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

--------- Co-authored-by: Haibo Chen <495810242@qq.com>
ossrs · Jun 14, 2024 · 9dba99a · 9dba99a · winlinvip · Jul 4, 2024
1 parent 242152b
commit 9dba99a
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Showing 17 changed files with 120 additions and 99 deletions.
diff --git a/.github/workflows/test.yml b/.github/workflows/test.yml
@@ -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

diff --git a/trunk/3rdparty/srs-bench/blackbox/hevc_test.go b/trunk/3rdparty/srs-bench/blackbox/hevc_test.go
@@ -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 {

diff --git a/trunk/doc/CHANGELOG.md b/trunk/doc/CHANGELOG.md
@@ -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)

diff --git a/trunk/src/app/srs_app_rtc_api.cpp b/trunk/src/app/srs_app_rtc_api.cpp
@@ -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

diff --git a/trunk/src/app/srs_app_rtc_conn.cpp b/trunk/src/app/srs_app_rtc_conn.cpp
@@ -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;
@@ -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");
     }
 
@@ -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());
     }
 
@@ -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;
@@ -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;
@@ -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;
@@ -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);
@@ -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);
@@ -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
 
@@ -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");
     }
 
@@ -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 {
@@ -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");
     }
 
@@ -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");
     }
 

diff --git a/trunk/src/app/srs_app_rtc_conn.hpp b/trunk/src/app/srs_app_rtc_conn.hpp
@@ -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>
@@ -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_;
@@ -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:

diff --git a/trunk/src/app/srs_app_rtc_server.cpp b/trunk/src/app/srs_app_rtc_server.cpp
@@ -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");
     }