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

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winlin 2014-07-26 12:22:39 +08:00
parent 50cd1d2360
commit b17c736f3f
4 changed files with 34 additions and 2 deletions
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3
README.md
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@ -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

18
trunk/src/app/srs_app_source.cpp
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@ -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",

13
trunk/src/app/srs_app_source.hpp
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@ -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;

2
trunk/src/core/srs_core.hpp
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@ -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"