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Refine SrsSource, bridge to RTC.

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This commit is contained in:
winlin 2020-05-12 19:53:21 +08:00
parent d2e5cd7bb7
commit 9e417d5416
4 changed files with 155 additions and 17 deletions
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61
trunk/src/app/srs_app_rtc_source.cpp
View file

@ -152,6 +152,7 @@ SrsRtcSource::SrsRtcSource()
_can_publish = true;
rtc_publisher_ = NULL;
bridger_ = new SrsRtcFromRtmpBridger(this);
}
SrsRtcSource::~SrsRtcSource()
@ -165,6 +166,7 @@ SrsRtcSource::~SrsRtcSource()
srs_freep(rtc);
srs_freep(req);
srs_freep(bridger_);
}
srs_error_t SrsRtcSource::initialize(SrsRequest* r)
@ -225,6 +227,11 @@ int SrsRtcSource::pre_source_id()
return _pre_source_id;
}
ISrsSourceBridger* SrsRtcSource::bridger()
{
return bridger_;
}
srs_error_t SrsRtcSource::create_consumer(SrsConnection* conn, SrsRtcConsumer*& consumer)
{
srs_error_t err = srs_success;
@ -382,6 +389,19 @@ srs_error_t SrsRtcSource::on_audio_imp(SrsSharedPtrMessage* msg)
}
}
// TODO: FIXME: Support parsing OPUS for RTC.
if ((err = format->on_audio(msg)) != srs_success) {
return srs_error_wrap(err, "format consume audio");
}
// Parse RTMP message to RTP packets, in FU-A if too large.
if ((err = rtc->on_audio(msg, format)) != srs_success) {
// TODO: We should support more strategies.
srs_warn("rtc: ignore audio error %s", srs_error_desc(err).c_str());
srs_error_reset(err);
rtc->on_unpublish();
}
// copy to all consumer
if (!drop_for_reduce) {
for (int i = 0; i < (int)consumers.size(); i++) {
@ -401,11 +421,6 @@ srs_error_t SrsRtcSource::on_audio_imp(SrsSharedPtrMessage* msg)
}
}
// when sequence header, donot push to gop cache and adjust the timestamp.
if (is_sequence_header) {
return err;
}
// if atc, update the sequence header to abs time.
if (meta->ash()) {
meta->ash()->timestamp = msg->timestamp;
@ -465,11 +480,6 @@ srs_error_t SrsRtcSource::on_video_imp(SrsSharedPtrMessage* msg)
}
}
// when sequence header, donot push to gop cache and adjust the timestamp.
if (is_sequence_header) {
return err;
}
// if atc, update the sequence header to abs time.
if (meta->vsh()) {
meta->vsh()->timestamp = msg->timestamp;
@ -551,3 +561,34 @@ SrsRtcSource* SrsRtcSourceManager::fetch(SrsRequest* r)
SrsRtcSourceManager* _srs_rtc_sources = new SrsRtcSourceManager();
SrsRtcFromRtmpBridger::SrsRtcFromRtmpBridger(SrsRtcSource* source)
{
source_ = source;
}
SrsRtcFromRtmpBridger::~SrsRtcFromRtmpBridger()
{
}
srs_error_t SrsRtcFromRtmpBridger::on_publish()
{
// TODO: FIXME: Should sync with bridger?
return source_->on_publish();
}
srs_error_t SrsRtcFromRtmpBridger::on_audio(SrsSharedPtrMessage* audio)
{
return source_->on_audio_imp(audio);
}
srs_error_t SrsRtcFromRtmpBridger::on_video(SrsSharedPtrMessage* video)
{
return source_->on_video_imp(video);
}
void SrsRtcFromRtmpBridger::on_unpublish()
{
// TODO: FIXME: Should sync with bridger?
source_->on_unpublish();
}

20
trunk/src/app/srs_app_rtc_source.hpp
View file

@ -40,6 +40,7 @@ class SrsSharedPtrMessage;
class SrsCommonMessage;
class SrsMessageArray;
class SrsRtcSource;
class SrsRtcFromRtmpBridger;
class SrsRtcConsumer : public ISrsConsumerQueue
{
@ -94,6 +95,8 @@ private:
int _pre_source_id;
SrsRequest* req;
SrsRtcPublisher* rtc_publisher_;
// Transmux RTMP to RTC.
SrsRtcFromRtmpBridger* bridger_;
private:
// To delivery stream to clients.
std::vector<SrsRtcConsumer*> consumers;
@ -118,6 +121,8 @@ public:
// Get current source id.
virtual int source_id();
virtual int pre_source_id();
// Get the bridger.
ISrsSourceBridger* bridger();
public:
// Create consumer
// @param consumer, output the create consumer.
@ -144,7 +149,6 @@ public:
// TODO: FIXME: Merge with on_audio.
srs_error_t on_rtc_audio(SrsSharedPtrMessage* audio);
virtual srs_error_t on_video(SrsCommonMessage* video);
private:
virtual srs_error_t on_audio_imp(SrsSharedPtrMessage* audio);
virtual srs_error_t on_video_imp(SrsSharedPtrMessage* video);
};
@ -171,5 +175,19 @@ private:
// Global singleton instance.
extern SrsRtcSourceManager* _srs_rtc_sources;
class SrsRtcFromRtmpBridger : public ISrsSourceBridger
{
private:
SrsRtcSource* source_;
public:
SrsRtcFromRtmpBridger(SrsRtcSource* source);
virtual ~SrsRtcFromRtmpBridger();
public:
virtual srs_error_t on_publish();
virtual srs_error_t on_audio(SrsSharedPtrMessage* audio);
virtual srs_error_t on_video(SrsSharedPtrMessage* video);
virtual void on_unpublish();
};
#endif

72
trunk/src/app/srs_app_source.cpp
View file

@ -50,6 +50,7 @@ using namespace std;
#include <srs_app_ng_exec.hpp>
#include <srs_app_dash.hpp>
#include <srs_protocol_format.hpp>
#include <srs_app_rtc_source.hpp>
#define CONST_MAX_JITTER_MS 250
#define CONST_MAX_JITTER_MS_NEG -250
@ -1715,6 +1716,7 @@ srs_error_t SrsSourceManager::fetch_or_create(SrsRequest* r, ISrsSourceHandler*
// Use lock to protect coroutine switch.
// @bug https://github.com/ossrs/srs/issues/1230
// TODO: FIXME: Use smaller lock.
SrsLocker(lock);
SrsSource* source = NULL;
@ -1729,17 +1731,41 @@ srs_error_t SrsSourceManager::fetch_or_create(SrsRequest* r, ISrsSourceHandler*
// should always not exists for create a source.
srs_assert (pool.find(stream_url) == pool.end());
#ifdef SRS_RTC
bool rtc_server_enabled = _srs_config->get_rtc_server_enabled();
bool rtc_enabled = _srs_config->get_rtc_enabled(r->vhost);
// Get the RTC source and bridger.
SrsRtcSource* rtc = NULL;
if (rtc_server_enabled && rtc_enabled) {
if ((err = _srs_rtc_sources->fetch_or_create(r, &rtc)) != srs_success) {
err = srs_error_wrap(err, "init rtc %s", r->get_stream_url().c_str());
goto failed;
}
}
#endif
srs_trace("new source, stream_url=%s", stream_url.c_str());
source = new SrsSource();
if ((err = source->initialize(r, h)) != srs_success) {
return srs_error_wrap(err, "init source %s", r->get_stream_url().c_str());
err = srs_error_wrap(err, "init source %s", r->get_stream_url().c_str());
goto failed;
}
#ifdef SRS_RTC
// If rtc enabled, bridge RTMP source to RTC,
// all RTMP packets will be forwarded to RTC source.
if (source && rtc) {
source->bridge_to(rtc->bridger());
}
#endif
pool[stream_url] = source;
*pps = source;
return err;
failed:
srs_freep(source);
return err;
}
@ -1832,6 +1858,14 @@ void SrsSourceManager::destroy()
pool.clear();
}
ISrsSourceBridger::ISrsSourceBridger()
{
}
ISrsSourceBridger::~ISrsSourceBridger()
{
}
SrsSource::SrsSource()
{
req = NULL;
@ -1842,6 +1876,9 @@ SrsSource::SrsSource()
_can_publish = true;
_pre_source_id = _source_id = -1;
die_at = 0;
handler = NULL;
bridger = NULL;
play_edge = new SrsPlayEdge();
publish_edge = new SrsPublishEdge();
@ -1948,6 +1985,11 @@ srs_error_t SrsSource::initialize(SrsRequest* r, ISrsSourceHandler* h)
return err;
}
void SrsSource::set_bridger(ISrsSourceBridger* v)
{
bridger = v;
}
srs_error_t SrsSource::on_reload_vhost_play(string vhost)
{
srs_error_t err = srs_success;
@ -2198,6 +2240,11 @@ srs_error_t SrsSource::on_audio_imp(SrsSharedPtrMessage* msg)
return srs_error_wrap(err, "consume audio");
}
// For bridger to consume the message.
if (bridger && (err = bridger->on_audio(msg)) != srs_success) {
return srs_error_wrap(err, "bridger consume audio");
}
// copy to all consumer
if (!drop_for_reduce) {
for (int i = 0; i < (int)consumers.size(); i++) {
@ -2226,7 +2273,7 @@ srs_error_t SrsSource::on_audio_imp(SrsSharedPtrMessage* msg)
if ((err = gop_cache->cache(msg)) != srs_success) {
return srs_error_wrap(err, "gop cache consume audio");
}
// if atc, update the sequence header to abs time.
if (atc) {
if (meta->ash()) {
@ -2323,6 +2370,11 @@ srs_error_t SrsSource::on_video_imp(SrsSharedPtrMessage* msg)
return srs_error_wrap(err, "hub consume video");
}
// For bridger to consume the message.
if (bridger && (err = bridger->on_video(msg)) != srs_success) {
return srs_error_wrap(err, "bridger consume video");
}
// copy to all consumer
if (!drop_for_reduce) {
for (int i = 0; i < (int)consumers.size(); i++) {
@ -2482,6 +2534,11 @@ srs_error_t SrsSource::on_publish()
if ((err = handler->on_publish(this, req)) != srs_success) {
return srs_error_wrap(err, "handle publish");
}
if (bridger && (err = bridger->on_publish()) != srs_success) {
return srs_error_wrap(err, "bridger publish");
}
SrsStatistic* stat = SrsStatistic::instance();
stat->on_stream_publish(req, _source_id);
@ -2517,7 +2574,12 @@ void SrsSource::on_unpublish()
srs_assert(handler);
SrsStatistic* stat = SrsStatistic::instance();
stat->on_stream_close(req);
handler->on_unpublish(this, req);
if (bridger) {
bridger->on_unpublish();
}
// no consumer, stream is die.
if (consumers.empty()) {

19
trunk/src/app/srs_app_source.hpp
View file

@ -488,7 +488,7 @@ public:
private:
virtual srs_error_t do_cycle();
public:
// when system exit, destroy the sources,
// when system exit, destroy th`e sources,
// For gmc to analysis mem leaks.
virtual void destroy();
};
@ -496,6 +496,19 @@ public:
// Global singleton instance.
extern SrsSourceManager* _srs_sources;
// For two sources to bridge with each other.
class ISrsSourceBridger
{
public:
ISrsSourceBridger();
virtual ~ISrsSourceBridger();
public:
virtual srs_error_t on_publish() = 0;
virtual srs_error_t on_audio(SrsSharedPtrMessage* audio) = 0;
virtual srs_error_t on_video(SrsSharedPtrMessage* video) = 0;
virtual void on_unpublish() = 0;
};
// live streaming source.
class SrsSource : public ISrsReloadHandler
{
@ -529,6 +542,8 @@ private:
int64_t last_packet_time;
// The event handler.
ISrsSourceHandler* handler;
// The source bridger for other source.
ISrsSourceBridger* bridger;
// The edge control service
SrsPlayEdge* play_edge;
SrsPublishEdge* publish_edge;
@ -555,6 +570,8 @@ public:
public:
// Initialize the hls with handlers.
virtual srs_error_t initialize(SrsRequest* r, ISrsSourceHandler* h);
// Bridge to other source, forward packets to it.
void bridge_to(ISrsSourceBridger* v);
// Interface ISrsReloadHandler
public:
virtual srs_error_t on_reload_vhost_play(std::string vhost);