commit fe328ca88a3d8732447bc00dcd8d8eceef3f3d8c
author Tomas Gunnarsson <tommi@webrtc.org> Wed Feb 16 19:02:12 2022
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Thu Feb 17 09:00:54 2022
tree 49873d902ea9fcf4eaf238d9916d5347a2f4cc0c
parent 27d5f14cf27b99ff02d0f04d6bc69dd0faeb9dd7

Add several thread checks to RtpSender classes.

Minor related updates to AudioTrack and VideoTrack's sequence checkers.

There's more that can be done (or arguably needs to), but this is
a start.

Bug: none
Change-Id: I3ccf8eb9bbb6bef62b83248a23a68871b9fcd9e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251843
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36021}

pc/rtp_sender.cc

diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
index 957f514..110b5aa 100644
--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
@@ -111,7 +111,8 @@
 RtpSenderBase::RtpSenderBase(rtc::Thread* worker_thread,
                              const std::string& id,
                              SetStreamsObserver* set_streams_observer)
-    : worker_thread_(worker_thread),
+    : signaling_thread_(rtc::Thread::Current()),
+      worker_thread_(worker_thread),
       id_(id),
       set_streams_observer_(set_streams_observer) {
   RTC_DCHECK(worker_thread);
@@ -120,6 +121,7 @@
 
 void RtpSenderBase::SetFrameEncryptor(
     rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   frame_encryptor_ = std::move(frame_encryptor);
   // Special Case: Set the frame encryptor to any value on any existing channel.
   if (media_channel_ && ssrc_ && !stopped_) {
@@ -136,6 +138,7 @@
 }
 
 RtpParameters RtpSenderBase::GetParametersInternal() const {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   if (stopped_) {
     return RtpParameters();
   }
@@ -150,6 +153,7 @@
 }
 
 RtpParameters RtpSenderBase::GetParameters() const {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   RtpParameters result = GetParametersInternal();
   last_transaction_id_ = rtc::CreateRandomUuid();
   result.transaction_id = last_transaction_id_.value();
@@ -157,6 +161,7 @@
 }
 
 RTCError RtpSenderBase::SetParametersInternal(const RtpParameters& parameters) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   RTC_DCHECK(!stopped_);
 
   if (UnimplementedRtpParameterHasValue(parameters)) {
@@ -186,6 +191,7 @@
 }
 
 RTCError RtpSenderBase::SetParameters(const RtpParameters& parameters) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   TRACE_EVENT0("webrtc", "RtpSenderBase::SetParameters");
   if (is_transceiver_stopped_) {
     LOG_AND_RETURN_ERROR(
@@ -225,6 +231,7 @@
 }
 
 bool RtpSenderBase::SetTrack(MediaStreamTrackInterface* track) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   TRACE_EVENT0("webrtc", "RtpSenderBase::SetTrack");
   if (stopped_) {
     RTC_LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
@@ -266,6 +273,7 @@
 }
 
 void RtpSenderBase::SetSsrc(uint32_t ssrc) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   TRACE_EVENT0("webrtc", "RtpSenderBase::SetSsrc");
   if (stopped_ || ssrc == ssrc_) {
     return;
@@ -315,6 +323,7 @@
 }
 
 void RtpSenderBase::Stop() {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   TRACE_EVENT0("webrtc", "RtpSenderBase::Stop");
   // TODO(deadbeef): Need to do more here to fully stop sending packets.
   if (stopped_) {
@@ -335,6 +344,7 @@
 
 RTCError RtpSenderBase::DisableEncodingLayers(
     const std::vector<std::string>& rids) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   if (stopped_) {
     LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
                          "Cannot disable encodings on a stopped sender.");
@@ -381,6 +391,7 @@
 
 void RtpSenderBase::SetEncoderToPacketizerFrameTransformer(
     rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   frame_transformer_ = std::move(frame_transformer);
   if (media_channel_ && ssrc_ && !stopped_) {
     worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
@@ -483,7 +494,7 @@
 }
 
 void AudioRtpSender::OnChanged() {
-  // Running on the signaling thread.
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   TRACE_EVENT0("webrtc", "AudioRtpSender::OnChanged");
   RTC_DCHECK(!stopped_);
   if (cached_track_enabled_ != track_->enabled()) {
@@ -518,10 +529,12 @@
 }
 
 rtc::scoped_refptr<DtmfSenderInterface> AudioRtpSender::GetDtmfSender() const {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   return dtmf_sender_proxy_;
 }
 
 void AudioRtpSender::SetSend() {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   RTC_DCHECK(!stopped_);
   RTC_DCHECK(can_send_track());
   if (!media_channel_) {
@@ -552,6 +565,7 @@
 }
 
 void AudioRtpSender::ClearSend() {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   RTC_DCHECK(ssrc_ != 0);
   RTC_DCHECK(!stopped_);
   if (!media_channel_) {
@@ -585,7 +599,7 @@
 }
 
 void VideoRtpSender::OnChanged() {
-  // Running on the signaling thread.
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   TRACE_EVENT0("webrtc", "VideoRtpSender::OnChanged");
   RTC_DCHECK(!stopped_);
 
@@ -604,11 +618,13 @@
 }
 
 rtc::scoped_refptr<DtmfSenderInterface> VideoRtpSender::GetDtmfSender() const {
-  RTC_LOG(LS_ERROR) << "Tried to get DTMF sender from video sender.";
+  RTC_DCHECK_RUN_ON(signaling_thread_);
+  RTC_DLOG(LS_ERROR) << "Tried to get DTMF sender from video sender.";
   return nullptr;
 }
 
 void VideoRtpSender::SetSend() {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   RTC_DCHECK(!stopped_);
   RTC_DCHECK(can_send_track());
   if (!media_channel_) {
@@ -640,6 +656,7 @@
 }
 
 void VideoRtpSender::ClearSend() {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
   RTC_DCHECK(ssrc_ != 0);
   RTC_DCHECK(!stopped_);
   if (!media_channel_) {