Remove 2 Invokes to the network thread when creating a channel.
...and one when destroying a channel object. This CL removes Init_n() and Deinit_n() from the BaseChannel class. Channel classes now use SetRtpTransport to do initialization and uninitialization on the network thread. Notably if an implementation has called SetRtpTransport() with a valid transport pointer, it is required that SetRtpTransport be called again with a nullptr before the channel object can be deleted. In situations where multiple channels are created, this can mean a substantial reduction in thread hops. We still hop to the worker in order to construct the objects - this can probably be avoided and SetChannel() is still a synchronous operation for the transceivers. Furthermore, teardown of channel objects also still happens synchronously and across network/worker/signaling threads. Bug: webrtc:11992 Change-Id: I68ca7596e181fc82996e3e290733d97381aa5e78 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246740 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35738}
This commit is contained in:
parent
4f19950660
commit
4f8a58c3d2
@ -184,28 +184,7 @@ void BaseChannel::DisconnectFromRtpTransport_n() {
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rtp_transport_->SignalWritableState.disconnect(this);
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rtp_transport_->SignalSentPacket.disconnect(this);
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rtp_transport_ = nullptr;
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}
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void BaseChannel::Init_n(webrtc::RtpTransportInternal* rtp_transport) {
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// Set the transport before we call SetInterface() since setting the interface
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// pointer will call us back to set transport options.
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SetRtpTransport(rtp_transport);
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// Both RTP and RTCP channels should be set, we can call SetInterface on
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// the media channel and it can set network options.
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RTC_DCHECK(!media_channel_->HasNetworkInterface());
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media_channel_->SetInterface(this);
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}
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void BaseChannel::Deinit_n() {
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// Packets arrive on the network thread, processing packets calls virtual
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// functions, so need to stop this process in Deinit that is called in
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// derived classes destructor.
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media_channel_->SetInterface(/*iface=*/nullptr);
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if (rtp_transport_) {
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DisconnectFromRtpTransport_n();
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}
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RTC_DCHECK(!network_initialized());
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media_channel_->SetInterface(nullptr);
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}
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bool BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) {
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@ -229,6 +208,10 @@ bool BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) {
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if (!ConnectToRtpTransport_n()) {
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return false;
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}
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RTC_DCHECK(!media_channel_->HasNetworkInterface());
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media_channel_->SetInterface(this);
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media_channel_->OnReadyToSend(rtp_transport_->IsReadyToSend());
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UpdateWritableState_n();
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@ -242,6 +225,7 @@ bool BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) {
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}
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}
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}
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return true;
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}
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@ -111,10 +111,6 @@ class BaseChannel : public ChannelInterface,
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rtc::UniqueRandomIdGenerator* ssrc_generator);
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virtual ~BaseChannel();
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void Init_n(webrtc::RtpTransportInternal* rtp_transport)
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RTC_RUN_ON(network_thread());
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void Deinit_n() RTC_RUN_ON(network_thread());
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rtc::Thread* worker_thread() const { return worker_thread_; }
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rtc::Thread* network_thread() const { return network_thread_; }
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const std::string& content_name() const override {
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@ -143,7 +143,6 @@ ChannelManager::GetSupportedVideoRtpHeaderExtensions() const {
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VoiceChannel* ChannelManager::CreateVoiceChannel(
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webrtc::Call* call,
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const MediaConfig& media_config,
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webrtc::RtpTransportInternal* rtp_transport,
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rtc::Thread* signaling_thread,
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const std::string& content_name,
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bool srtp_required,
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@ -157,9 +156,9 @@ VoiceChannel* ChannelManager::CreateVoiceChannel(
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// thread.
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if (!worker_thread_->IsCurrent()) {
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return worker_thread_->Invoke<VoiceChannel*>(RTC_FROM_HERE, [&] {
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return CreateVoiceChannel(call, media_config, rtp_transport,
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signaling_thread, content_name, srtp_required,
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crypto_options, ssrc_generator, options);
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return CreateVoiceChannel(call, media_config, signaling_thread,
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content_name, srtp_required, crypto_options,
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ssrc_generator, options);
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});
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}
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@ -176,11 +175,6 @@ VoiceChannel* ChannelManager::CreateVoiceChannel(
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absl::WrapUnique(media_channel), content_name, srtp_required,
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crypto_options, ssrc_generator);
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network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
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RTC_DCHECK_RUN_ON(voice_channel->network_thread());
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voice_channel->Init_n(rtp_transport);
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});
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VoiceChannel* voice_channel_ptr = voice_channel.get();
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voice_channels_.push_back(std::move(voice_channel));
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return voice_channel_ptr;
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@ -190,11 +184,6 @@ void ChannelManager::DestroyVoiceChannel(VoiceChannel* voice_channel) {
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TRACE_EVENT0("webrtc", "ChannelManager::DestroyVoiceChannel");
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RTC_DCHECK(voice_channel);
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network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
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RTC_DCHECK_RUN_ON(voice_channel->network_thread());
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voice_channel->Deinit_n();
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});
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if (!worker_thread_->IsCurrent()) {
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worker_thread_->Invoke<void>(RTC_FROM_HERE,
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[&] { DestroyVoiceChannel(voice_channel); });
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@ -212,7 +201,6 @@ void ChannelManager::DestroyVoiceChannel(VoiceChannel* voice_channel) {
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VideoChannel* ChannelManager::CreateVideoChannel(
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webrtc::Call* call,
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const MediaConfig& media_config,
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webrtc::RtpTransportInternal* rtp_transport,
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rtc::Thread* signaling_thread,
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const std::string& content_name,
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bool srtp_required,
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@ -227,9 +215,9 @@ VideoChannel* ChannelManager::CreateVideoChannel(
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// thread.
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if (!worker_thread_->IsCurrent()) {
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return worker_thread_->Invoke<VideoChannel*>(RTC_FROM_HERE, [&] {
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return CreateVideoChannel(call, media_config, rtp_transport,
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signaling_thread, content_name, srtp_required,
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crypto_options, ssrc_generator, options,
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return CreateVideoChannel(call, media_config, signaling_thread,
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content_name, srtp_required, crypto_options,
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ssrc_generator, options,
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video_bitrate_allocator_factory);
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});
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}
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@ -248,11 +236,6 @@ VideoChannel* ChannelManager::CreateVideoChannel(
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absl::WrapUnique(media_channel), content_name, srtp_required,
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crypto_options, ssrc_generator);
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network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
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RTC_DCHECK_RUN_ON(video_channel->network_thread());
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video_channel->Init_n(rtp_transport);
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});
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VideoChannel* video_channel_ptr = video_channel.get();
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video_channels_.push_back(std::move(video_channel));
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return video_channel_ptr;
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@ -262,11 +245,6 @@ void ChannelManager::DestroyVideoChannel(VideoChannel* video_channel) {
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TRACE_EVENT0("webrtc", "ChannelManager::DestroyVideoChannel");
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RTC_DCHECK(video_channel);
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network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
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RTC_DCHECK_RUN_ON(video_channel->network_thread());
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video_channel->Deinit_n();
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});
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if (!worker_thread_->IsCurrent()) {
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worker_thread_->Invoke<void>(RTC_FROM_HERE,
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[&] { DestroyVideoChannel(video_channel); });
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@ -81,7 +81,6 @@ class ChannelManager final {
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// Creates a voice channel, to be associated with the specified session.
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VoiceChannel* CreateVoiceChannel(webrtc::Call* call,
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const MediaConfig& media_config,
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webrtc::RtpTransportInternal* rtp_transport,
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rtc::Thread* signaling_thread,
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const std::string& content_name,
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bool srtp_required,
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@ -97,7 +96,6 @@ class ChannelManager final {
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VideoChannel* CreateVideoChannel(
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webrtc::Call* call,
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const MediaConfig& media_config,
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webrtc::RtpTransportInternal* rtp_transport,
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rtc::Thread* signaling_thread,
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const std::string& content_name,
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bool srtp_required,
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@ -69,16 +69,18 @@ class ChannelManagerTest : public ::testing::Test {
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void TestCreateDestroyChannels(webrtc::RtpTransportInternal* rtp_transport) {
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RTC_DCHECK_RUN_ON(worker_);
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cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
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&fake_call_, cricket::MediaConfig(), rtp_transport,
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rtc::Thread::Current(), cricket::CN_AUDIO, kDefaultSrtpRequired,
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webrtc::CryptoOptions(), &ssrc_generator_, AudioOptions());
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EXPECT_TRUE(voice_channel != nullptr);
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&fake_call_, cricket::MediaConfig(), rtc::Thread::Current(),
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cricket::CN_AUDIO, kDefaultSrtpRequired, webrtc::CryptoOptions(),
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&ssrc_generator_, AudioOptions());
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ASSERT_TRUE(voice_channel != nullptr);
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cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
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&fake_call_, cricket::MediaConfig(), rtp_transport,
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rtc::Thread::Current(), cricket::CN_VIDEO, kDefaultSrtpRequired,
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webrtc::CryptoOptions(), &ssrc_generator_, VideoOptions(),
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&fake_call_, cricket::MediaConfig(), rtc::Thread::Current(),
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cricket::CN_VIDEO, kDefaultSrtpRequired, webrtc::CryptoOptions(),
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&ssrc_generator_, VideoOptions(),
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video_bitrate_allocator_factory_.get());
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EXPECT_TRUE(video_channel != nullptr);
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ASSERT_TRUE(video_channel != nullptr);
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cm_->DestroyVideoChannel(video_channel);
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cm_->DestroyVoiceChannel(voice_channel);
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}
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@ -280,11 +280,11 @@ class ChannelTest : public ::testing::Test, public sigslot::has_slots<> {
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network_thread_->Invoke<void>(RTC_FROM_HERE, [this]() {
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if (channel1_) {
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RTC_DCHECK_RUN_ON(channel1_->network_thread());
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channel1_->Deinit_n();
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channel1_->SetRtpTransport(nullptr);
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}
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if (channel2_) {
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RTC_DCHECK_RUN_ON(channel2_->network_thread());
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channel2_->Deinit_n();
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channel2_->SetRtpTransport(nullptr);
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}
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});
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}
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@ -1450,7 +1450,7 @@ std::unique_ptr<cricket::VoiceChannel> ChannelTest<VoiceTraits>::CreateChannel(
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&ssrc_generator_);
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network_thread->Invoke<void>(RTC_FROM_HERE, [&]() {
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RTC_DCHECK_RUN_ON(channel->network_thread());
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channel->Init_n(rtp_transport);
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channel->SetRtpTransport(rtp_transport);
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});
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return channel;
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}
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@ -1536,7 +1536,7 @@ std::unique_ptr<cricket::VideoChannel> ChannelTest<VideoTraits>::CreateChannel(
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&ssrc_generator_);
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network_thread->Invoke<void>(RTC_FROM_HERE, [&]() {
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RTC_DCHECK_RUN_ON(channel->network_thread());
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channel->Init_n(rtp_transport);
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channel->SetRtpTransport(rtp_transport);
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});
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return channel;
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}
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@ -122,14 +122,18 @@ class RtpSenderReceiverTest
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rtp_transport_ = CreateDtlsSrtpTransport();
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voice_channel_ = channel_manager_->CreateVoiceChannel(
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&fake_call_, cricket::MediaConfig(), rtp_transport_.get(),
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rtc::Thread::Current(), cricket::CN_AUDIO, srtp_required,
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webrtc::CryptoOptions(), &ssrc_generator_, cricket::AudioOptions());
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&fake_call_, cricket::MediaConfig(), rtc::Thread::Current(),
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cricket::CN_AUDIO, srtp_required, webrtc::CryptoOptions(),
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&ssrc_generator_, cricket::AudioOptions());
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video_channel_ = channel_manager_->CreateVideoChannel(
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&fake_call_, cricket::MediaConfig(), rtp_transport_.get(),
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rtc::Thread::Current(), cricket::CN_VIDEO, srtp_required,
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webrtc::CryptoOptions(), &ssrc_generator_, cricket::VideoOptions(),
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&fake_call_, cricket::MediaConfig(), rtc::Thread::Current(),
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cricket::CN_VIDEO, srtp_required, webrtc::CryptoOptions(),
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&ssrc_generator_, cricket::VideoOptions(),
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video_bitrate_allocator_factory_.get());
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voice_channel_->SetRtpTransport(rtp_transport_.get());
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video_channel_->SetRtpTransport(rtp_transport_.get());
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voice_channel_->Enable(true);
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video_channel_->Enable(true);
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voice_media_channel_ = media_engine_->GetVoiceChannel(0);
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@ -170,6 +174,9 @@ class RtpSenderReceiverTest
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video_track_ = nullptr;
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audio_track_ = nullptr;
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voice_channel_->SetRtpTransport(nullptr);
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video_channel_->SetRtpTransport(nullptr);
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channel_manager_->DestroyVoiceChannel(voice_channel_);
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channel_manager_->DestroyVideoChannel(video_channel_);
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@ -156,7 +156,9 @@ RtpTransceiver::~RtpTransceiver() {
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}
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}
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void RtpTransceiver::SetChannel(cricket::ChannelInterface* channel) {
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void RtpTransceiver::SetChannel(
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cricket::ChannelInterface* channel,
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std::function<RtpTransportInternal*(const std::string&)> transport_lookup) {
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RTC_DCHECK_RUN_ON(thread_);
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// Cannot set a non-null channel on a stopped transceiver.
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if (stopped_ && channel) {
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@ -164,6 +166,7 @@ void RtpTransceiver::SetChannel(cricket::ChannelInterface* channel) {
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}
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RTC_DCHECK(channel || channel_);
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RTC_DCHECK(!channel || transport_lookup) << "lookup function not supplied";
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RTC_LOG_THREAD_BLOCK_COUNT();
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@ -189,11 +192,13 @@ void RtpTransceiver::SetChannel(cricket::ChannelInterface* channel) {
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channel_manager_->network_thread()->Invoke<void>(RTC_FROM_HERE, [&]() {
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if (channel_) {
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channel_->SetFirstPacketReceivedCallback(nullptr);
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channel_->SetRtpTransport(nullptr);
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}
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channel_ = channel;
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if (channel_) {
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channel_->SetRtpTransport(transport_lookup(channel_->content_name()));
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channel_->SetFirstPacketReceivedCallback(
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[thread = thread_, flag = signaling_thread_safety_, this]() mutable {
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thread->PostTask(ToQueuedTask(
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@ -100,8 +100,36 @@ class RtpTransceiver final
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cricket::ChannelInterface* channel() const { return channel_; }
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// Sets the Voice/VideoChannel. The caller must pass in the correct channel
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// implementation based on the type of the transceiver.
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void SetChannel(cricket::ChannelInterface* channel);
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// implementation based on the type of the transceiver. The call must
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// furthermore be made on the signaling thread.
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//
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// `channel`: The channel instance to be associated with the transceiver.
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// When a valid pointer is passed for `channel`, the state of the object
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// is expected to be newly constructed and not initalized for network
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// activity (see next parameter for more).
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//
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// NOTE: For all practical purposes, the ownership of the channel
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// object should be considered to lie with the transceiver until
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// `SetChannel()` is called again with nullptr set as the new channel.
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// Moving forward, this parameter will change to either be a
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// std::unique_ptr<> or the full construction of the channel object will
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// be moved to happen within the context of the transceiver class.
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//
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// `transport_lookup`: When `channel` points to a valid channel object, this
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// callback function will be used to look up the `RtpTransport` object
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// to associate with the channel via `BaseChannel::SetRtpTransport`.
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// The lookup function will be called on the network thread, synchronously
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// during the call to `SetChannel`. This means that the caller of
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// `SetChannel()` may provide a callback function that references state
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// that exists within the calling scope of SetChannel (e.g. a variable
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// on the stack).
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// The reason for this design is to limit the number of times we jump
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// synchronously to the network thread from the signaling thread.
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// The callback allows us to combine the transport lookup with network
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// state initialization of the channel object.
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void SetChannel(cricket::ChannelInterface* channel,
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std::function<RtpTransportInternal*(const std::string&)>
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transport_lookup);
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// Adds an RtpSender of the appropriate type to be owned by this transceiver.
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// Must not be null.
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@ -36,13 +36,19 @@ namespace webrtc {
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TEST(RtpTransceiverTest, CannotSetChannelOnStoppedTransceiver) {
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auto cm = cricket::ChannelManager::Create(
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nullptr, true, rtc::Thread::Current(), rtc::Thread::Current());
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const std::string content_name("my_mid");
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RtpTransceiver transceiver(cricket::MediaType::MEDIA_TYPE_AUDIO, cm.get());
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cricket::MockChannelInterface channel1;
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EXPECT_CALL(channel1, media_type())
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.WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
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EXPECT_CALL(channel1, content_name()).WillRepeatedly(ReturnRef(content_name));
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EXPECT_CALL(channel1, SetFirstPacketReceivedCallback(_));
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EXPECT_CALL(channel1, SetRtpTransport(_)).WillRepeatedly(Return(true));
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transceiver.SetChannel(&channel1);
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transceiver.SetChannel(&channel1, [&](const std::string& mid) {
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EXPECT_EQ(mid, content_name);
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return nullptr;
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});
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EXPECT_EQ(&channel1, transceiver.channel());
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// Stop the transceiver.
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@ -54,7 +60,7 @@ TEST(RtpTransceiverTest, CannotSetChannelOnStoppedTransceiver) {
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.WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
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// Channel can no longer be set, so this call should be a no-op.
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transceiver.SetChannel(&channel2);
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transceiver.SetChannel(&channel2, [](const std::string&) { return nullptr; });
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EXPECT_EQ(&channel1, transceiver.channel());
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}
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@ -62,14 +68,20 @@ TEST(RtpTransceiverTest, CannotSetChannelOnStoppedTransceiver) {
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TEST(RtpTransceiverTest, CanUnsetChannelOnStoppedTransceiver) {
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auto cm = cricket::ChannelManager::Create(
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nullptr, true, rtc::Thread::Current(), rtc::Thread::Current());
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const std::string content_name("my_mid");
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RtpTransceiver transceiver(cricket::MediaType::MEDIA_TYPE_VIDEO, cm.get());
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cricket::MockChannelInterface channel;
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EXPECT_CALL(channel, media_type())
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.WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_VIDEO));
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EXPECT_CALL(channel, content_name()).WillRepeatedly(ReturnRef(content_name));
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EXPECT_CALL(channel, SetFirstPacketReceivedCallback(_))
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.WillRepeatedly(testing::Return());
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EXPECT_CALL(channel, SetRtpTransport(_)).WillRepeatedly(Return(true));
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transceiver.SetChannel(&channel);
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transceiver.SetChannel(&channel, [&](const std::string& mid) {
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EXPECT_EQ(mid, content_name);
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return nullptr;
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});
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EXPECT_EQ(&channel, transceiver.channel());
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// Stop the transceiver.
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@ -77,7 +89,7 @@ TEST(RtpTransceiverTest, CanUnsetChannelOnStoppedTransceiver) {
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EXPECT_EQ(&channel, transceiver.channel());
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// Set the channel to `nullptr`.
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transceiver.SetChannel(nullptr);
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transceiver.SetChannel(nullptr, nullptr);
|
||||
EXPECT_EQ(nullptr, transceiver.channel());
|
||||
}
|
||||
|
||||
@ -279,6 +291,7 @@ TEST_F(RtpTransceiverTestForHeaderExtensions,
|
||||
|
||||
TEST_F(RtpTransceiverTestForHeaderExtensions,
|
||||
NoNegotiatedHdrExtsWithChannelWithoutNegotiation) {
|
||||
const std::string content_name("my_mid");
|
||||
EXPECT_CALL(*receiver_.get(), SetMediaChannel(_));
|
||||
EXPECT_CALL(*receiver_.get(), StopAndEndTrack());
|
||||
EXPECT_CALL(*sender_.get(), SetMediaChannel(_));
|
||||
@ -289,11 +302,16 @@ TEST_F(RtpTransceiverTestForHeaderExtensions,
|
||||
EXPECT_CALL(mock_channel, media_type())
|
||||
.WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
|
||||
EXPECT_CALL(mock_channel, media_channel()).WillRepeatedly(Return(nullptr));
|
||||
transceiver_.SetChannel(&mock_channel);
|
||||
EXPECT_CALL(mock_channel, content_name())
|
||||
.WillRepeatedly(ReturnRef(content_name));
|
||||
EXPECT_CALL(mock_channel, SetRtpTransport(_)).WillRepeatedly(Return(true));
|
||||
transceiver_.SetChannel(&mock_channel,
|
||||
[](const std::string&) { return nullptr; });
|
||||
EXPECT_THAT(transceiver_.HeaderExtensionsNegotiated(), ElementsAre());
|
||||
}
|
||||
|
||||
TEST_F(RtpTransceiverTestForHeaderExtensions, ReturnsNegotiatedHdrExts) {
|
||||
const std::string content_name("my_mid");
|
||||
EXPECT_CALL(*receiver_.get(), SetMediaChannel(_));
|
||||
EXPECT_CALL(*receiver_.get(), StopAndEndTrack());
|
||||
EXPECT_CALL(*sender_.get(), SetMediaChannel(_));
|
||||
@ -305,6 +323,9 @@ TEST_F(RtpTransceiverTestForHeaderExtensions, ReturnsNegotiatedHdrExts) {
|
||||
EXPECT_CALL(mock_channel, media_type())
|
||||
.WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
|
||||
EXPECT_CALL(mock_channel, media_channel()).WillRepeatedly(Return(nullptr));
|
||||
EXPECT_CALL(mock_channel, content_name())
|
||||
.WillRepeatedly(ReturnRef(content_name));
|
||||
EXPECT_CALL(mock_channel, SetRtpTransport(_)).WillRepeatedly(Return(true));
|
||||
|
||||
cricket::RtpHeaderExtensions extensions = {webrtc::RtpExtension("uri1", 1),
|
||||
webrtc::RtpExtension("uri2", 2)};
|
||||
@ -312,7 +333,8 @@ TEST_F(RtpTransceiverTestForHeaderExtensions, ReturnsNegotiatedHdrExts) {
|
||||
description.set_rtp_header_extensions(extensions);
|
||||
transceiver_.OnNegotiationUpdate(SdpType::kAnswer, &description);
|
||||
|
||||
transceiver_.SetChannel(&mock_channel);
|
||||
transceiver_.SetChannel(&mock_channel,
|
||||
[](const std::string&) { return nullptr; });
|
||||
EXPECT_THAT(transceiver_.HeaderExtensionsNegotiated(),
|
||||
ElementsAre(RtpHeaderExtensionCapability(
|
||||
"uri1", 1, RtpTransceiverDirection::kSendRecv),
|
||||
|
||||
@ -3522,7 +3522,7 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiverChannel(
|
||||
cricket::ChannelInterface* channel = transceiver->internal()->channel();
|
||||
if (content.rejected) {
|
||||
if (channel) {
|
||||
transceiver->internal()->SetChannel(nullptr);
|
||||
transceiver->internal()->SetChannel(nullptr, nullptr);
|
||||
DestroyChannelInterface(channel);
|
||||
}
|
||||
} else {
|
||||
@ -3537,7 +3537,9 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiverChannel(
|
||||
return RTCError(RTCErrorType::INTERNAL_ERROR,
|
||||
"Failed to create channel for mid=" + content.name);
|
||||
}
|
||||
transceiver->internal()->SetChannel(channel);
|
||||
transceiver->internal()->SetChannel(channel, [&](const std::string& mid) {
|
||||
return transport_controller()->GetRtpTransport(mid);
|
||||
});
|
||||
}
|
||||
}
|
||||
return RTCError::OK();
|
||||
@ -4714,7 +4716,10 @@ RTCError SdpOfferAnswerHandler::CreateChannels(const SessionDescription& desc) {
|
||||
return RTCError(RTCErrorType::INTERNAL_ERROR,
|
||||
"Failed to create voice channel.");
|
||||
}
|
||||
rtp_manager()->GetAudioTransceiver()->internal()->SetChannel(voice_channel);
|
||||
rtp_manager()->GetAudioTransceiver()->internal()->SetChannel(
|
||||
voice_channel, [&](const std::string& mid) {
|
||||
return transport_controller()->GetRtpTransport(mid);
|
||||
});
|
||||
}
|
||||
|
||||
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(&desc);
|
||||
@ -4725,7 +4730,10 @@ RTCError SdpOfferAnswerHandler::CreateChannels(const SessionDescription& desc) {
|
||||
return RTCError(RTCErrorType::INTERNAL_ERROR,
|
||||
"Failed to create video channel.");
|
||||
}
|
||||
rtp_manager()->GetVideoTransceiver()->internal()->SetChannel(video_channel);
|
||||
rtp_manager()->GetVideoTransceiver()->internal()->SetChannel(
|
||||
video_channel, [&](const std::string& mid) {
|
||||
return transport_controller()->GetRtpTransport(mid);
|
||||
});
|
||||
}
|
||||
|
||||
const cricket::ContentInfo* data = cricket::GetFirstDataContent(&desc);
|
||||
@ -4748,18 +4756,13 @@ cricket::VoiceChannel* SdpOfferAnswerHandler::CreateVoiceChannel(
|
||||
if (!channel_manager()->media_engine())
|
||||
return nullptr;
|
||||
|
||||
// TODO(tommi): Avoid hop to network thread.
|
||||
RtpTransportInternal* rtp_transport = pc_->GetRtpTransport(mid);
|
||||
|
||||
// TODO(bugs.webrtc.org/11992): CreateVoiceChannel internally switches to the
|
||||
// worker thread. We shouldn't be using the `call_ptr_` hack here but simply
|
||||
// be on the worker thread and use `call_` (update upstream code).
|
||||
// TODO(tommi): This hops to the worker and from the worker to the network
|
||||
// thread (blocking both signal and worker).
|
||||
return channel_manager()->CreateVoiceChannel(
|
||||
pc_->call_ptr(), pc_->configuration()->media_config, rtp_transport,
|
||||
signaling_thread(), mid, pc_->SrtpRequired(), pc_->GetCryptoOptions(),
|
||||
&ssrc_generator_, audio_options());
|
||||
pc_->call_ptr(), pc_->configuration()->media_config, signaling_thread(),
|
||||
mid, pc_->SrtpRequired(), pc_->GetCryptoOptions(), &ssrc_generator_,
|
||||
audio_options());
|
||||
}
|
||||
|
||||
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
|
||||
@ -4770,17 +4773,13 @@ cricket::VideoChannel* SdpOfferAnswerHandler::CreateVideoChannel(
|
||||
if (!channel_manager()->media_engine())
|
||||
return nullptr;
|
||||
|
||||
// NOTE: This involves a non-ideal hop (Invoke) over to the network thread.
|
||||
RtpTransportInternal* rtp_transport = pc_->GetRtpTransport(mid);
|
||||
|
||||
// TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to the
|
||||
// worker thread. We shouldn't be using the `call_ptr_` hack here but simply
|
||||
// be on the worker thread and use `call_` (update upstream code).
|
||||
return channel_manager()->CreateVideoChannel(
|
||||
pc_->call_ptr(), pc_->configuration()->media_config, rtp_transport,
|
||||
signaling_thread(), mid, pc_->SrtpRequired(), pc_->GetCryptoOptions(),
|
||||
&ssrc_generator_, video_options(),
|
||||
video_bitrate_allocator_factory_.get());
|
||||
pc_->call_ptr(), pc_->configuration()->media_config, signaling_thread(),
|
||||
mid, pc_->SrtpRequired(), pc_->GetCryptoOptions(), &ssrc_generator_,
|
||||
video_options(), video_bitrate_allocator_factory_.get());
|
||||
}
|
||||
|
||||
bool SdpOfferAnswerHandler::CreateDataChannel(const std::string& mid) {
|
||||
@ -4825,7 +4824,7 @@ void SdpOfferAnswerHandler::DestroyTransceiverChannel(
|
||||
// so if ownership of the channel object lies with the transceiver, we could
|
||||
// un-set the channel pointer and uninitialize/destruct the channel object
|
||||
// at the same time, rather than in separate steps.
|
||||
transceiver->internal()->SetChannel(nullptr);
|
||||
transceiver->internal()->SetChannel(nullptr, nullptr);
|
||||
// TODO(tommi): All channel objects end up getting deleted on the
|
||||
// worker thread (ideally should be on the network thread but the
|
||||
// MediaChannel objects are tied to the worker. Can the teardown be done
|
||||
|
||||
@ -208,7 +208,8 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase {
|
||||
webrtc::CryptoOptions(), &ssrc_generator_, transport_name);
|
||||
GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)
|
||||
->internal()
|
||||
->SetChannel(voice_channel_.get());
|
||||
->SetChannel(voice_channel_.get(),
|
||||
[](const std::string&) { return nullptr; });
|
||||
return voice_media_channel_ptr;
|
||||
}
|
||||
|
||||
@ -225,7 +226,8 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase {
|
||||
webrtc::CryptoOptions(), &ssrc_generator_, transport_name);
|
||||
GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
|
||||
->internal()
|
||||
->SetChannel(video_channel_.get());
|
||||
->SetChannel(video_channel_.get(),
|
||||
[](const std::string&) { return nullptr; });
|
||||
return video_media_channel_ptr;
|
||||
}
|
||||
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user