fix #124, gop cache support disable video in publishing. 0.9.171.

README.md

@@ -209,7 +209,8 @@ Supported operating systems and hardware:
 * 2013-10-17, Created.<br/>
 
 ## History
-* v1.0, 2014-07-19, fix [#121](https://github.com/winlinvip/simple-rtmp-server/issues/121), srs_info detail log compile failed. 0.9.168.
+* v1.0, 2014-07-26, fix [#124](https://github.com/winlinvip/simple-rtmp-server/issues/124), gop cache support disable video in publishing. 0.9.171.
+* v1.0, 2014-07-23, fix [#121](https://github.com/winlinvip/simple-rtmp-server/issues/121), srs_info detail log compile failed. 0.9.168.
 * v1.0, 2014-07-19, fix [#119](https://github.com/winlinvip/simple-rtmp-server/issues/119), use iformat and oformat for ffmpeg transcode. 0.9.163.
 * <strong>v1.0, 2014-07-13, [1.0 mainline6(0.9.160)](https://github.com/winlinvip/simple-rtmp-server/releases/tag/1.0.mainline8) released. 50029 lines.</strong>
 * v1.0, 2014-07-13, refine the bandwidth check/test, add as/js library, use srs-librtmp for linux tool. 0.9.159

trunk/src/app/srs_app_source.cpp

@@ -46,6 +46,10 @@ using namespace std;
 #define CONST_MAX_JITTER_MS         500
 #define DEFAULT_FRAME_TIME_MS         40
 
+// for 26ms per audio packet,
+// 115 packets is 3s.
+#define __SRS_PURE_AUDIO_GUESS_COUNT 115
+
 int _srs_time_jitter_string2int(std::string time_jitter)
 {
     if (time_jitter == "full") {
@@ -351,6 +355,7 @@ SrsGopCache::SrsGopCache()
 {
     cached_video_count = 0;
     enable_gop_cache = true;
+    audio_count_after_last_video = 0;
 }
 
 SrsGopCache::~SrsGopCache()
@@ -383,6 +388,7 @@ int SrsGopCache::cache(SrsSharedPtrMessage* msg)
     // got video, update the video count if acceptable
     if (msg->header.is_video()) {
         cached_video_count++;
+        audio_count_after_last_video = 0;
     }
     
     // no acceptable video or pure audio, disable the cache.
@@ -391,6 +397,18 @@ int SrsGopCache::cache(SrsSharedPtrMessage* msg)
         return ret;
     }
     
+    // ok, gop cache enabled, and got an audio.
+    if (msg->header.is_audio()) {
+        audio_count_after_last_video++;
+    }
+    
+    // clear gop cache when pure audio count overflow
+    if (audio_count_after_last_video > __SRS_PURE_AUDIO_GUESS_COUNT) {
+        srs_warn("clear gop cache for guess pure audio overflow");
+        clear();
+        return ret;
+    }
+    
     // clear gop cache when got key frame
     if (msg->header.is_video() && SrsFlvCodec::video_is_keyframe(msg->payload, msg->size)) {
         srs_info("clear gop cache when got keyframe. vcount=%d, count=%d",

trunk/src/app/srs_app_source.hpp

@@ -213,6 +213,19 @@ private:
     */
     int cached_video_count;
     /**
+    * when user disabled video when publishing, and gop cache enalbed,
+    * we will cache the audio/video for we already got video, but we never
+    * know when to clear the gop cache, for there is no video in future,
+    * so we must guess whether user disabled the video.
+    * when we got some audios after laster video, for instance, 600 audio packets,
+    * about 3s(26ms per packet) 115 audio packets, clear gop cache.
+    * 
+    * @remark, it is ok for performance, for when we clear the gop cache,
+    *       gop cache is disabled for pure audio stream.
+    * @see: https://github.com/winlinvip/simple-rtmp-server/issues/124
+    */
+    int audio_count_after_last_video;
+    /**
     * cached gop.
     */
     std::vector<SrsSharedPtrMessage*> gop_cache;

trunk/src/core/srs_core.hpp

@@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 // current release version
 #define VERSION_MAJOR "0"
 #define VERSION_MINOR "9"
-#define VERSION_REVISION "170"
+#define VERSION_REVISION "171"
 #define RTMP_SIG_SRS_VERSION VERSION_MAJOR"."VERSION_MINOR"."VERSION_REVISION
 // server info.
 #define RTMP_SIG_SRS_KEY "SRS"