refine code, extract the rtmp jitter for hls

@@ -41,6 +41,7 @@ SrsHLS::SrsHLS()
 	codec = new SrsCodec();
 	sample = new SrsCodecSample();
 	muxer = NULL;
+	jitter = new SrsRtmpJitter();
 }
 
 SrsHLS::~SrsHLS()
@@ -48,6 +49,7 @@ SrsHLS::~SrsHLS()
 	srs_freep(codec);
 	srs_freep(sample);
 	srs_freep(muxer);
+	srs_freep(jitter);
 }
 
 int SrsHLS::on_publish(std::string _vhost)

@@ -38,6 +38,7 @@ class SrsCodecBuffer;
 class SrsMpegtsFrame;
 class SrsTSMuxer;
 class SrsCodec;
+class SrsRtmpJitter;
 
 class SrsHLS
 {
@@ -47,6 +48,7 @@ private:
 	SrsCodec* codec;
 	SrsCodecSample* sample;
 	SrsTSMuxer* muxer;
+	SrsRtmpJitter* jitter;
 public:
 	SrsHLS();
 	virtual ~SrsHLS();

@@ -36,23 +36,72 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #define DEFAULT_FRAME_TIME_MS 		10
 #define PAUSED_SHRINK_SIZE			250
 
-std::map<std::string, SrsSource*> SrsSource::pool;
+SrsRtmpJitter::SrsRtmpJitter()
+{
+	last_pkt_correct_time = last_pkt_time = 0;
+}
 
-SrsSource* SrsSource::find(std::string stream_url)
+SrsRtmpJitter::~SrsRtmpJitter()
 {
-	if (pool.find(stream_url) == pool.end()) {
-		pool[stream_url] = new SrsSource(stream_url);
-		srs_verbose("create new source for url=%s", stream_url.c_str());
+}
+
+int SrsRtmpJitter::correct(SrsSharedPtrMessage* msg, int audio_sample_rate, int video_frame_rate)
+{
+	int ret = ERROR_SUCCESS;
+	
+	/**
+	* we use a very simple time jitter detect/correct algorithm:
+	* 1. delta: ensure the delta is positive and valid,
+	* 	we set the delta to DEFAULT_FRAME_TIME_MS,
+	* 	if the delta of time is nagative or greater than CONST_MAX_JITTER_MS.
+	* 2. last_pkt_time: specifies the original packet time,
+	* 	is used to detect next jitter.
+	* 3. last_pkt_correct_time: simply add the positive delta, 
+	* 	and enforce the time monotonically.
+	*/
+	u_int32_t time = msg->header.timestamp;
+	int32_t delta = time - last_pkt_time;
+
+	// if jitter detected, reset the delta.
+	if (delta < 0 || delta > CONST_MAX_JITTER_MS) {
+		// calc the right diff by audio sample rate
+		if (msg->header.is_audio() && audio_sample_rate > 0) {
+			delta = (int32_t)(delta * 1000.0 / audio_sample_rate);
+		} else if (msg->header.is_video() && video_frame_rate > 0) {
+			delta = (int32_t)(delta * 1.0 / video_frame_rate);
+		} else {
+			delta = DEFAULT_FRAME_TIME_MS;
+		}
+
+		// sometimes, the time is absolute time, so correct it again.
+		if (delta < 0 || delta > CONST_MAX_JITTER_MS) {
+			delta = DEFAULT_FRAME_TIME_MS;
+		}
+		
+		srs_info("jitter detected, last_pts=%d, pts=%d, diff=%d, last_time=%d, time=%d, diff=%d",
+			last_pkt_time, time, time - last_pkt_time, last_pkt_correct_time, last_pkt_correct_time + delta, delta);
+	} else {
+		srs_verbose("timestamp no jitter. time=%d, last_pkt=%d, correct_to=%d", 
+			time, last_pkt_time, last_pkt_correct_time + delta);
 	}
 	
-	return pool[stream_url];
+	last_pkt_correct_time = srs_max(0, last_pkt_correct_time + delta);
+	msg->header.timestamp = last_pkt_correct_time;
+	last_pkt_time = time;
+	
+	return ret;
+}
+
+int SrsRtmpJitter::get_time()
+{
+	return (int)last_pkt_correct_time;
 }
 
 SrsConsumer::SrsConsumer(SrsSource* _source)
 {
 	source = _source;
-	last_pkt_correct_time = last_pkt_time = 0;
 	paused = false;
+	jitter = new SrsRtmpJitter();
 }
 
 SrsConsumer::~SrsConsumer()
@@ -60,18 +109,19 @@ SrsConsumer::~SrsConsumer()
 	clear();
 	
 	source->on_consumer_destroy(this);
+	srs_freep(jitter);
 }
 
 int SrsConsumer::get_time()
 {
-	return (int)last_pkt_correct_time;
+	return jitter->get_time();
 }
 
 int SrsConsumer::enqueue(SrsSharedPtrMessage* msg, int audio_sample_rate, int video_frame_rate)
 {
 	int ret = ERROR_SUCCESS;
 	
-	if ((ret = jitter_correct(msg, audio_sample_rate, video_frame_rate)) != ERROR_SUCCESS) {
+	if ((ret = jitter->correct(msg, audio_sample_rate, video_frame_rate)) != ERROR_SUCCESS) {
 		return ret;
 	}
 	
@@ -178,53 +228,6 @@ void SrsConsumer::shrink()
 	}
 }
 
-int SrsConsumer::jitter_correct(SrsSharedPtrMessage* msg, int audio_sample_rate, int video_frame_rate)
-{
-	int ret = ERROR_SUCCESS;
-	
-	/**
-	* we use a very simple time jitter detect/correct algorithm:
-	* 1. delta: ensure the delta is positive and valid,
-	* 	we set the delta to DEFAULT_FRAME_TIME_MS,
-	* 	if the delta of time is nagative or greater than CONST_MAX_JITTER_MS.
-	* 2. last_pkt_time: specifies the original packet time,
-	* 	is used to detect next jitter.
-	* 3. last_pkt_correct_time: simply add the positive delta, 
-	* 	and enforce the time monotonically.
-	*/
-	u_int32_t time = msg->header.timestamp;
-	int32_t delta = time - last_pkt_time;
-
-	// if jitter detected, reset the delta.
-	if (delta < 0 || delta > CONST_MAX_JITTER_MS) {
-		// calc the right diff by audio sample rate
-		if (msg->header.is_audio() && audio_sample_rate > 0) {
-			delta = (int32_t)(delta * 1000.0 / audio_sample_rate);
-		} else if (msg->header.is_video() && video_frame_rate > 0) {
-			delta = (int32_t)(delta * 1.0 / video_frame_rate);
-		} else {
-			delta = DEFAULT_FRAME_TIME_MS;
-		}
-
-		// sometimes, the time is absolute time, so correct it again.
-		if (delta < 0 || delta > CONST_MAX_JITTER_MS) {
-			delta = DEFAULT_FRAME_TIME_MS;
-		}
-		
-		srs_info("jitter detected, last_pts=%d, pts=%d, diff=%d, last_time=%d, time=%d, diff=%d",
-			last_pkt_time, time, time - last_pkt_time, last_pkt_correct_time, last_pkt_correct_time + delta, delta);
-	} else {
-		srs_verbose("timestamp no jitter. time=%d, last_pkt=%d, correct_to=%d", 
-			time, last_pkt_time, last_pkt_correct_time + delta);
-	}
-	
-	last_pkt_correct_time = srs_max(0, last_pkt_correct_time + delta);
-	msg->header.timestamp = last_pkt_correct_time;
-	last_pkt_time = time;
-	
-	return ret;
-}
-
 void SrsConsumer::clear()
 {
 	std::vector<SrsSharedPtrMessage*>::iterator it;
@@ -235,6 +238,18 @@ void SrsConsumer::clear()
 	msgs.clear();
 }
 
+std::map<std::string, SrsSource*> SrsSource::pool;
+
+SrsSource* SrsSource::find(std::string stream_url)
+{
+	if (pool.find(stream_url) == pool.end()) {
+		pool[stream_url] = new SrsSource(stream_url);
+		srs_verbose("create new source for url=%s", stream_url.c_str());
+	}
+	
+	return pool[stream_url];
+}
+
 SrsSource::SrsSource(std::string _stream_url)
 {
 	stream_url = _stream_url;

@@ -41,13 +41,35 @@ class SrsSharedPtrMessage;
 class SrsHLS;
 
 /**
-* the consumer for SrsSource, that is a play client.
+* time jitter detect and correct,
+* to ensure the rtmp stream is monotonically.
 */
-class SrsConsumer
+class SrsRtmpJitter
 {
 private:
 	u_int32_t last_pkt_time;
 	u_int32_t last_pkt_correct_time;
+public:
+	SrsRtmpJitter();
+	virtual ~SrsRtmpJitter();
+public:
+	/**
+	* detect the time jitter and correct it.
+	*/
+	virtual int correct(SrsSharedPtrMessage* msg, int audio_sample_rate, int video_frame_rate);
+	/**
+	* get current client time, the last packet time.
+	*/
+	virtual int get_time();
+};
+
+/**
+* the consumer for SrsSource, that is a play client.
+*/
+class SrsConsumer
+{
+private:
+	SrsRtmpJitter* jitter;
 	SrsSource* source;
 	std::vector<SrsSharedPtrMessage*> msgs;
 	bool paused;
@@ -82,10 +104,6 @@ private:
 	* remove to cache only one gop.
 	*/
 	virtual void shrink();
-	/**
-	* detect the time jitter and correct it.
-	*/
-	virtual int jitter_correct(SrsSharedPtrMessage* msg, int audio_sample_rate, int video_frame_rate);
 	virtual void clear();
 };