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merge from 2.0

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pull/499/head
winlin 10 years ago
parent 12eab8a128 56938efdbd
commit b0ce38dc8f
8 changed files with 70 additions and 105 deletions
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1
README.md
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@ -348,6 +348,7 @@ 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.

6
trunk/src/app/srs_app_dvr.cpp
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@ -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;
}

9
trunk/src/app/srs_app_forward.cpp
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@ -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;
}

6
trunk/src/app/srs_app_hls.cpp
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@ -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;
}

20
trunk/src/app/srs_app_http_conn.cpp
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@ -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;
}

13
trunk/src/app/srs_app_http_conn.hpp
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@ -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);
};
/**

90
trunk/src/app/srs_app_source.cpp
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@ -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 {
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;
}
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;
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();

28
trunk/src/app/srs_app_source.hpp
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@ -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.
@ -491,14 +482,6 @@ private:
// the event handler.
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
*/
@ -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

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