int SrsFormat::avc_demux_annexb_format(SrsBuffer* stream)
 {
     int ret = ERROR_SUCCESS;
     
@@ -1231,7 +1160,7 @@ int SrsAvcAacCodec::avc_demux_annexb_format(SrsBuffer* stream, SrsCodecSample* s
         if (!srs_avc_startswith_annexb(stream, &nb_start_code)) {
             return ret;
         }
-
+        
         // skip the start code.
         if (nb_start_code > 0) {
             stream->skip(nb_start_code);
@@ -1257,7 +1186,7 @@ int SrsAvcAacCodec::avc_demux_annexb_format(SrsBuffer* stream, SrsCodecSample* s
         }
         
         // got the NALU.
-        if ((ret = sample->add_sample_unit(p, (int)(pp - p))) != ERROR_SUCCESS) {
+        if ((ret = video->add_sample(p, (int)(pp - p))) != ERROR_SUCCESS) {
             srs_error("annexb add video sample failed. ret=%d", ret);
             return ret;
         }
@@ -1266,7 +1195,7 @@ int SrsAvcAacCodec::avc_demux_annexb_format(SrsBuffer* stream, SrsCodecSample* s
     return ret;
 }
 
-int SrsAvcAacCodec::avc_demux_ibmf_format(SrsBuffer* stream, SrsCodecSample* sample)
+int SrsFormat::avc_demux_ibmf_format(SrsBuffer* stream)
 {
     int ret = ERROR_SUCCESS;
     
@@ -1277,20 +1206,20 @@ int SrsAvcAacCodec::avc_demux_ibmf_format(SrsBuffer* stream, SrsCodecSample* sam
     // 5.2.4.1.2 Semantics
     // The value of this field shall be one of 0, 1, or 3 corresponding to a
     // length encoded with 1, 2, or 4 bytes, respectively.
-    srs_assert(NAL_unit_length != 2);
+    srs_assert(vcodec->NAL_unit_length != 2);
     
     // 5.3.4.2.1 Syntax, ISO_IEC_14496-15-AVC-format-2012.pdf, page 20
     for (int i = 0; i < PictureLength;) {
         // unsigned int((NAL_unit_length+1)*8) NALUnitLength;
-        if (!stream->require(NAL_unit_length + 1)) {
+        if (!stream->require(vcodec->NAL_unit_length + 1)) {
             ret = ERROR_HLS_DECODE_ERROR;
             srs_error("avc decode NALU size failed. ret=%d", ret);
             return ret;
         }
         int32_t NALUnitLength = 0;
-        if (NAL_unit_length == 3) {
+        if (vcodec->NAL_unit_length == 3) {
             NALUnitLength = stream->read_4bytes();
-        } else if (NAL_unit_length == 1) {
+        } else if (vcodec->NAL_unit_length == 1) {
             NALUnitLength = stream->read_2bytes();
         } else {
             NALUnitLength = stream->read_1bytes();
@@ -1311,15 +1240,198 @@ int SrsAvcAacCodec::avc_demux_ibmf_format(SrsBuffer* stream, SrsCodecSample* sam
             return ret;
         }
         // 7.3.1 NAL unit syntax, ISO_IEC_14496-10-AVC-2003.pdf, page 44.
-        if ((ret = sample->add_sample_unit(stream->data() + stream->pos(), NALUnitLength)) != ERROR_SUCCESS) {
+        if ((ret = video->add_sample(stream->data() + stream->pos(), NALUnitLength)) != ERROR_SUCCESS) {
             srs_error("avc add video sample failed. ret=%d", ret);
             return ret;
         }
         stream->skip(NALUnitLength);
         
-        i += NAL_unit_length + 1 + NALUnitLength;
+        i += vcodec->NAL_unit_length + 1 + NALUnitLength;
     }
     
     return ret;
 }
 
+int SrsFormat::audio_aac_demux(SrsBuffer* stream, int64_t timestamp)
+{
+    int ret = ERROR_SUCCESS;
+    
+    audio->cts = 0;
+    audio->dts = timestamp;
+    
+    // @see: E.4.2 Audio Tags, video_file_format_spec_v10_1.pdf, page 76
+    int8_t sound_format = stream->read_1bytes();
+    
+    int8_t sound_type = sound_format & 0x01;
+    int8_t sound_size = (sound_format >> 1) & 0x01;
+    int8_t sound_rate = (sound_format >> 2) & 0x03;
+    sound_format = (sound_format >> 4) & 0x0f;
+    
+    SrsCodecAudio codec_id = (SrsCodecAudio)sound_format;
+    acodec->id = codec_id;
+    
+    acodec->sound_type = (SrsCodecAudioSoundType)sound_type;
+    acodec->sound_rate = (SrsCodecAudioSampleRate)sound_rate;
+    acodec->sound_size = (SrsCodecAudioSampleSize)sound_size;
+    
+    // we support h.264+mp3 for hls.
+    if (codec_id == SrsCodecAudioMP3) {
+        return ERROR_HLS_TRY_MP3;
+    }
+    
+    // only support aac
+    if (codec_id != SrsCodecAudioAAC) {
+        ret = ERROR_HLS_DECODE_ERROR;
+        srs_error("aac only support mp3/aac codec. actual=%d, ret=%d", codec_id, ret);
+        return ret;
+    }
+    
+    if (!stream->require(1)) {
+        ret = ERROR_HLS_DECODE_ERROR;
+        srs_error("aac decode aac_packet_type failed. ret=%d", ret);
+        return ret;
+    }
+    
+    SrsCodecAudioType aac_packet_type = (SrsCodecAudioType)stream->read_1bytes();
+    audio->aac_packet_type = (SrsCodecAudioType)aac_packet_type;
+    
+    if (aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
+        // AudioSpecificConfig
+        // 1.6.2.1 AudioSpecificConfig, in ISO_IEC_14496-3-AAC-2001.pdf, page 33.
+        acodec->aac_extra_size = stream->size() - stream->pos();
+        if (acodec->aac_extra_size > 0) {
+            srs_freepa(acodec->aac_extra_data);
+            acodec->aac_extra_data = new char[acodec->aac_extra_size];
+            memcpy(acodec->aac_extra_data, stream->data() + stream->pos(), acodec->aac_extra_size);
+            
+            if ((ret = audio_aac_sequence_header_demux(acodec->aac_extra_data, acodec->aac_extra_size)) != ERROR_SUCCESS) {
+                return ret;
+            }
+        }
+    } else if (aac_packet_type == SrsCodecAudioTypeRawData) {
+        // ensure the sequence header demuxed
+        if (!acodec->is_aac_codec_ok()) {
+            srs_warn("aac ignore type=%d for no sequence header. ret=%d", aac_packet_type, ret);
+            return ret;
+        }
+        
+        // Raw AAC frame data in UI8 []
+        // 6.3 Raw Data, aac-iso-13818-7.pdf, page 28
+        if ((ret = audio->add_sample(stream->data() + stream->pos(), stream->size() - stream->pos())) != ERROR_SUCCESS) {
+            srs_error("aac add sample failed. ret=%d", ret);
+            return ret;
+        }
+    } else {
+        // ignored.
+    }
+    
+    // reset the sample rate by sequence header
+    if (acodec->aac_sample_rate != SRS_AAC_SAMPLE_RATE_UNSET) {
+        static int aac_sample_rates[] = {
+            96000, 88200, 64000, 48000,
+            44100, 32000, 24000, 22050,
+            16000, 12000, 11025,  8000,
+            7350,     0,     0,    0
+        };
+        switch (aac_sample_rates[acodec->aac_sample_rate]) {
+            case 11025:
+                acodec->sound_rate = SrsCodecAudioSampleRate11025;
+                break;
+            case 22050:
+                acodec->sound_rate = SrsCodecAudioSampleRate22050;
+                break;
+            case 44100:
+                acodec->sound_rate = SrsCodecAudioSampleRate44100;
+                break;
+            default:
+                break;
+        };
+    }
+    
+    srs_info("aac decoded, type=%d, codec=%d, asize=%d, rate=%d, format=%d, size=%d",
+             sound_type, codec_id, sound_size, sound_rate, sound_format, size);
+    
+    return ret;
+}
+
+int SrsFormat::audio_mp3_demux(SrsBuffer* stream, int64_t timestamp)
+{
+    int ret = ERROR_SUCCESS;
+    
+    audio->cts = 0;
+    audio->dts = timestamp;
+    
+    // we always decode aac then mp3.
+    srs_assert(acodec->id == SrsCodecAudioMP3);
+    
+    stream->skip(1);
+    if (stream->empty()) {
+        return ret;
+    }
+    
+    char* data = stream->data() + stream->pos();
+    int size = stream->size() - stream->pos();
+    
+    // mp3 payload.
+    if ((ret = audio->add_sample(data, size)) != 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",
+             acodec->sound_type, acodec->id, acodec->sound_size, acodec->sound_rate, acodec->acodec, size);
+    
+    return ret;
+}
+
+int SrsFormat::audio_aac_sequence_header_demux(char* data, int size)
+{
+    int ret = ERROR_SUCCESS;
+    
+    if ((ret = buffer->initialize(data, size)) != ERROR_SUCCESS) {
+        return ret;
+    }
+    
+    // only need to decode the first 2bytes:
+    //      audioObjectType, aac_profile, 5bits.
+    //      samplingFrequencyIndex, aac_sample_rate, 4bits.
+    //      channelConfiguration, aac_channels, 4bits
+    if (!buffer->require(2)) {
+        ret = ERROR_HLS_DECODE_ERROR;
+        srs_error("audio codec decode aac sequence header failed. ret=%d", ret);
+        return ret;
+    }
+    uint8_t profile_ObjectType = buffer->read_1bytes();
+    uint8_t samplingFrequencyIndex = buffer->read_1bytes();
+    
+    acodec->aac_channels = (samplingFrequencyIndex >> 3) & 0x0f;
+    samplingFrequencyIndex = ((profile_ObjectType << 1) & 0x0e) | ((samplingFrequencyIndex >> 7) & 0x01);
+    profile_ObjectType = (profile_ObjectType >> 3) & 0x1f;
+    
+    // set the aac sample rate.
+    acodec->aac_sample_rate = samplingFrequencyIndex;
+    
+    // convert the object type in sequence header to aac profile of ADTS.
+    acodec->aac_object = (SrsAacObjectType)profile_ObjectType;
+    if (acodec->aac_object == SrsAacObjectTypeReserved) {
+        ret = ERROR_HLS_DECODE_ERROR;
+        srs_error("audio codec decode aac sequence header failed, "
+                  "adts object=%d invalid. ret=%d", profile_ObjectType, ret);
+        return ret;
+    }
+    
+    // TODO: FIXME: to support aac he/he-v2, see: ngx_rtmp_codec_parse_aac_header
+    // @see: https://github.com/winlinvip/nginx-rtmp-module/commit/3a5f9eea78fc8d11e8be922aea9ac349b9dcbfc2
+    //
+    // donot force to LC, @see: https://github.com/ossrs/srs/issues/81
+    // the source will print the sequence header info.
+    //if (aac_profile > 3) {
+    // Mark all extended profiles as LC
+    // to make Android as happy as possible.
+    // @see: ngx_rtmp_hls_parse_aac_header
+    //aac_profile = 1;
+    //}
+    
+    return ret;
+}
+
diff --git a/trunk/src/kernel/srs_kernel_codec.hpp b/trunk/src/kernel/srs_kernel_codec.hpp
index 66248f165..e7eaa8a66 100644
--- a/trunk/src/kernel/srs_kernel_codec.hpp
+++ b/trunk/src/kernel/srs_kernel_codec.hpp
@@ -201,7 +201,6 @@ enum SrsCodecFlvTag
 
 /**
 * Annex E. The FLV File Format
-* @see SrsAvcAacCodec for the media stream codec.
 */
 class SrsFlvCodec
 {
@@ -375,6 +374,7 @@ std::string srs_codec_aac_profile2str(SrsAacProfile aac_profile);
 enum SrsAacObjectType
 {
     SrsAacObjectTypeReserved = 0,
+    SrsAacObjectTypeForbidden = 0,
     
     // Table 1.1 - Audio Object Type definition
     // @see @see ISO_IEC_14496-3-AAC-2001.pdf, page 23
@@ -454,9 +454,9 @@ class SrsSample
 {
 public:
     // The size of unit.
-    int nb_unit;
+    int size;
     // The ptr of unit, user must manage it.
-    char* unit;
+    char* bytes;
 public:
     SrsSample();
     virtual ~SrsSample();
@@ -472,9 +472,6 @@ class SrsCodec
 public:
     SrsCodec();
     virtual ~SrsCodec();
-public:
-    // Get the codec type.
-    virtual SrsCodecFlvTag codec() = 0;
 };
 
 /**
@@ -484,17 +481,41 @@ class SrsAudioCodec : public SrsCodec
 {
 public:
     // audio specified
-    SrsCodecAudio acodec;
+    SrsCodecAudio id;
     // audio aac specified.
     SrsCodecAudioSampleRate sound_rate;
     SrsCodecAudioSampleSize sound_size;
     SrsCodecAudioSoundType sound_type;
-    SrsCodecAudioType aac_packet_type;
+    int audio_data_rate; // in bps
+public:
+    /**
+     * audio specified
+     * audioObjectType, in 1.6.2.1 AudioSpecificConfig, page 33,
+     * 1.5.1.1 Audio object type definition, page 23,
+     *           in ISO_IEC_14496-3-AAC-2001.pdf.
+     */
+    SrsAacObjectType aac_object;
+    /**
+     * samplingFrequencyIndex
+     */
+    uint8_t aac_sample_rate;
+    /**
+     * channelConfiguration
+     */
+    uint8_t aac_channels;
+public:
+    /**
+     * the aac extra data, the AAC sequence header,
+     * without the flv codec header,
+     * @see: ffmpeg, AVCodecContext::extradata
+     */
+    int aac_extra_size;
+    char* aac_extra_data;
 public:
     SrsAudioCodec();
     virtual ~SrsAudioCodec();
 public:
-    virtual SrsCodecFlvTag codec();
+    virtual bool is_aac_codec_ok();
 };
 
 /**
@@ -503,29 +524,54 @@ public:
 class SrsVideoCodec : public SrsCodec
 {
 public:
-    // video specified
-    SrsCodecVideoAVCFrame frame_type;
-    SrsCodecVideoAVCType avc_packet_type;
-    // whether sample_units contains IDR frame.
-    bool has_idr;
-    // Whether exists AUD NALU.
-    bool has_aud;
-    // Whether exists SPS/PPS NALU.
-    bool has_sps_pps;
-    // The first nalu type.
-    SrsAvcNaluType first_nalu_type;
+    SrsCodecVideo id;
+    int video_data_rate; // in bps
+    double frame_rate;
+    double duration;
+    int width;
+    int height;
+public:
+    /**
+     * the avc extra data, the AVC sequence header,
+     * without the flv codec header,
+     * @see: ffmpeg, AVCodecContext::extradata
+     */
+    int avc_extra_size;
+    char* avc_extra_data;
+public:
+    /**
+     * video specified
+     */
+    // profile_idc, ISO_IEC_14496-10-AVC-2003.pdf, page 45.
+    SrsAvcProfile avc_profile;
+    // level_idc, ISO_IEC_14496-10-AVC-2003.pdf, page 45.
+    SrsAvcLevel avc_level;
+    // lengthSizeMinusOne, ISO_IEC_14496-15-AVC-format-2012.pdf, page 16
+    int8_t NAL_unit_length;
+    uint16_t sequenceParameterSetLength;
+    char* sequenceParameterSetNALUnit;
+    uint16_t pictureParameterSetLength;
+    char* pictureParameterSetNALUnit;
+public:
+    // the avc payload format.
+    SrsAvcPayloadFormat payload_format;
 public:
     SrsVideoCodec();
     virtual ~SrsVideoCodec();
 public:
-    virtual SrsCodecFlvTag codec();
+    virtual bool is_avc_codec_ok();
 };
 
 /**
- * A codec frame, consists of a codec and a group of samples.
+ * A frame, consists of a codec and a group of samples.
  */
 class SrsFrame
 {
+public:
+    // The DTS/PTS in milliseconds, which is TBN=1000.
+    int64_t dts;
+    // PTS = DTS + CTS.
+    int32_t cts;
 public:
     // The codec info of frame.
     SrsCodec* codec;
@@ -536,43 +582,32 @@ public:
 public:
     SrsFrame();
     virtual ~SrsFrame();
+public:
+    // Initialize the frame, to parse sampels.
+    virtual int initialize(SrsCodec* c);
+    // Add a sample to frame.
+    virtual int add_sample(char* bytes, int size);
 };
 
 /**
-* the samples in the flv audio/video packet.
-* the sample used to analysis a video/audio packet,
-* split the h.264 NALUs to buffers, or aac raw data to a buffer,
-* and decode the video/audio specified infos.
-* 
-* the sample unit:
-*       a video packet codec in h.264 contains many NALUs, each is a sample unit.
-*       a audio packet codec in aac is a sample unit.
-* @remark, the video/audio sequence header is not sample unit,
-*       all sequence header stores as extra data, 
-*       @see SrsAvcAacCodec.avc_extra_data and SrsAvcAacCodec.aac_extra_data
-* @remark, user must clear all samples before decode a new video/audio packet.
-*/
-class SrsCodecSample
+ * A audio frame, besides a frame, contains the audio frame info, such as frame type.
+ */
+class SrsAudioFrame : public SrsFrame
 {
 public:
-    /**
-    * each audio/video raw data packet will dumps to one or multiple buffers,
-    * the buffers will write to hls and clear to reset.
-    * generally, aac audio packet corresponding to one buffer,
-    * where avc/h264 video packet may contains multiple buffer.
-    */
-    int nb_sample_units;
-    SrsCodecSampleUnit sample_units[SRS_MAX_CODEC_SAMPLE];
+    SrsCodecAudioType aac_packet_type;
 public:
-    /**
-    * whether the sample is video sample which demux from video packet.
-    */
-    bool is_video;
-    /**
-    * CompositionTime, video_file_format_spec_v10_1.pdf, page 78.
-    * cts = pts - dts, where dts = flvheader->timestamp.
-    */
-    int32_t cts;
+    SrsAudioFrame();
+    virtual ~SrsAudioFrame();
+public:
+    virtual SrsAudioCodec* acodec();
+};
+
+/**
+ * A video frame, besides a frame, contains the video frame info, such as frame type.
+ */
+class SrsVideoFrame : public SrsFrame
+{
 public:
     // video specified
     SrsCodecVideoAVCFrame frame_type;
@@ -586,171 +621,74 @@ public:
     // The first nalu type.
     SrsAvcNaluType first_nalu_type;
 public:
-    // audio specified
-    SrsCodecAudio acodec;
-    // audio aac specified.
-    SrsCodecAudioSampleRate sound_rate;
-    SrsCodecAudioSampleSize sound_size;
-    SrsCodecAudioSoundType sound_type;
-    SrsCodecAudioType aac_packet_type;
+    SrsVideoFrame();
+    virtual ~SrsVideoFrame();
 public:
-    SrsCodecSample();
-    virtual ~SrsCodecSample();
+    // Add the sample without ANNEXB or IBMF header, or RAW AAC or MP3 data.
+    virtual int add_sample(char* bytes, int size);
 public:
-    /**
-    * clear all samples.
-    * the sample units never copy the bytes, it directly use the ptr,
-    * so when video/audio packet is destroyed, the sample must be clear.
-    * in a word, user must clear sample before demux it.
-    * @remark demux sample use SrsAvcAacCodec.audio_aac_demux or video_avc_demux.
-    */
-    void clear();
-    /**
-    * add the a sample unit, it's a h.264 NALU or aac raw data.
-    * the sample unit directly use the ptr of packet bytes,
-    * so user must never use sample unit when packet is destroyed.
-    * in a word, user must clear sample before demux it.
-    */
-    int add_sample_unit(char* bytes, int size);
+    virtual SrsVideoCodec* vcodec();
 };
 
 /**
-* the h264/avc and aac codec, for media stream.
-*
-* to demux the FLV/RTMP video/audio packet to sample,
-* add each NALUs of h.264 as a sample unit to sample,
-* while the entire aac raw data as a sample unit.
-*
-* for sequence header,
-* demux it and save it in the avc_extra_data and aac_extra_data,
-* 
-* for the codec info, such as audio sample rate,
-* decode from FLV/RTMP header, then use codec info in sequence 
-* header to override it.
-*/
-class SrsAvcAacCodec
+ * A codec format, including one or many stream, each stream identified by a frame.
+ * For example, a typical RTMP stream format, consits of a video and audio frame.
+ * Maybe some RTMP stream only has a audio stream, for instance, redio application.
+ */
+class SrsFormat
 {
-private:
-    SrsBuffer* stream;
 public:
-    /**
-    * metadata specified
-    */
-    int             duration;
-    int             width;
-    int             height;
-    int             frame_rate;
-    // @see: SrsCodecVideo
-    int             video_codec_id;
-    int             video_data_rate; // in bps
-    // @see: SrsCod ecAudioType
-    int             audio_codec_id;
-    int             audio_data_rate; // in bps
-public:
-    /**
-    * video specified
-    */
-    // profile_idc, ISO_IEC_14496-10-AVC-2003.pdf, page 45.
-    SrsAvcProfile   avc_profile; 
-    // level_idc, ISO_IEC_14496-10-AVC-2003.pdf, page 45.
-    SrsAvcLevel     avc_level; 
-    // lengthSizeMinusOne, ISO_IEC_14496-15-AVC-format-2012.pdf, page 16
-    int8_t          NAL_unit_length;
-    uint16_t       sequenceParameterSetLength;
-    char*           sequenceParameterSetNALUnit;
-    uint16_t       pictureParameterSetLength;
-    char*           pictureParameterSetNALUnit;
-private:
-    // the avc payload format.
-    SrsAvcPayloadFormat payload_format;
-public:
-    /**
-    * audio specified
-    * audioObjectType, in 1.6.2.1 AudioSpecificConfig, page 33,
-    * 1.5.1.1 Audio object type definition, page 23,
-    *           in ISO_IEC_14496-3-AAC-2001.pdf.
-    */
-    SrsAacObjectType    aac_object;
-    /**
-    * samplingFrequencyIndex
-    */
-    uint8_t        aac_sample_rate;
-    /**
-    * channelConfiguration
-    */
-    uint8_t        aac_channels;
-public:
-    /**
-    * the avc extra data, the AVC sequence header,
-    * without the flv codec header,
-    * @see: ffmpeg, AVCodecContext::extradata
-    */
-    int             avc_extra_size;
-    char*           avc_extra_data;
-    /**
-    * the aac extra data, the AAC sequence header,
-    * without the flv codec header,
-    * @see: ffmpeg, AVCodecContext::extradata
-    */
-    int             aac_extra_size;
-    char*           aac_extra_data;
+    SrsAudioFrame* audio;
+    SrsAudioCodec* acodec;
+    SrsVideoFrame* video;
+    SrsVideoCodec* vcodec;
+    SrsBuffer* buffer;
 public:
     // for sequence header, whether parse the h.264 sps.
+    // TODO: FIXME: Refine it.
     bool            avc_parse_sps;
 public:
-    SrsAvcAacCodec();
-    virtual ~SrsAvcAacCodec();
+    SrsFormat();
+    virtual ~SrsFormat();
 public:
-    // whether avc or aac codec sequence header or extra data is decoded ok.
-    virtual bool is_avc_codec_ok();
-    virtual bool is_aac_codec_ok();
-// the following function used for hls to build the sample and codec.
+    // Initialize the format.
+    virtual int initialize();
+    // When got a parsed audio packet.
+    virtual int on_audio(int64_t timestamp, char* data, int size);
+    // When got a parsed video packet.
+    virtual int on_video(int64_t timestamp, char* data, int size);
+    // When got a audio aac sequence header.
+    virtual int on_aac_sequence_header(char* data, int size);
 public:
-    /**
-    * demux the audio packet in aac codec.
-    * the packet mux in FLV/RTMP format defined in flv specification.
-    * demux the audio speicified data(sound_format, sound_size, ...) to sample.
-    * demux the aac specified data(aac_profile, ...) to codec from sequence header.
-    * 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.
-    * demux the video specified data(frame_type, codec_id, ...) to sample.
-    * demux the h.264 sepcified data(avc_profile, ...) to codec from sequence header.
-    * demux the h.264 NALUs to sampe units.
-    */
-    virtual int video_avc_demux(char* data, int size, SrsCodecSample* sample);
+    virtual bool is_aac_sequence_header();
+    virtual bool is_avc_sequence_header();
 private:
-    virtual int video_nalu_demux(SrsBuffer* stream, SrsCodecSample* sample);
-public:
-    /**
-    * directly demux the sequence header, without RTMP packet header.
-    */
-    virtual int audio_aac_sequence_header_demux(char* data, int size);
+    // Demux the video packet in H.264 codec.
+    // The packet is muxed in FLV format, defined in flv specification.
+    //          Demux the sps/pps from sequence header.
+    //          Demux the samples from NALUs.
+    virtual int video_avc_demux(SrsBuffer* stream, int64_t timestamp);
 private:
-    /**
-    * when avc packet type is SrsCodecVideoAVCTypeSequenceHeader,
-    * decode the sps and pps.
-    */
+    // Parse the H.264 SPS/PPS.
     virtual int avc_demux_sps_pps(SrsBuffer* stream);
-    /**
-     * decode the sps rbsp stream.
-     */
     virtual int avc_demux_sps();
     virtual int avc_demux_sps_rbsp(char* rbsp, int nb_rbsp);
-    /**
-    * demux the avc NALU in "AnnexB" 
-    * from ISO_IEC_14496-10-AVC-2003.pdf, page 211.
-    */
-    virtual int avc_demux_annexb_format(SrsBuffer* stream, SrsCodecSample* sample);
-    /**
-    * demux the avc NALU in "ISO Base Media File Format" 
-    * from ISO_IEC_14496-15-AVC-format-2012.pdf, page 20
-    */
-    virtual int avc_demux_ibmf_format(SrsBuffer* stream, SrsCodecSample* sample);
+private:
+    // Parse the H.264 NALUs.
+    virtual int video_nalu_demux(SrsBuffer* stream);
+    // Demux the avc NALU in "AnnexB" from ISO_IEC_14496-10-AVC-2003.pdf, page 211.
+    virtual int avc_demux_annexb_format(SrsBuffer* stream);
+    // Demux the avc NALU in "ISO Base Media File Format" from ISO_IEC_14496-15-AVC-format-2012.pdf, page 20
+    virtual int avc_demux_ibmf_format(SrsBuffer* stream);
+private:
+    // Demux the audio packet in AAC codec.
+    //          Demux the asc from sequence header.
+    //          Demux the sampels from RAW data.
+    virtual int audio_aac_demux(SrsBuffer* stream, int64_t timestamp);
+    virtual int audio_mp3_demux(SrsBuffer* stream, int64_t timestamp);
+public:
+    // Directly demux the sequence header, without RTMP packet header.
+    virtual int audio_aac_sequence_header_demux(char* data, int size);
 };
 
 #endif
diff --git a/trunk/src/kernel/srs_kernel_mp3.cpp b/trunk/src/kernel/srs_kernel_mp3.cpp
index 7c1e9627e..f04cebd41 100644
--- a/trunk/src/kernel/srs_kernel_mp3.cpp
+++ b/trunk/src/kernel/srs_kernel_mp3.cpp
@@ -106,7 +106,6 @@ int SrsMp3Encoder::write_audio(int64_t timestamp, char* data, int size)
     // @see: E.4.2 Audio Tags, video_file_format_spec_v10_1.pdf, page 76
     int8_t sound_format = stream->read_1bytes();
     
-    // @see: SrsAvcAacCodec::audio_aac_demux
     //int8_t sound_type = sound_format & 0x01;
     //int8_t sound_size = (sound_format >> 1) & 0x01;
     //int8_t sound_rate = (sound_format >> 2) & 0x03;
diff --git a/trunk/src/kernel/srs_kernel_ts.cpp b/trunk/src/kernel/srs_kernel_ts.cpp
index d5dd3377a..b451b7f23 100644
--- a/trunk/src/kernel/srs_kernel_ts.cpp
+++ b/trunk/src/kernel/srs_kernel_ts.cpp
@@ -2708,7 +2708,7 @@ int SrsTsPayloadPMT::psi_encode(SrsBuffer* stream)
     return ret;
 }
 
-SrsTSMuxer::SrsTSMuxer(SrsFileWriter* w, SrsTsContext* c, SrsCodecAudio ac, SrsCodecVideo vc)
+SrsTsMuxer::SrsTsMuxer(SrsFileWriter* w, SrsTsContext* c, SrsCodecAudio ac, SrsCodecVideo vc)
 {
     writer = w;
     context = c;
@@ -2717,12 +2717,12 @@ SrsTSMuxer::SrsTSMuxer(SrsFileWriter* w, SrsTsContext* c, SrsCodecAudio ac, SrsC
     vcodec = vc;
 }
 
-SrsTSMuxer::~SrsTSMuxer()
+SrsTsMuxer::~SrsTsMuxer()
 {
     close();
 }
 
-int SrsTSMuxer::open(string p)
+int SrsTsMuxer::open(string p)
 {
     int ret = ERROR_SUCCESS;
     
@@ -2740,13 +2740,13 @@ int SrsTSMuxer::open(string p)
     return ret;
 }
 
-int SrsTSMuxer::update_acodec(SrsCodecAudio ac)
+int SrsTsMuxer::update_acodec(SrsCodecAudio ac)
 {
     acodec = ac;
     return ERROR_SUCCESS;
 }
 
-int SrsTSMuxer::write_audio(SrsTsMessage* audio)
+int SrsTsMuxer::write_audio(SrsTsMessage* audio)
 {
     int ret = ERROR_SUCCESS;
 
@@ -2762,7 +2762,7 @@ int SrsTSMuxer::write_audio(SrsTsMessage* audio)
     return ret;
 }
 
-int SrsTSMuxer::write_video(SrsTsMessage* video)
+int SrsTsMuxer::write_video(SrsTsMessage* video)
 {
     int ret = ERROR_SUCCESS;
 
@@ -2778,12 +2778,12 @@ int SrsTSMuxer::write_video(SrsTsMessage* video)
     return ret;
 }
 
-void SrsTSMuxer::close()
+void SrsTsMuxer::close()
 {
     writer->close();
 }
 
-SrsCodecVideo SrsTSMuxer::video_codec()
+SrsCodecVideo SrsTsMuxer::video_codec()
 {
     return vcodec;
 }
@@ -2800,7 +2800,7 @@ SrsTsCache::~SrsTsCache()
     srs_freep(video);
 }
     
-int SrsTsCache::cache_audio(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample* sample)
+int SrsTsCache::cache_audio(SrsAudioFrame* frame, int64_t dts)
 {
     int ret = ERROR_SUCCESS;
 
@@ -2817,16 +2817,16 @@ int SrsTsCache::cache_audio(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample*
     audio->sid = SrsTsPESStreamIdAudioCommon;
     
     // must be aac or mp3
-    SrsCodecAudio acodec = (SrsCodecAudio)codec->audio_codec_id;
-    srs_assert(acodec == SrsCodecAudioAAC || acodec == SrsCodecAudioMP3);
+    SrsAudioCodec* acodec = frame->acodec();
+    srs_assert(acodec->id == SrsCodecAudioAAC || acodec->id == SrsCodecAudioMP3);
     
     // write video to cache.
-    if (codec->audio_codec_id == SrsCodecAudioAAC) {
-        if ((ret = do_cache_aac(codec, sample)) != ERROR_SUCCESS) {
+    if (acodec->id == SrsCodecAudioAAC) {
+        if ((ret = do_cache_aac(frame)) != ERROR_SUCCESS) {
             return ret;
         }
     } else {
-        if ((ret = do_cache_mp3(codec, sample)) != ERROR_SUCCESS) {
+        if ((ret = do_cache_mp3(frame)) != ERROR_SUCCESS) {
             return ret;
         }
     }
@@ -2834,52 +2834,55 @@ int SrsTsCache::cache_audio(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample*
     return ret;
 }
     
-int SrsTsCache::cache_video(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample* sample)
+int SrsTsCache::cache_video(SrsVideoFrame* frame, int64_t dts)
 {
     int ret = ERROR_SUCCESS;
     
     // create the ts video message.
     if (!video) {
         video = new SrsTsMessage();
-        video->write_pcr = sample->frame_type == SrsCodecVideoAVCFrameKeyFrame;
+        video->write_pcr = (frame->frame_type == SrsCodecVideoAVCFrameKeyFrame);
         video->start_pts = dts;
     }
 
     video->dts = dts;
-    video->pts = video->dts + sample->cts * 90;
+    video->pts = video->dts + frame->cts * 90;
     video->sid = SrsTsPESStreamIdVideoCommon;
     
     // write video to cache.
-    if ((ret = do_cache_avc(codec, sample)) != ERROR_SUCCESS) {
+    if ((ret = do_cache_avc(frame)) != ERROR_SUCCESS) {
         return ret;
     }
 
     return ret;
 }
 
-int SrsTsCache::do_cache_mp3(SrsAvcAacCodec* codec, SrsCodecSample* sample)
+int SrsTsCache::do_cache_mp3(SrsAudioFrame* frame)
 {
     int ret = ERROR_SUCCESS;
         
     // for mp3, directly write to cache.
     // TODO: FIXME: implements the ts jitter.
-    for (int i = 0; i < sample->nb_sample_units; i++) {
-        SrsCodecSampleUnit* sample_unit = &sample->sample_units[i];
-        audio->payload->append(sample_unit->bytes, sample_unit->size);
+    for (int i = 0; i < frame->nb_samples; i++) {
+        SrsSample* sample = &frame->samples[i];
+        audio->payload->append(sample->bytes, sample->size);
     }
     
     return ret;
 }
 
-int SrsTsCache::do_cache_aac(SrsAvcAacCodec* codec, SrsCodecSample* sample)
+int SrsTsCache::do_cache_aac(SrsAudioFrame* frame)
 {
     int ret = ERROR_SUCCESS;
     
-    for (int i = 0; i < sample->nb_sample_units; i++) {
-        SrsCodecSampleUnit* sample_unit = &sample->sample_units[i];
-        int32_t size = sample_unit->size;
+    SrsAudioCodec* codec = frame->acodec();
+    srs_assert(codec);
+    
+    for (int i = 0; i < frame->nb_samples; i++) {
+        SrsSample* sample = &frame->samples[i];
+        int32_t size = sample->size;
         
-        if (!sample_unit->bytes || size <= 0 || size > 0x1fff) {
+        if (!sample->bytes || size <= 0 || size > 0x1fff) {
             ret = ERROR_HLS_AAC_FRAME_LENGTH;
             srs_error("invalid aac frame length=%d, ret=%d", size, ret);
             return ret;
@@ -2933,7 +2936,7 @@ int SrsTsCache::do_cache_aac(SrsAvcAacCodec* codec, SrsCodecSample* sample)
 
         // copy to audio buffer
         audio->payload->append((const char*)adts_header, sizeof(adts_header));
-        audio->payload->append(sample_unit->bytes, sample_unit->size);
+        audio->payload->append(sample->bytes, sample->size);
     }
     
     return ret;
@@ -2995,7 +2998,7 @@ void srs_avc_insert_aud(SrsSimpleStream* payload, bool& aud_inserted)
     }
 }
 
-int SrsTsCache::do_cache_avc(SrsAvcAacCodec* codec, SrsCodecSample* sample)
+int SrsTsCache::do_cache_avc(SrsVideoFrame* frame)
 {
     int ret = ERROR_SUCCESS;
     
@@ -3003,7 +3006,7 @@ int SrsTsCache::do_cache_avc(SrsAvcAacCodec* codec, SrsCodecSample* sample)
     bool aud_inserted = false;
     
     // Insert a default AUD NALU when no AUD in samples.
-    if (!sample->has_aud) {
+    if (!frame->has_aud) {
         // the aud(access unit delimiter) before each frame.
         // 7.3.2.4 Access unit delimiter RBSP syntax
         // ISO_IEC_14496-10-AVC-2012.pdf, page 66.
@@ -3039,12 +3042,15 @@ int SrsTsCache::do_cache_avc(SrsAvcAacCodec* codec, SrsCodecSample* sample)
         video->payload->append((const char*)default_aud_nalu, 2);
     }
     
+    SrsVideoCodec* codec = frame->vcodec();
+    srs_assert(codec);
+    
     // all sample use cont nalu header, except the sps-pps before IDR frame.
-    for (int i = 0; i < sample->nb_sample_units; i++) {
-        SrsCodecSampleUnit* sample_unit = &sample->sample_units[i];
-        int32_t size = sample_unit->size;
+    for (int i = 0; i < frame->nb_samples; i++) {
+        SrsSample* sample = &frame->samples[i];
+        int32_t size = sample->size;
         
-        if (!sample_unit->bytes || size <= 0) {
+        if (!sample->bytes || size <= 0) {
             ret = ERROR_HLS_AVC_SAMPLE_SIZE;
             srs_error("invalid avc sample length=%d, ret=%d", size, ret);
             return ret;
@@ -3052,11 +3058,11 @@ int SrsTsCache::do_cache_avc(SrsAvcAacCodec* codec, SrsCodecSample* sample)
         
         // 5bits, 7.3.1 NAL unit syntax,
         // ISO_IEC_14496-10-AVC-2012.pdf, page 83.
-        SrsAvcNaluType nal_unit_type = (SrsAvcNaluType)(sample_unit->bytes[0] & 0x1f);
+        SrsAvcNaluType nal_unit_type = (SrsAvcNaluType)(sample->bytes[0] & 0x1f);
         
         // Insert sps/pps before IDR when there is no sps/pps in samples.
         // The sps/pps is parsed from sequence header(generally the first flv packet).
-        if (nal_unit_type == SrsAvcNaluTypeIDR && !sample->has_sps_pps) {
+        if (nal_unit_type == SrsAvcNaluTypeIDR && !frame->has_sps_pps) {
             if (codec->sequenceParameterSetLength > 0) {
                 srs_avc_insert_aud(video->payload, aud_inserted);
                 video->payload->append(codec->sequenceParameterSetNALUnit, codec->sequenceParameterSetLength);
@@ -3069,7 +3075,7 @@ int SrsTsCache::do_cache_avc(SrsAvcAacCodec* codec, SrsCodecSample* sample)
         
         // Insert the NALU to video in annexb.
         srs_avc_insert_aud(video->payload, aud_inserted);
-        video->payload->append(sample_unit->bytes, sample_unit->size);
+        video->payload->append(sample->bytes, sample->size);
     }
     
     return ret;
@@ -3078,8 +3084,7 @@ int SrsTsCache::do_cache_avc(SrsAvcAacCodec* codec, SrsCodecSample* sample)
 SrsTsEncoder::SrsTsEncoder()
 {
     writer = NULL;
-    codec = new SrsAvcAacCodec();
-    sample = new SrsCodecSample();
+    format = new SrsFormat();
     cache = new SrsTsCache();
     context = new SrsTsContext();
     muxer = NULL;
@@ -3087,8 +3092,7 @@ SrsTsEncoder::SrsTsEncoder()
 
 SrsTsEncoder::~SrsTsEncoder()
 {
-    srs_freep(codec);
-    srs_freep(sample);
+    srs_freep(format);
     srs_freep(cache);
     srs_freep(muxer);
     srs_freep(context);
@@ -3098,6 +3102,10 @@ int SrsTsEncoder::initialize(SrsFileWriter* fw)
 {
     int ret = ERROR_SUCCESS;
     
+    if ((ret = format->initialize()) != ERROR_SUCCESS) {
+        return ret;
+    }
+    
     srs_assert(fw);
     
     if (!fw->is_open()) {
@@ -3109,7 +3117,7 @@ int SrsTsEncoder::initialize(SrsFileWriter* fw)
     writer = fw;
 
     srs_freep(muxer);
-    muxer = new SrsTSMuxer(fw, context, SrsCodecAudioAAC, SrsCodecVideoAVC);
+    muxer = new SrsTsMuxer(fw, context, SrsCodecAudioAAC, SrsCodecVideoAVC);
 
     if ((ret = muxer->open("")) != ERROR_SUCCESS) {
         return ret;
@@ -3122,32 +3130,24 @@ int SrsTsEncoder::write_audio(int64_t timestamp, char* data, int size)
 {
     int ret = ERROR_SUCCESS;
     
-    sample->clear();
-    if ((ret = codec->audio_aac_demux(data, size, sample)) != ERROR_SUCCESS) {
-        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;
-        }
+    if ((ret = format->on_audio(timestamp, data, size)) != ERROR_SUCCESS) {
+        return ret;
     }
-    SrsCodecAudio acodec = (SrsCodecAudio)codec->audio_codec_id;
     
     // ts support audio codec: aac/mp3
-    if (acodec != SrsCodecAudioAAC && acodec != SrsCodecAudioMP3) {
+    srs_assert(format->acodec && format->audio);
+    if (format->acodec->id != SrsCodecAudioAAC && format->acodec->id != SrsCodecAudioMP3) {
         return ret;
     }
 
     // when codec changed, write new header.
-    if ((ret = muxer->update_acodec(acodec)) != ERROR_SUCCESS) {
+    if ((ret = muxer->update_acodec(format->acodec->id)) != ERROR_SUCCESS) {
         srs_error("http: ts audio write header failed. ret=%d", ret);
         return ret;
     }
     
     // for aac: ignore sequence header
-    if (acodec == SrsCodecAudioAAC && sample->aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
+    if (format->acodec->id == SrsCodecAudioAAC && format->audio->aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
         return ret;
     }
 
@@ -3157,7 +3157,7 @@ int SrsTsEncoder::write_audio(int64_t timestamp, char* data, int size)
     int64_t dts = timestamp * 90;
     
     // write audio to cache.
-    if ((ret = cache->cache_audio(codec, dts, sample)) != ERROR_SUCCESS) {
+    if ((ret = cache->cache_audio(format->audio, dts)) != ERROR_SUCCESS) {
         return ret;
     }
     
@@ -3172,32 +3172,31 @@ int SrsTsEncoder::write_video(int64_t timestamp, char* data, int size)
 {
     int ret = ERROR_SUCCESS;
     
-    sample->clear();
-    if ((ret = codec->video_avc_demux(data, size, sample)) != ERROR_SUCCESS) {
-        srs_error("http: ts codec demux video failed. ret=%d", ret);
+    if ((ret = format->on_video(timestamp, data, size)) != ERROR_SUCCESS) {
         return ret;
     }
     
     // ignore info frame,
     // @see https://github.com/ossrs/srs/issues/288#issuecomment-69863909
-    if (sample->frame_type == SrsCodecVideoAVCFrameVideoInfoFrame) {
+    srs_assert(format->video && format->vcodec);
+    if (format->video->frame_type == SrsCodecVideoAVCFrameVideoInfoFrame) {
         return ret;
     }
     
-    if (codec->video_codec_id != SrsCodecVideoAVC) {
+    if (format->vcodec->id != SrsCodecVideoAVC) {
         return ret;
     }
     
     // ignore sequence header
-    if (sample->frame_type == SrsCodecVideoAVCFrameKeyFrame
-         && sample->avc_packet_type == SrsCodecVideoAVCTypeSequenceHeader) {
+    if (format->video->frame_type == SrsCodecVideoAVCFrameKeyFrame
+         && format->video->avc_packet_type == SrsCodecVideoAVCTypeSequenceHeader) {
         return ret;
     }
     
     int64_t dts = timestamp * 90;
     
     // write video to cache.
-    if ((ret = cache->cache_video(codec, dts, sample)) != ERROR_SUCCESS) {
+    if ((ret = cache->cache_video(format->video, dts)) != ERROR_SUCCESS) {
         return ret;
     }
 
diff --git a/trunk/src/kernel/srs_kernel_ts.hpp b/trunk/src/kernel/srs_kernel_ts.hpp
index 3276c5031..8382527ff 100644
--- a/trunk/src/kernel/srs_kernel_ts.hpp
+++ b/trunk/src/kernel/srs_kernel_ts.hpp
@@ -42,8 +42,7 @@ class SrsTsCache;
 class SrsTSMuxer;
 class SrsFileWriter;
 class SrsFileReader;
-class SrsAvcAacCodec;
-class SrsCodecSample;
+class SrsFormat;
 class SrsSimpleStream;
 class SrsTsAdaptationField;
 class SrsTsPayload;
@@ -1560,7 +1559,7 @@ protected:
 * write data from frame(header info) and buffer(data) to ts file.
 * it's a simple object wrapper for utility from nginx-rtmp: SrsMpegtsWriter
 */
-class SrsTSMuxer
+class SrsTsMuxer
 {
 private:
     SrsCodecVideo vcodec;
@@ -1570,8 +1569,8 @@ private:
     SrsFileWriter* writer;
     std::string path;
 public:
-    SrsTSMuxer(SrsFileWriter* w, SrsTsContext* c, SrsCodecAudio ac, SrsCodecVideo vc);
-    virtual ~SrsTSMuxer();
+    SrsTsMuxer(SrsFileWriter* w, SrsTsContext* c, SrsCodecAudio ac, SrsCodecVideo vc);
+    virtual ~SrsTsMuxer();
 public:
     /**
      * open the writer, donot write the PSI of ts.
@@ -1585,6 +1584,7 @@ public:
     * @remark for audio aac codec, for example, SRS1, it's ok to write PSI when open ts.
     * @see https://github.com/ossrs/srs/issues/301
     */
+    // TODO: FIXME: Remove it.
     virtual int update_acodec(SrsCodecAudio ac);
     /**
     * write an audio frame to ts, 
@@ -1628,29 +1628,29 @@ public:
     /**
     * write audio to cache
     */
-    virtual int cache_audio(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample* sample);
+    virtual int cache_audio(SrsAudioFrame* frame, int64_t dts);
     /**
     * write video to muxer.
     */
-    virtual int cache_video(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample* sample);
+    virtual int cache_video(SrsVideoFrame* frame, int64_t dts);
 private:
-    virtual int do_cache_mp3(SrsAvcAacCodec* codec, SrsCodecSample* sample);
-    virtual int do_cache_aac(SrsAvcAacCodec* codec, SrsCodecSample* sample);
-    virtual int do_cache_avc(SrsAvcAacCodec* codec, SrsCodecSample* sample);
+    virtual int do_cache_mp3(SrsAudioFrame* frame);
+    virtual int do_cache_aac(SrsAudioFrame* frame);
+    virtual int do_cache_avc(SrsVideoFrame* frame);
 };
 
 /**
 * encode data to ts file.
 */
+// TODO: FIXME: Rename it.
 class SrsTsEncoder
 {
 private:
     SrsFileWriter* writer;
 private:
-    SrsAvcAacCodec* codec;
-    SrsCodecSample* sample;
+    SrsFormat* format;
     SrsTsCache* cache;
-    SrsTSMuxer* muxer;
+    SrsTsMuxer* muxer;
     SrsTsContext* context;
 public:
     SrsTsEncoder();
diff --git a/trunk/src/protocol/srs_protocol_format.cpp b/trunk/src/protocol/srs_protocol_format.cpp
index 75425233f..7bbf4164a 100644
--- a/trunk/src/protocol/srs_protocol_format.cpp
+++ b/trunk/src/protocol/srs_protocol_format.cpp
@@ -26,24 +26,19 @@
 #include <srs_kernel_error.hpp>
 #include <srs_kernel_codec.hpp>
 #include <srs_rtmp_stack.hpp>
+#include <srs_kernel_buffer.hpp>
+#include <srs_core_autofree.hpp>
+#include <srs_kernel_utility.hpp>
 
-SrsFormat::SrsFormat()
+SrsRtmpFormat::SrsRtmpFormat()
 {
-    audio = video = NULL;
 }
 
-SrsFormat::~SrsFormat()
+SrsRtmpFormat::~SrsRtmpFormat()
 {
-    srs_freep(audio);
-    srs_freep(video);
 }
 
-int SrsFormat::initialize()
-{
-    return ERROR_SUCCESS;
-}
-
-int SrsFormat::on_metadata(SrsOnMetaDataPacket* meta)
+int SrsRtmpFormat::on_metadata(SrsOnMetaDataPacket* meta)
 {
     int ret = ERROR_SUCCESS;
     
@@ -52,15 +47,31 @@ int SrsFormat::on_metadata(SrsOnMetaDataPacket* meta)
     return ret;
 }
 
-int SrsFormat::on_audio(SrsSharedPtrMessage* shared_audio)
+int SrsRtmpFormat::on_audio(SrsSharedPtrMessage* shared_audio)
 {
-    int ret = ERROR_SUCCESS;
-    return ret;
+    SrsSharedPtrMessage* msg = shared_audio;
+    char* data = msg->payload;
+    int size = msg->size;
+    
+    return SrsFormat::on_audio(msg->timestamp, data, size);
 }
 
-int SrsFormat::on_video(SrsSharedPtrMessage* shared_video, bool is_sequence_header)
+int SrsRtmpFormat::on_audio(int64_t timestamp, char* data, int size)
 {
-    int ret = ERROR_SUCCESS;
-    return ret;
+    return SrsFormat::on_audio(timestamp, data, size);
+}
+
+int SrsRtmpFormat::on_video(SrsSharedPtrMessage* shared_video)
+{
+    SrsSharedPtrMessage* msg = shared_video;
+    char* data = msg->payload;
+    int size = msg->size;
+    
+    return SrsFormat::on_video(msg->timestamp, data, size);
+}
+
+int SrsRtmpFormat::on_video(int64_t timestamp, char* data, int size)
+{
+    return SrsFormat::on_video(timestamp, data, size);
 }
 
diff --git a/trunk/src/protocol/srs_protocol_format.hpp b/trunk/src/protocol/srs_protocol_format.hpp
index d3f61aa22..be01943ba 100644
--- a/trunk/src/protocol/srs_protocol_format.hpp
+++ b/trunk/src/protocol/srs_protocol_format.hpp
@@ -30,32 +30,33 @@
 
 #include <srs_core.hpp>
 
-class SrsFrame;
+#include <srs_kernel_codec.hpp>
+
+class SrsBuffer;
+class SrsAudioFrame;
+class SrsVideoFrame;
+class SrsAudioCodec;
+class SrsVideoCodec;
 class SrsOnMetaDataPacket;
 class SrsSharedPtrMessage;
 
 /**
- * A codec format, including one or many stream, each stream identified by a frame.
- * For example, a typical RTMP stream format, consits of a video and audio frame.
- * Maybe some RTMP stream only has a audio stream, for instance, redio application.
+ * Create special structure from RTMP stream, for example, the metadata.
  */
-class SrsFormat
+class SrsRtmpFormat : public SrsFormat
 {
 public:
-    SrsFrame* audio;
-    SrsFrame* video;
+    SrsRtmpFormat();
+    virtual ~SrsRtmpFormat();
 public:
-    SrsFormat();
-    virtual ~SrsFormat();
-public:
-    // Initialize the format.
-    virtual int initialize();
     // Initialize the format from metadata, optional.
     virtual int on_metadata(SrsOnMetaDataPacket* meta);
     // When got a parsed audio packet.
     virtual int on_audio(SrsSharedPtrMessage* shared_audio);
+    virtual int on_audio(int64_t timestamp, char* data, int size);
     // When got a parsed video packet.
-    virtual int on_video(SrsSharedPtrMessage* shared_video, bool is_sequence_header);
+    virtual int on_video(SrsSharedPtrMessage* shared_video);
+    virtual int on_video(int64_t timestamp, char* data, int size);
 };
 
 #endif
diff --git a/trunk/src/protocol/srs_rtsp_stack.cpp b/trunk/src/protocol/srs_rtsp_stack.cpp
index 7c2ec3c62..de46b6844 100644
--- a/trunk/src/protocol/srs_rtsp_stack.cpp
+++ b/trunk/src/protocol/srs_rtsp_stack.cpp
@@ -136,7 +136,7 @@ SrsRtpPacket::SrsRtpPacket()
     ssrc = 0;
 
     payload = new SrsSimpleStream();
-    audio_samples = new SrsCodecSample();
+    audio = new SrsAudioFrame();
     chunked = false;
     completed = false;
 }
@@ -144,7 +144,7 @@ SrsRtpPacket::SrsRtpPacket()
 SrsRtpPacket::~SrsRtpPacket()
 {
     srs_freep(payload);
-    srs_freep(audio_samples);
+    srs_freep(audio);
 }
 
 void SrsRtpPacket::copy(SrsRtpPacket* src)
@@ -161,7 +161,9 @@ void SrsRtpPacket::copy(SrsRtpPacket* src)
 
     chunked = src->chunked;
     completed = src->completed;
-    audio_samples = new SrsCodecSample();
+    
+    srs_freep(audio);
+    audio = new SrsAudioFrame();
 }
 
 void SrsRtpPacket::reap(SrsRtpPacket* src)
@@ -172,9 +174,9 @@ void SrsRtpPacket::reap(SrsRtpPacket* src)
     payload = src->payload;
     src->payload = NULL;
     
-    srs_freep(audio_samples);
-    audio_samples = src->audio_samples;
-    src->audio_samples = NULL;
+    srs_freep(audio);
+    audio = src->audio;
+    src->audio = NULL;
 }
 
 int SrsRtpPacket::decode(SrsBuffer* stream)
@@ -263,7 +265,7 @@ int SrsRtpPacket::decode_97(SrsBuffer* stream)
             return ret;
         }
 
-        if ((ret = audio_samples->add_sample_unit(sample, sample_size)) != ERROR_SUCCESS) {
+        if ((ret = audio->add_sample(sample, sample_size)) != ERROR_SUCCESS) {
             srs_error("rtsp: rtp type97 add sample failed. ret=%d", ret);
             return ret;
         }
diff --git a/trunk/src/protocol/srs_rtsp_stack.hpp b/trunk/src/protocol/srs_rtsp_stack.hpp
index 16dae3f8e..4a77d0433 100644
--- a/trunk/src/protocol/srs_rtsp_stack.hpp
+++ b/trunk/src/protocol/srs_rtsp_stack.hpp
@@ -41,7 +41,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 
 class SrsBuffer;
 class SrsSimpleStream;
-class SrsCodecSample;
+class SrsAudioFrame;
 class ISrsProtocolReaderWriter;
 
 // rtsp specification
@@ -299,7 +299,7 @@ public:
     /**
     * the audio samples, one rtp packets may contains multiple audio samples.
     */
-    SrsCodecSample* audio_samples;
+    SrsAudioFrame* audio;
 public:
     SrsRtpPacket();
     virtual ~SrsRtpPacket();