fix #425, refine the time jitter, correct (-inf,-250)+(250,+inf) to 10ms. 2.0.175

@@ -344,6 +344,7 @@ Remark:
 
 ### SRS 2.0 history
 
+* v2.0, 2015-06-10, fix [#425](https://github.com/simple-rtmp-server/srs/issues/425) refine the time jitter, correct (-inf,-250)+(250,+inf) to 10ms. 2.0.175
 * v2.0, 2015-06-10, fix [#424](https://github.com/simple-rtmp-server/srs/issues/424) fix aggregate timestamp bug. 2.0.174
 * v2.0, 2015-06-06, fix [#421](https://github.com/simple-rtmp-server/srs/issues/421) drop video for unkown RTMP header.
 * v2.0, 2015-06-05, fix [#420](https://github.com/simple-rtmp-server/srs/issues/420) remove ts for hls ram mode.

@@ -273,7 +273,7 @@ int SrsFlvSegment::write_audio(SrsSharedPtrMessage* shared_audio)
     SrsSharedPtrMessage* audio = shared_audio->copy();
     SrsAutoFree(SrsSharedPtrMessage, audio);
     
-    if ((jitter->correct(audio, 0, 0, jitter_algorithm)) != ERROR_SUCCESS) {
+    if ((jitter->correct(audio, jitter_algorithm)) != ERROR_SUCCESS) {
         return ret;
     }
     
@@ -324,7 +324,7 @@ int SrsFlvSegment::write_video(SrsSharedPtrMessage* shared_video)
         }
     }
     
-    if ((jitter->correct(video, 0, 0, jitter_algorithm)) != ERROR_SUCCESS) {
+    if ((jitter->correct(video, jitter_algorithm)) != ERROR_SUCCESS) {
         return ret;
     }
     
@@ -334,7 +334,7 @@ int SrsFlvSegment::write_video(SrsSharedPtrMessage* shared_video)
         return ret;
     }
     
-    int32_t timestamp = plan->filter_timestamp(video->timestamp);
+    int32_t timestamp = (int32_t)plan->filter_timestamp(video->timestamp);
     if ((ret = enc->write_video(timestamp, payload, size)) != ERROR_SUCCESS) {
         return ret;
     }

@@ -163,7 +163,8 @@ int SrsForwarder::on_meta_data(SrsSharedPtrMessage* shared_metadata)
 
     SrsSharedPtrMessage* metadata = shared_metadata->copy();
     
-    if ((ret = jitter->correct(metadata, 0, 0, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
+    // TODO: FIXME: config the jitter of Forwarder.
+    if ((ret = jitter->correct(metadata, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
         srs_freep(metadata);
         return ret;
     }
@@ -181,7 +182,8 @@ int SrsForwarder::on_audio(SrsSharedPtrMessage* shared_audio)
     
     SrsSharedPtrMessage* msg = shared_audio->copy();
     
-    if ((ret = jitter->correct(msg, 0, 0, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
+    // TODO: FIXME: config the jitter of Forwarder.
+    if ((ret = jitter->correct(msg, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
         srs_freep(msg);
         return ret;
     }
@@ -204,7 +206,8 @@ int SrsForwarder::on_video(SrsSharedPtrMessage* shared_video)
 
     SrsSharedPtrMessage* msg = shared_video->copy();
     
-    if ((ret = jitter->correct(msg, 0, 0, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
+    // TODO: FIXME: config the jitter of Forwarder.
+    if ((ret = jitter->correct(msg, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
         srs_freep(msg);
         return ret;
     }

@@ -1337,7 +1337,8 @@ int SrsHls::on_audio(SrsSharedPtrMessage* shared_audio)
         return hls_cache->on_sequence_header(muxer);
     }
     
-    if ((ret = jitter->correct(audio, 0, 0, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
+    // TODO: FIXME: config the jitter of HLS.
+    if ((ret = jitter->correct(audio, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
         srs_error("rtmp jitter correct audio failed. ret=%d", ret);
         return ret;
     }
@@ -1394,7 +1395,8 @@ int SrsHls::on_video(SrsSharedPtrMessage* shared_video)
         return hls_cache->on_sequence_header(muxer);
     }
     
-    if ((ret = jitter->correct(video, 0, 0, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
+    // TODO: FIXME: config the jitter of HLS.
+    if ((ret = jitter->correct(video, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
         srs_error("rtmp jitter correct video failed. ret=%d", ret);
         return ret;
     }

@@ -1265,7 +1265,7 @@ int SrsStreamCache::start()
     return pthread->start();
 }
 
-int SrsStreamCache::dump_cache(SrsConsumer* consumer)
+int SrsStreamCache::dump_cache(SrsConsumer* consumer, SrsRtmpJitterAlgorithm jitter)
 {
     int ret = ERROR_SUCCESS;
 
@@ -1276,8 +1276,8 @@ int SrsStreamCache::dump_cache(SrsConsumer* consumer)
         return ret;
     }
     
-    // TODO: FIXME: config it.
-    if ((ret = queue->dump_packets(consumer, false, 0, 0, SrsRtmpJitterAlgorithmOFF)) != ERROR_SUCCESS) {
+    // the jitter is get from SrsSource, which means the time_jitter of vhost.
+    if ((ret = queue->dump_packets(consumer, false, jitter)) != ERROR_SUCCESS) {
         return ret;
     }
     
@@ -1398,7 +1398,7 @@ bool SrsTsStreamEncoder::has_cache()
     return false;
 }
 
-int SrsTsStreamEncoder::dump_cache(SrsConsumer* /*consumer*/)
+int SrsTsStreamEncoder::dump_cache(SrsConsumer* /*consumer*/, SrsRtmpJitterAlgorithm /*jitter*/)
 {
     // for ts stream, ignore cache.
     return ERROR_SUCCESS;
@@ -1451,7 +1451,7 @@ bool SrsFlvStreamEncoder::has_cache()
     return false;
 }
 
-int SrsFlvStreamEncoder::dump_cache(SrsConsumer* /*consumer*/)
+int SrsFlvStreamEncoder::dump_cache(SrsConsumer* /*consumer*/, SrsRtmpJitterAlgorithm /*jitter*/)
 {
     // for flv stream, ignore cache.
     return ERROR_SUCCESS;
@@ -1518,10 +1518,10 @@ bool SrsAacStreamEncoder::has_cache()
     return true;
 }
 
-int SrsAacStreamEncoder::dump_cache(SrsConsumer* consumer)
+int SrsAacStreamEncoder::dump_cache(SrsConsumer* consumer, SrsRtmpJitterAlgorithm jitter)
 {
     srs_assert(cache);
-    return cache->dump_cache(consumer);
+    return cache->dump_cache(consumer, jitter);
 }
 
 SrsMp3StreamEncoder::SrsMp3StreamEncoder()
@@ -1574,10 +1574,10 @@ bool SrsMp3StreamEncoder::has_cache()
     return true;
 }
 
-int SrsMp3StreamEncoder::dump_cache(SrsConsumer* consumer)
+int SrsMp3StreamEncoder::dump_cache(SrsConsumer* consumer, SrsRtmpJitterAlgorithm jitter)
 {
     srs_assert(cache);
-    return cache->dump_cache(consumer);
+    return cache->dump_cache(consumer, jitter);
 }
 
 SrsStreamWriter::SrsStreamWriter(ISrsHttpResponseWriter* w)
@@ -1686,7 +1686,7 @@ int SrsLiveStream::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r)
     
     // if gop cache enabled for encoder, dump to consumer.
     if (enc->has_cache()) {
-        if ((ret = enc->dump_cache(consumer)) != ERROR_SUCCESS) {
+        if ((ret = enc->dump_cache(consumer, source->jitter())) != ERROR_SUCCESS) {
             srs_error("http: dump cache to consumer failed. ret=%d", ret);
             return ret;
         }

@@ -46,6 +46,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #include <srs_kernel_file.hpp>
 #include <srs_app_thread.hpp>
 #include <srs_app_conn.hpp>
+#include <srs_app_source.hpp>
 
 class SrsServer;
 class SrsSource;
@@ -407,7 +408,7 @@ public:
     virtual ~SrsStreamCache();
 public:
     virtual int start();
-    virtual int dump_cache(SrsConsumer* consumer);
+    virtual int dump_cache(SrsConsumer* consumer, SrsRtmpJitterAlgorithm jitter);
 // interface ISrsEndlessThreadHandler.
 public:
     virtual int cycle();
@@ -444,7 +445,7 @@ public:
     /**
     * dumps the cache of encoder to consumer.
     */
-    virtual int dump_cache(SrsConsumer* consumer) = 0;
+    virtual int dump_cache(SrsConsumer* consumer, SrsRtmpJitterAlgorithm jitter) = 0;
 };
 
 /**
@@ -464,7 +465,7 @@ public:
     virtual int write_metadata(int64_t timestamp, char* data, int size);
 public:
     virtual bool has_cache();
-    virtual int dump_cache(SrsConsumer* consumer);
+    virtual int dump_cache(SrsConsumer* consumer, SrsRtmpJitterAlgorithm jitter);
 };
 
 #ifdef SRS_PERF_FAST_FLV_ENCODER
@@ -502,7 +503,7 @@ public:
     virtual int write_metadata(int64_t timestamp, char* data, int size);
 public:
     virtual bool has_cache();
-    virtual int dump_cache(SrsConsumer* consumer);
+    virtual int dump_cache(SrsConsumer* consumer, SrsRtmpJitterAlgorithm jitter);
 };
 
 /**
@@ -523,7 +524,7 @@ public:
     virtual int write_metadata(int64_t timestamp, char* data, int size);
 public:
     virtual bool has_cache();
-    virtual int dump_cache(SrsConsumer* consumer);
+    virtual int dump_cache(SrsConsumer* consumer, SrsRtmpJitterAlgorithm jitter);
 };
 
 /**
@@ -544,7 +545,7 @@ public:
     virtual int write_metadata(int64_t timestamp, char* data, int size);
 public:
     virtual bool has_cache();
-    virtual int dump_cache(SrsConsumer* consumer);
+    virtual int dump_cache(SrsConsumer* consumer, SrsRtmpJitterAlgorithm jitter);
 };
 
 /**

@@ -47,8 +47,9 @@ using namespace std;
 #include <srs_core_autofree.hpp>
 #include <srs_rtmp_utility.hpp>
 
-#define CONST_MAX_JITTER_MS         500
-#define DEFAULT_FRAME_TIME_MS         40
+#define CONST_MAX_JITTER_MS         250
+#define CONST_MAX_JITTER_MS_NEG         -250
+#define DEFAULT_FRAME_TIME_MS         10
 
 // for 26ms per audio packet,
 // 115 packets is 3s.
@@ -70,14 +71,15 @@ int _srs_time_jitter_string2int(std::string time_jitter)
 
 SrsRtmpJitter::SrsRtmpJitter()
 {
-    last_pkt_correct_time = last_pkt_time = 0;
+    last_pkt_correct_time = -1;
+    last_pkt_time = 0;
 }
 
 SrsRtmpJitter::~SrsRtmpJitter()
 {
 }
 
-int SrsRtmpJitter::correct(SrsSharedPtrMessage* msg, int tba, int tbv, SrsRtmpJitterAlgorithm ag)
+int SrsRtmpJitter::correct(SrsSharedPtrMessage* msg, SrsRtmpJitterAlgorithm ag)
 {
     int ret = ERROR_SUCCESS;
     
@@ -90,9 +92,8 @@ int SrsRtmpJitter::correct(SrsSharedPtrMessage* msg, int tba, int tbv, SrsRtmpJi
     
         // start at zero, but donot ensure monotonically increasing.
         if (ag == SrsRtmpJitterAlgorithmZERO) {
-            // for the first time, last_pkt_correct_time is zero.
-            // while when timestamp overflow, the timestamp become smaller, reset the last_pkt_correct_time.
-            if (last_pkt_correct_time <= 0 || msg->timestamp < last_pkt_correct_time) {
+            // for the first time, last_pkt_correct_time is -1.
+            if (last_pkt_correct_time == -1) {
                 last_pkt_correct_time = msg->timestamp;
             }
             msg->timestamp -= last_pkt_correct_time;
@@ -104,16 +105,12 @@ int SrsRtmpJitter::correct(SrsSharedPtrMessage* msg, int tba, int tbv, SrsRtmpJi
     }
     
     // full jitter algorithm, do jitter correct.
-    
     // set to 0 for metadata.
     if (!msg->is_av()) {
         msg->timestamp = 0;
         return ret;
     }
     
-    int sample_rate = tba;
-    int frame_rate = tbv;
-    
     /**
     * we use a very simple time jitter detect/correct algorithm:
     * 1. delta: ensure the delta is positive and valid,
@@ -128,20 +125,10 @@ int SrsRtmpJitter::correct(SrsSharedPtrMessage* msg, int tba, int tbv, SrsRtmpJi
     int64_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->is_audio() && sample_rate > 0) {
-            delta = (int64_t)(delta * 1000.0 / sample_rate);
-        } else if (msg->is_video() && frame_rate > 0) {
-            delta = (int64_t)(delta * 1.0 / frame_rate);
-        } else {
+    if (delta < CONST_MAX_JITTER_MS_NEG || delta > CONST_MAX_JITTER_MS) {
+        // use default 10ms to notice the problem of stream.
+        // @see https://github.com/simple-rtmp-server/srs/issues/425
         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=%"PRId64", pts=%"PRId64", diff=%"PRId64", last_time=%"PRId64", time=%"PRId64", diff=%"PRId64"",
             last_pkt_time, time, time - last_pkt_time, last_pkt_correct_time, last_pkt_correct_time + delta, delta);
@@ -338,7 +325,7 @@ int SrsMessageQueue::dump_packets(int max_count, SrsSharedPtrMessage** pmsgs, in
     return ret;
 }
 
-int SrsMessageQueue::dump_packets(SrsConsumer* consumer, bool atc, int tba, int tbv, SrsRtmpJitterAlgorithm ag)
+int SrsMessageQueue::dump_packets(SrsConsumer* consumer, bool atc, SrsRtmpJitterAlgorithm ag)
 {
     int ret = ERROR_SUCCESS;
     
@@ -350,7 +337,7 @@ int SrsMessageQueue::dump_packets(SrsConsumer* consumer, bool atc, int tba, int
     SrsSharedPtrMessage** omsgs = msgs.data();
     for (int i = 0; i < nb_msgs; i++) {
         SrsSharedPtrMessage* msg = omsgs[i];
-        if ((ret = consumer->enqueue(msg, atc, tba, tbv, ag)) != ERROR_SUCCESS) {
+        if ((ret = consumer->enqueue(msg, atc, ag)) != ERROR_SUCCESS) {
             return ret;
         }
     }
@@ -473,14 +460,14 @@ int SrsConsumer::get_time()
     return jitter->get_time();
 }
 
-int SrsConsumer::enqueue(SrsSharedPtrMessage* shared_msg, bool atc, int tba, int tbv, SrsRtmpJitterAlgorithm ag)
+int SrsConsumer::enqueue(SrsSharedPtrMessage* shared_msg, bool atc, SrsRtmpJitterAlgorithm ag)
 {
     int ret = ERROR_SUCCESS;
     
     SrsSharedPtrMessage* msg = shared_msg->copy();
 
     if (!atc) {
-        if ((ret = jitter->correct(msg, tba, tbv, ag)) != ERROR_SUCCESS) {
+        if ((ret = jitter->correct(msg, ag)) != ERROR_SUCCESS) {
             srs_freep(msg);
             return ret;
         }
@@ -683,14 +670,14 @@ void SrsGopCache::clear()
     audio_after_last_video_count = 0;
 }
     
-int SrsGopCache::dump(SrsConsumer* consumer, bool atc, int tba, int tbv, SrsRtmpJitterAlgorithm jitter_algorithm)
+int SrsGopCache::dump(SrsConsumer* consumer, bool atc, SrsRtmpJitterAlgorithm jitter_algorithm)
 {
     int ret = ERROR_SUCCESS;
     
     std::vector<SrsSharedPtrMessage*>::iterator it;
     for (it = gop_cache.begin(); it != gop_cache.end(); ++it) {
         SrsSharedPtrMessage* msg = *it;
-        if ((ret = consumer->enqueue(msg, atc, tba, tbv, jitter_algorithm)) != ERROR_SUCCESS) {
+        if ((ret = consumer->enqueue(msg, atc, jitter_algorithm)) != ERROR_SUCCESS) {
             srs_error("dispatch cached gop failed. ret=%d", ret);
             return ret;
         }
@@ -906,7 +893,6 @@ SrsSource::SrsSource()
     
     cache_metadata = cache_sh_video = cache_sh_audio = NULL;
     
-    frame_rate = sample_rate = 0;
     _can_publish = true;
     _source_id = -1;
     
@@ -1413,17 +1399,6 @@ int SrsSource::on_meta_data(SrsCommonMessage* msg, SrsOnMetaDataPacket* metadata
     // add version to metadata, please donot remove it, for debug.
     metadata->metadata->set("server_version", SrsAmf0Any::str(RTMP_SIG_SRS_VERSION));
     
-    if ((prop = metadata->metadata->get_property("audiosamplerate")) != NULL) {
-        if (prop->is_number()) {
-            sample_rate = (int)prop->to_number();
-        }
-    }
-    if ((prop = metadata->metadata->get_property("framerate")) != NULL) {
-        if (prop->is_number()) {
-            frame_rate = (int)prop->to_number();
-        }
-    }
-    
     // if allow atc_auto and bravo-atc detected, open atc for vhost.
     atc = _srs_config->get_atc(_req->vhost);
     if (_srs_config->get_atc_auto(_req->vhost)) {
@@ -1466,7 +1441,7 @@ int SrsSource::on_meta_data(SrsCommonMessage* msg, SrsOnMetaDataPacket* metadata
         std::vector<SrsConsumer*>::iterator it;
         for (it = consumers.begin(); it != consumers.end(); ++it) {
             SrsConsumer* consumer = *it;
-            if ((ret = consumer->enqueue(cache_metadata, atc, sample_rate, frame_rate, jitter_algorithm)) != ERROR_SUCCESS) {
+            if ((ret = consumer->enqueue(cache_metadata, atc, jitter_algorithm)) != ERROR_SUCCESS) {
                 srs_error("dispatch the metadata failed. ret=%d", ret);
                 return ret;
             }
@@ -1600,7 +1575,7 @@ int SrsSource::on_audio_imp(SrsSharedPtrMessage* msg)
         SrsConsumer** pconsumer = consumers.data();
         for (int i = 0; i < nb_consumers; i++) {
             SrsConsumer* consumer = pconsumer[i];
-            if ((ret = consumer->enqueue(msg, atc, sample_rate, frame_rate, jitter_algorithm)) != ERROR_SUCCESS) {
+            if ((ret = consumer->enqueue(msg, atc, jitter_algorithm)) != ERROR_SUCCESS) {
                 srs_error("dispatch the audio failed. ret=%d", ret);
                 return ret;
             }
@@ -1802,7 +1777,7 @@ int SrsSource::on_video_imp(SrsSharedPtrMessage* msg)
     if (true) {
         for (int i = 0; i < (int)consumers.size(); i++) {
             SrsConsumer* consumer = consumers.at(i);
-            if ((ret = consumer->enqueue(msg, atc, sample_rate, frame_rate, jitter_algorithm)) != ERROR_SUCCESS) {
+            if ((ret = consumer->enqueue(msg, atc, jitter_algorithm)) != ERROR_SUCCESS) {
                 srs_error("dispatch the video failed. ret=%d", ret);
                 return ret;
             }
@@ -2091,8 +2066,6 @@ void SrsSource::on_unpublish()
     gop_cache->clear();
 
     srs_freep(cache_metadata);
-    frame_rate = sample_rate = 0;
-    
     srs_freep(cache_sh_video);
     srs_freep(cache_sh_audio);
     
@@ -2132,12 +2105,8 @@ int SrsSource::create_consumer(SrsConsumer*& consumer, bool ds, bool dm, bool dg
         }
     }
     
-    int tba = sample_rate;
-    int tbv = frame_rate;
-    SrsRtmpJitterAlgorithm ag = jitter_algorithm;
-    
     // copy metadata.
-    if (dm && cache_metadata && (ret = consumer->enqueue(cache_metadata, atc, tba, tbv, ag)) != ERROR_SUCCESS) {
+    if (dm && cache_metadata && (ret = consumer->enqueue(cache_metadata, atc, jitter_algorithm)) != ERROR_SUCCESS) {
         srs_error("dispatch metadata failed. ret=%d", ret);
         return ret;
     }
@@ -2146,28 +2115,28 @@ int SrsSource::create_consumer(SrsConsumer*& consumer, bool ds, bool dm, bool dg
     // copy sequence header
     // copy audio sequence first, for hls to fast parse the "right" audio codec.
     // @see https://github.com/simple-rtmp-server/srs/issues/301
-    if (ds && cache_sh_audio && (ret = consumer->enqueue(cache_sh_audio, atc, tba, tbv, ag)) != ERROR_SUCCESS) {
+    if (ds && cache_sh_audio && (ret = consumer->enqueue(cache_sh_audio, atc, jitter_algorithm)) != ERROR_SUCCESS) {
         srs_error("dispatch audio sequence header failed. ret=%d", ret);
         return ret;
     }
     srs_info("dispatch audio sequence header success");
 
-    if (ds && cache_sh_video && (ret = consumer->enqueue(cache_sh_video, atc, tba, tbv, ag)) != ERROR_SUCCESS) {
+    if (ds && cache_sh_video && (ret = consumer->enqueue(cache_sh_video, atc, jitter_algorithm)) != ERROR_SUCCESS) {
         srs_error("dispatch video sequence header failed. ret=%d", ret);
         return ret;
     }
     srs_info("dispatch video sequence header success");
     
     // copy gop cache to client.
-    if (dg && (ret = gop_cache->dump(consumer, atc, tba, tbv, ag)) != ERROR_SUCCESS) {
+    if (dg && (ret = gop_cache->dump(consumer, atc, jitter_algorithm)) != ERROR_SUCCESS) {
         return ret;
     }
     
     // print status.
     if (dg) {
-        srs_trace("create consumer, queue_size=%.2f, tba=%d, tbv=%d", queue_size, sample_rate, frame_rate);
+        srs_trace("create consumer, queue_size=%.2f, jitter=%d", queue_size, jitter_algorithm);
     } else {
-        srs_trace("create consumer, ignore gop cache, tba=%d, tbv=%d", sample_rate, frame_rate);
+        srs_trace("create consumer, ignore gop cache, jitter=%d", jitter_algorithm);
     }
     
     return ret;
@@ -2192,6 +2161,11 @@ void SrsSource::set_cache(bool enabled)
     gop_cache->set(enabled);
 }
 
+SrsRtmpJitterAlgorithm SrsSource::jitter()
+{
+    return jitter_algorithm;
+}
+
 int SrsSource::on_edge_start_play()
 {
     return play_edge->on_client_play();

@@ -95,12 +95,9 @@ public:
 public:
     /**
     * detect the time jitter and correct it.
-    * @param tba, the audio timebase, used to calc the "right" delta if jitter detected.
-    * @param tbv, the video timebase, used to calc the "right" delta if jitter detected.
-    * @param start_at_zero whether ensure stream start at zero.
-    * @param mono_increasing whether ensure stream is monotonically inscreasing.
+    * @param ag the algorithm to use for time jitter.
     */
-    virtual int correct(SrsSharedPtrMessage* msg, int tba, int tbv, SrsRtmpJitterAlgorithm ag);
+    virtual int correct(SrsSharedPtrMessage* msg, SrsRtmpJitterAlgorithm ag);
     /**
     * get current client time, the last packet time.
     */
@@ -186,7 +183,7 @@ public:
     * dumps packets to consumer, use specified args.
     * @remark the atc/tba/tbv/ag are same to SrsConsumer.enqueue().
     */
-    virtual int dump_packets(SrsConsumer* consumer, bool atc, int tba, int tbv, SrsRtmpJitterAlgorithm ag);
+    virtual int dump_packets(SrsConsumer* consumer, bool atc, SrsRtmpJitterAlgorithm ag);
 private:
     /**
     * remove a gop from the front.
@@ -255,13 +252,9 @@ public:
     * enqueue an shared ptr message.
     * @param shared_msg, directly ptr, copy it if need to save it.
     * @param whether atc, donot use jitter correct if true.
-    * @param tba timebase of audio.
-    *         used to calc the audio time delta if time-jitter detected.
-    * @param tbv timebase of video.
-    *        used to calc the video time delta if time-jitter detected.
     * @param ag the algorithm of time jitter.
     */
-    virtual int enqueue(SrsSharedPtrMessage* shared_msg, bool atc, int tba, int tbv, SrsRtmpJitterAlgorithm ag);
+    virtual int enqueue(SrsSharedPtrMessage* shared_msg, bool atc, SrsRtmpJitterAlgorithm ag);
     /**
     * get packets in consumer queue.
     * @param msgs the msgs array to dump packets to send.
@@ -351,9 +344,7 @@ public:
     /**
     * dump the cached gop to consumer.
     */
-    virtual int dump(SrsConsumer* consumer, 
-        bool atc, int tba, int tbv, SrsRtmpJitterAlgorithm jitter_algorithm
-    );
+    virtual int dump(SrsConsumer* consumer, bool atc, SrsRtmpJitterAlgorithm jitter_algorithm);
     /**
     * used for atc to get the time of gop cache,
     * the atc will adjust the sequence header timestamp to gop cache.
@@ -492,14 +483,6 @@ private:
     ISrsSourceHandler* handler;
 private:
     /**
-    * the sample rate of audio in metadata.
-    */
-    int sample_rate;
-    /**
-    * the video frame rate in metadata.
-    */
-    int frame_rate;
-    /**
     * can publish, true when is not streaming
     */
     bool _can_publish;
@@ -595,6 +578,7 @@ public:
     );
     virtual void on_consumer_destroy(SrsConsumer* consumer);
     virtual void set_cache(bool enabled);
+    virtual SrsRtmpJitterAlgorithm jitter();
 // internal
 public:
     // for edge, when play edge stream, check the state

@@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 // current release version
 #define VERSION_MAJOR       2
 #define VERSION_MINOR       0
-#define VERSION_REVISION    174
+#define VERSION_REVISION    175
 
 // server info.
 #define RTMP_SIG_SRS_KEY "SRS"