fix #151, always reap ts whatever audio or video packet. 0.9.223.

@@ -208,6 +208,7 @@ Supported operating systems and hardware:
 * 2013-10-17, Created.<br/>
 * 2013-10-17, Created.<br/>
 
 
 ## History
 ## History
+* v1.0, 2014-10-08, fix [#151](https://github.com/winlinvip/simple-rtmp-server/issues/151), always reap ts whatever audio or video packet. 0.9.223.
 * v1.0, 2014-10-08, fix [#162](https://github.com/winlinvip/simple-rtmp-server/issues/162), failed if no epoll. 0.9.222.
 * v1.0, 2014-10-08, fix [#162](https://github.com/winlinvip/simple-rtmp-server/issues/162), failed if no epoll. 0.9.222.
 * v1.0, 2014-09-30, fix [#180](https://github.com/winlinvip/simple-rtmp-server/issues/180), crash for multiple edge publishing the same stream. 0.9.220.
 * v1.0, 2014-09-30, fix [#180](https://github.com/winlinvip/simple-rtmp-server/issues/180), crash for multiple edge publishing the same stream. 0.9.220.
 * v1.0, 2014-09-26, fix hls bug, refine config and log, according to clion of jetbrains. 0.9.216. 
 * v1.0, 2014-09-26, fix hls bug, refine config and log, according to clion of jetbrains. 0.9.216. 

@@ -135,6 +135,8 @@ u_int8_t mpegts_header[] = {
     /* PMT */
     /* PMT */
     0xe1, 0x00,
     0xe1, 0x00,
     0xf0, 0x00,
     0xf0, 0x00,
+    // must generate header with/without video, @see:
+    // https://github.com/winlinvip/simple-rtmp-server/issues/40
     0x1b, 0xe1, 0x00, 0xf0, 0x00, /* h264, pid=0x100=256 */
     0x1b, 0xe1, 0x00, 0xf0, 0x00, /* h264, pid=0x100=256 */
     0x0f, 0xe1, 0x01, 0xf0, 0x00, /* aac, pid=0x101=257 */
     0x0f, 0xe1, 0x01, 0xf0, 0x00, /* aac, pid=0x101=257 */
     /*0x03, 0xe1, 0x01, 0xf0, 0x00,*/ /* mp3 */
     /*0x03, 0xe1, 0x01, 0xf0, 0x00,*/ /* mp3 */
@@ -967,8 +969,6 @@ SrsHlsCache::SrsHlsCache()
     
     
     af = new SrsMpegtsFrame();
     af = new SrsMpegtsFrame();
     vf = new SrsMpegtsFrame();
     vf = new SrsMpegtsFrame();
-
-    video_count = 0;
 }
 }
 
 
 SrsHlsCache::~SrsHlsCache()
 SrsHlsCache::~SrsHlsCache()
@@ -999,9 +999,6 @@ int SrsHlsCache::on_publish(SrsHlsMuxer* muxer, SrsRequest* req, int64_t segment
     // get the hls path config
     // get the hls path config
     std::string hls_path = _srs_config->get_hls_path(vhost);
     std::string hls_path = _srs_config->get_hls_path(vhost);
     
     
-    // reset video count for new publish session.
-    video_count = 0;
-    
     // TODO: FIXME: support load exists m3u8, to continue publish stream.
     // TODO: FIXME: support load exists m3u8, to continue publish stream.
     // for the HLS donot requires the EXT-X-MEDIA-SEQUENCE be monotonically increase.
     // for the HLS donot requires the EXT-X-MEDIA-SEQUENCE be monotonically increase.
     
     
@@ -1082,9 +1079,13 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
         }
         }
     }
     }
     
     
-    // for pure audio
-    // start new segment when duration overflow.
-    if (video_count == 0 && muxer->is_segment_overflow()) {
+    // reap when current source is pure audio.
+    // it maybe changed when stream info changed,
+    // for example, pure audio when start, audio/video when publishing,
+    // pure audio again for audio disabled.
+    // so we reap event when the audio incoming when segment overflow.
+    // @see https://github.com/winlinvip/simple-rtmp-server/issues/151
+    if (muxer->is_segment_overflow()) {
         if ((ret = reap_segment("audio", muxer, af->pts)) != ERROR_SUCCESS) {
         if ((ret = reap_segment("audio", muxer, af->pts)) != ERROR_SUCCESS) {
             return ret;
             return ret;
         }
         }
@@ -1098,8 +1099,6 @@ int SrsHlsCache::write_video(
 {
 {
     int ret = ERROR_SUCCESS;
     int ret = ERROR_SUCCESS;
     
     
-    video_count++;
-    
     // write video to cache.
     // write video to cache.
     if ((ret = cache_video(codec, sample)) != ERROR_SUCCESS) {
     if ((ret = cache_video(codec, sample)) != ERROR_SUCCESS) {
         return ret;
         return ret;

@@ -239,14 +239,6 @@ private:
     int64_t audio_buffer_start_pts;
     int64_t audio_buffer_start_pts;
     // time jitter for aac
     // time jitter for aac
     SrsHlsAacJitter* aac_jitter;
     SrsHlsAacJitter* aac_jitter;
-private:
-    /**
-    * for pure audio HLS application,
-    * the video count used to count the video,
-    * if zero and audio buffer overflow, reap the ts,
-    * just like we got a keyframe.
-    */
-    u_int32_t video_count;
 public:
 public:
     SrsHlsCache();
     SrsHlsCache();
     virtual ~SrsHlsCache();
     virtual ~SrsHlsCache();

@@ -400,7 +400,7 @@ int SrsGopCache::cache(SrsSharedPtrMessage* msg)
     }
     }
     
     
     // no acceptable video or pure audio, disable the cache.
     // no acceptable video or pure audio, disable the cache.
-    if (cached_video_count == 0) {
+    if (pure_audio()) {
         srs_verbose("ignore any frame util got a h264 video frame.");
         srs_verbose("ignore any frame util got a h264 video frame.");
         return ret;
         return ret;
     }
     }
@@ -482,6 +482,11 @@ int64_t SrsGopCache::start_time()
     return msg->header.timestamp;
     return msg->header.timestamp;
 }
 }
 
 
+bool SrsGopCache::pure_audio()
+{
+    return cached_video_count == 0;
+}
+
 std::map<std::string, SrsSource*> SrsSource::pool;
 std::map<std::string, SrsSource*> SrsSource::pool;
 
 
 int SrsSource::find(SrsRequest* req, SrsSource** ppsource)
 int SrsSource::find(SrsRequest* req, SrsSource** ppsource)

@@ -263,6 +263,11 @@ public:
     * @return 0 if no packets.
     * @return 0 if no packets.
     */
     */
     virtual int64_t start_time();
     virtual int64_t start_time();
+    /**
+    * whether current stream is pure audio,
+    * when no video in gop cache, the stream is pure audio right now.
+    */
+    virtual bool pure_audio();
 };
 };
 
 
 /**
 /**

@@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 // current release version
 // current release version
 #define VERSION_MAJOR "0"
 #define VERSION_MAJOR "0"
 #define VERSION_MINOR "9"
 #define VERSION_MINOR "9"
-#define VERSION_REVISION "222"
+#define VERSION_REVISION "223"
 #define RTMP_SIG_SRS_VERSION VERSION_MAJOR"."VERSION_MINOR"."VERSION_REVISION
 #define RTMP_SIG_SRS_VERSION VERSION_MAJOR"."VERSION_MINOR"."VERSION_REVISION
 // server info.
 // server info.
 #define RTMP_SIG_SRS_KEY "SRS"
 #define RTMP_SIG_SRS_KEY "SRS"