Change RtpReceivers to interact with the media channel directly
Currently, the RtpReceivers take a BaseChannel which is (mostly)
just used for proxying calls to the MediaChannel. This change
removes the extra layer and moves the proxying logic to RtpReceiver.
Bug: webrtc:8587
Change-Id: I01b0e3d57b4629e43d9d148cc94d6dd2941d320e
Reviewed-on: https://webrtc-review.googlesource.com/38120
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21562}diff --git a/ortc/ortcrtpreceiveradapter.cc b/ortc/ortcrtpreceiveradapter.cc
index 403ddca..67f4de6 100644
--- a/ortc/ortcrtpreceiveradapter.cc
+++ b/ortc/ortcrtpreceiveradapter.cc
@@ -150,17 +150,20 @@
}
internal_receiver_ = nullptr;
switch (kind_) {
- case cricket::MEDIA_TYPE_AUDIO:
- internal_receiver_ =
- new AudioRtpReceiver(rtc::CreateRandomUuid(), {}, ssrc,
- rtp_transport_controller_->voice_channel());
+ case cricket::MEDIA_TYPE_AUDIO: {
+ auto* voice_channel = rtp_transport_controller_->voice_channel();
+ internal_receiver_ = new AudioRtpReceiver(
+ rtp_transport_controller_->worker_thread(), rtc::CreateRandomUuid(),
+ {}, ssrc, (voice_channel ? voice_channel->media_channel() : nullptr));
break;
- case cricket::MEDIA_TYPE_VIDEO:
- internal_receiver_ =
- new VideoRtpReceiver(rtc::CreateRandomUuid(), {},
- rtp_transport_controller_->worker_thread(), ssrc,
- rtp_transport_controller_->video_channel());
+ }
+ case cricket::MEDIA_TYPE_VIDEO: {
+ auto* video_channel = rtp_transport_controller_->video_channel();
+ internal_receiver_ = new VideoRtpReceiver(
+ rtp_transport_controller_->worker_thread(), rtc::CreateRandomUuid(),
+ {}, ssrc, (video_channel ? video_channel->media_channel() : nullptr));
break;
+ }
case cricket::MEDIA_TYPE_DATA:
RTC_NOTREACHED();
}
diff --git a/pc/channel.cc b/pc/channel.cc
index 14ab3df..5c91c0b 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -38,13 +38,6 @@
using webrtc::SdpType;
namespace {
-// See comment below for why we need to use a pointer to a unique_ptr.
-bool SetRawAudioSink_w(VoiceMediaChannel* channel,
- uint32_t ssrc,
- std::unique_ptr<webrtc::AudioSinkInterface>* sink) {
- channel->SetRawAudioSink(ssrc, std::move(*sink));
- return true;
-}
struct SendPacketMessageData : public rtc::MessageData {
rtc::CopyOnWriteBuffer packet;
@@ -1232,22 +1225,6 @@
Bind(&VoiceChannel::InsertDtmf_w, this, ssrc, event_code, duration));
}
-bool VoiceChannel::SetOutputVolume(uint32_t ssrc, double volume) {
- return InvokeOnWorker<bool>(
- RTC_FROM_HERE,
- Bind(&VoiceMediaChannel::SetOutputVolume, media_channel(), ssrc, volume));
-}
-
-void VoiceChannel::SetRawAudioSink(
- uint32_t ssrc,
- std::unique_ptr<webrtc::AudioSinkInterface> sink) {
- // We need to work around Bind's lack of support for unique_ptr and ownership
- // passing. So we invoke to our own little routine that gets a pointer to
- // our local variable. This is OK since we're synchronously invoking.
- InvokeOnWorker<bool>(RTC_FROM_HERE,
- Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink));
-}
-
webrtc::RtpParameters VoiceChannel::GetRtpSendParameters(uint32_t ssrc) const {
return worker_thread()->Invoke<webrtc::RtpParameters>(
RTC_FROM_HERE, Bind(&VoiceChannel::GetRtpSendParameters_w, this, ssrc));
@@ -1271,46 +1248,11 @@
return media_channel()->SetRtpSendParameters(ssrc, parameters);
}
-webrtc::RtpParameters VoiceChannel::GetRtpReceiveParameters(
- uint32_t ssrc) const {
- return worker_thread()->Invoke<webrtc::RtpParameters>(
- RTC_FROM_HERE,
- Bind(&VoiceChannel::GetRtpReceiveParameters_w, this, ssrc));
-}
-
-webrtc::RtpParameters VoiceChannel::GetRtpReceiveParameters_w(
- uint32_t ssrc) const {
- return media_channel()->GetRtpReceiveParameters(ssrc);
-}
-
-bool VoiceChannel::SetRtpReceiveParameters(
- uint32_t ssrc,
- const webrtc::RtpParameters& parameters) {
- return InvokeOnWorker<bool>(
- RTC_FROM_HERE,
- Bind(&VoiceChannel::SetRtpReceiveParameters_w, this, ssrc, parameters));
-}
-
-bool VoiceChannel::SetRtpReceiveParameters_w(uint32_t ssrc,
- webrtc::RtpParameters parameters) {
- return media_channel()->SetRtpReceiveParameters(ssrc, parameters);
-}
-
bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
return InvokeOnWorker<bool>(RTC_FROM_HERE, Bind(&VoiceMediaChannel::GetStats,
media_channel(), stats));
}
-std::vector<webrtc::RtpSource> VoiceChannel::GetSources(uint32_t ssrc) const {
- return worker_thread()->Invoke<std::vector<webrtc::RtpSource>>(
- RTC_FROM_HERE, Bind(&VoiceChannel::GetSources_w, this, ssrc));
-}
-
-std::vector<webrtc::RtpSource> VoiceChannel::GetSources_w(uint32_t ssrc) const {
- RTC_DCHECK(worker_thread()->IsCurrent());
- return media_channel()->GetSources(ssrc);
-}
-
void VoiceChannel::StartMediaMonitor(int cms) {
media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(),
rtc::Thread::Current()));
@@ -1559,14 +1501,6 @@
Deinit();
}
-bool VideoChannel::SetSink(uint32_t ssrc,
- rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
- worker_thread()->Invoke<void>(
- RTC_FROM_HERE,
- Bind(&VideoMediaChannel::SetSink, media_channel(), ssrc, sink));
- return true;
-}
-
bool VideoChannel::SetVideoSend(
uint32_t ssrc,
bool mute,
@@ -1600,31 +1534,6 @@
return media_channel()->SetRtpSendParameters(ssrc, parameters);
}
-webrtc::RtpParameters VideoChannel::GetRtpReceiveParameters(
- uint32_t ssrc) const {
- return worker_thread()->Invoke<webrtc::RtpParameters>(
- RTC_FROM_HERE,
- Bind(&VideoChannel::GetRtpReceiveParameters_w, this, ssrc));
-}
-
-webrtc::RtpParameters VideoChannel::GetRtpReceiveParameters_w(
- uint32_t ssrc) const {
- return media_channel()->GetRtpReceiveParameters(ssrc);
-}
-
-bool VideoChannel::SetRtpReceiveParameters(
- uint32_t ssrc,
- const webrtc::RtpParameters& parameters) {
- return InvokeOnWorker<bool>(
- RTC_FROM_HERE,
- Bind(&VideoChannel::SetRtpReceiveParameters_w, this, ssrc, parameters));
-}
-
-bool VideoChannel::SetRtpReceiveParameters_w(uint32_t ssrc,
- webrtc::RtpParameters parameters) {
- return media_channel()->SetRtpReceiveParameters(ssrc, parameters);
-}
-
void VideoChannel::UpdateMediaSendRecvState_w() {
// Send outgoing data if we're the active call, we have the remote content,
// and we have had some form of connectivity.
diff --git a/pc/channel.h b/pc/channel.h
index a7d140e..ec9f913 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -490,22 +490,13 @@
// The valid value for the |event| are 0 which corresponding to DTMF
// event 0-9, *, #, A-D.
bool InsertDtmf(uint32_t ssrc, int event_code, int duration);
- bool SetOutputVolume(uint32_t ssrc, double volume);
- void SetRawAudioSink(uint32_t ssrc,
- std::unique_ptr<webrtc::AudioSinkInterface> sink);
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const;
bool SetRtpSendParameters(uint32_t ssrc,
const webrtc::RtpParameters& parameters);
- webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const;
- bool SetRtpReceiveParameters(uint32_t ssrc,
- const webrtc::RtpParameters& parameters);
// Get statistics about the current media session.
bool GetStats(VoiceMediaInfo* stats);
- std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const;
- std::vector<webrtc::RtpSource> GetSources_w(uint32_t ssrc) const;
-
// Monitoring functions
sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&>
SignalConnectionMonitor;
@@ -524,9 +515,6 @@
void GetActiveStreams_w(AudioInfo::StreamList* actives);
webrtc::RtpParameters GetRtpSendParameters_w(uint32_t ssrc) const;
bool SetRtpSendParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
- webrtc::RtpParameters GetRtpReceiveParameters_w(uint32_t ssrc) const;
- bool SetRtpReceiveParameters_w(uint32_t ssrc,
- webrtc::RtpParameters parameters);
cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_AUDIO; }
private:
@@ -543,7 +531,6 @@
std::string* error_desc) override;
void HandleEarlyMediaTimeout();
bool InsertDtmf_w(uint32_t ssrc, int event, int duration);
- bool SetOutputVolume_w(uint32_t ssrc, double volume);
void OnMessage(rtc::Message* pmsg) override;
void OnConnectionMonitorUpdate(
@@ -584,8 +571,6 @@
return static_cast<VideoMediaChannel*>(BaseChannel::media_channel());
}
- bool SetSink(uint32_t ssrc,
- rtc::VideoSinkInterface<webrtc::VideoFrame>* sink);
void FillBitrateInfo(BandwidthEstimationInfo* bwe_info);
// Get statistics about the current media session.
bool GetStats(VideoMediaInfo* stats);
@@ -606,9 +591,6 @@
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const;
bool SetRtpSendParameters(uint32_t ssrc,
const webrtc::RtpParameters& parameters);
- webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const;
- bool SetRtpReceiveParameters(uint32_t ssrc,
- const webrtc::RtpParameters& parameters);
cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_VIDEO; }
private:
@@ -623,9 +605,6 @@
bool GetStats_w(VideoMediaInfo* stats);
webrtc::RtpParameters GetRtpSendParameters_w(uint32_t ssrc) const;
bool SetRtpSendParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
- webrtc::RtpParameters GetRtpReceiveParameters_w(uint32_t ssrc) const;
- bool SetRtpReceiveParameters_w(uint32_t ssrc,
- webrtc::RtpParameters parameters);
void OnConnectionMonitorUpdate(
ConnectionMonitor* monitor,
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index 0d155c4..ae5782c 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -2399,68 +2399,6 @@
Base::TestOnTransportReadyToSendWithRtcpMux();
}
-// Test that we can scale the output volume properly for 1:1 calls.
-TEST_F(VoiceChannelSingleThreadTest, TestScaleVolume1to1Call) {
- CreateChannels(0, 0);
- EXPECT_TRUE(SendInitiate());
- EXPECT_TRUE(SendAccept());
- double volume;
-
- // Default is (1.0).
- EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
- EXPECT_DOUBLE_EQ(1.0, volume);
- // invalid ssrc.
- EXPECT_FALSE(media_channel1_->GetOutputVolume(3, &volume));
-
- // Set scale to (1.5).
- EXPECT_TRUE(channel1_->SetOutputVolume(0, 1.5));
- EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
- EXPECT_DOUBLE_EQ(1.5, volume);
-
- // Set scale to (0).
- EXPECT_TRUE(channel1_->SetOutputVolume(0, 0.0));
- EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
- EXPECT_DOUBLE_EQ(0.0, volume);
-}
-
-// Test that we can scale the output volume properly for multiway calls.
-TEST_F(VoiceChannelSingleThreadTest, TestScaleVolumeMultiwayCall) {
- CreateChannels(0, 0);
- EXPECT_TRUE(SendInitiate());
- EXPECT_TRUE(SendAccept());
- EXPECT_TRUE(AddStream1(1));
- EXPECT_TRUE(AddStream1(2));
-
- double volume;
- // Default is (1.0).
- EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
- EXPECT_DOUBLE_EQ(1.0, volume);
- EXPECT_TRUE(media_channel1_->GetOutputVolume(1, &volume));
- EXPECT_DOUBLE_EQ(1.0, volume);
- EXPECT_TRUE(media_channel1_->GetOutputVolume(2, &volume));
- EXPECT_DOUBLE_EQ(1.0, volume);
- // invalid ssrc.
- EXPECT_FALSE(media_channel1_->GetOutputVolume(3, &volume));
-
- // Set scale to (1.5) for ssrc = 1.
- EXPECT_TRUE(channel1_->SetOutputVolume(1, 1.5));
- EXPECT_TRUE(media_channel1_->GetOutputVolume(1, &volume));
- EXPECT_DOUBLE_EQ(1.5, volume);
- EXPECT_TRUE(media_channel1_->GetOutputVolume(2, &volume));
- EXPECT_DOUBLE_EQ(1.0, volume);
- EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
- EXPECT_DOUBLE_EQ(1.0, volume);
-
- // Set scale to (0) for all ssrcs.
- EXPECT_TRUE(channel1_->SetOutputVolume(0, 0.0));
- EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
- EXPECT_DOUBLE_EQ(0.0, volume);
- EXPECT_TRUE(media_channel1_->GetOutputVolume(1, &volume));
- EXPECT_DOUBLE_EQ(0.0, volume);
- EXPECT_TRUE(media_channel1_->GetOutputVolume(2, &volume));
- EXPECT_DOUBLE_EQ(0.0, volume);
-}
-
TEST_F(VoiceChannelSingleThreadTest, SendBundleToBundle) {
Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), false, false);
}
@@ -2750,68 +2688,6 @@
Base::TestOnTransportReadyToSendWithRtcpMux();
}
-// Test that we can scale the output volume properly for 1:1 calls.
-TEST_F(VoiceChannelDoubleThreadTest, TestScaleVolume1to1Call) {
- CreateChannels(0, 0);
- EXPECT_TRUE(SendInitiate());
- EXPECT_TRUE(SendAccept());
- double volume;
-
- // Default is (1.0).
- EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
- EXPECT_DOUBLE_EQ(1.0, volume);
- // invalid ssrc.
- EXPECT_FALSE(media_channel1_->GetOutputVolume(3, &volume));
-
- // Set scale to (1.5).
- EXPECT_TRUE(channel1_->SetOutputVolume(0, 1.5));
- EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
- EXPECT_DOUBLE_EQ(1.5, volume);
-
- // Set scale to (0).
- EXPECT_TRUE(channel1_->SetOutputVolume(0, 0.0));
- EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
- EXPECT_DOUBLE_EQ(0.0, volume);
-}
-
-// Test that we can scale the output volume properly for multiway calls.
-TEST_F(VoiceChannelDoubleThreadTest, TestScaleVolumeMultiwayCall) {
- CreateChannels(0, 0);
- EXPECT_TRUE(SendInitiate());
- EXPECT_TRUE(SendAccept());
- EXPECT_TRUE(AddStream1(1));
- EXPECT_TRUE(AddStream1(2));
-
- double volume;
- // Default is (1.0).
- EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
- EXPECT_DOUBLE_EQ(1.0, volume);
- EXPECT_TRUE(media_channel1_->GetOutputVolume(1, &volume));
- EXPECT_DOUBLE_EQ(1.0, volume);
- EXPECT_TRUE(media_channel1_->GetOutputVolume(2, &volume));
- EXPECT_DOUBLE_EQ(1.0, volume);
- // invalid ssrc.
- EXPECT_FALSE(media_channel1_->GetOutputVolume(3, &volume));
-
- // Set scale to (1.5) for ssrc = 1.
- EXPECT_TRUE(channel1_->SetOutputVolume(1, 1.5));
- EXPECT_TRUE(media_channel1_->GetOutputVolume(1, &volume));
- EXPECT_DOUBLE_EQ(1.5, volume);
- EXPECT_TRUE(media_channel1_->GetOutputVolume(2, &volume));
- EXPECT_DOUBLE_EQ(1.0, volume);
- EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
- EXPECT_DOUBLE_EQ(1.0, volume);
-
- // Set scale to (0) for all ssrcs.
- EXPECT_TRUE(channel1_->SetOutputVolume(0, 0.0));
- EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
- EXPECT_DOUBLE_EQ(0.0, volume);
- EXPECT_TRUE(media_channel1_->GetOutputVolume(1, &volume));
- EXPECT_DOUBLE_EQ(0.0, volume);
- EXPECT_TRUE(media_channel1_->GetOutputVolume(2, &volume));
- EXPECT_DOUBLE_EQ(0.0, volume);
-}
-
TEST_F(VoiceChannelDoubleThreadTest, SendBundleToBundle) {
Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), false, false);
}
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index 020f3c3..3bab2e1d 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -1283,14 +1283,15 @@
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), new AudioRtpSender(nullptr, stats_.get()));
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
- signaling_thread(), new AudioRtpReceiver(receiver_id, {}, 0, nullptr));
+ signaling_thread(),
+ new AudioRtpReceiver(worker_thread(), receiver_id, {}, 0, nullptr));
} else {
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, media_type);
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), new VideoRtpSender(nullptr));
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(),
- new VideoRtpReceiver(receiver_id, {}, worker_thread(), 0, nullptr));
+ new VideoRtpReceiver(worker_thread(), receiver_id, {}, 0, nullptr));
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
transceiver = RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
@@ -2713,8 +2714,9 @@
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(),
- new AudioRtpReceiver(remote_sender_info.sender_id, streams,
- remote_sender_info.first_ssrc, voice_channel()));
+ new AudioRtpReceiver(worker_thread(), remote_sender_info.sender_id,
+ streams, remote_sender_info.first_ssrc,
+ voice_media_channel()));
stream->AddTrack(
static_cast<AudioTrackInterface*>(receiver->internal()->track().get()));
GetAudioTransceiver()->internal()->AddReceiver(receiver);
@@ -2729,9 +2731,9 @@
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(),
- new VideoRtpReceiver(remote_sender_info.sender_id, streams,
- worker_thread(), remote_sender_info.first_ssrc,
- video_channel()));
+ new VideoRtpReceiver(worker_thread(), remote_sender_info.sender_id,
+ streams, remote_sender_info.first_ssrc,
+ video_media_channel()));
stream->AddTrack(
static_cast<VideoTrackInterface*>(receiver->internal()->track().get()));
GetVideoTransceiver()->internal()->AddReceiver(receiver);
diff --git a/pc/peerconnection.h b/pc/peerconnection.h
index 3fd944c..4c7973b 100644
--- a/pc/peerconnection.h
+++ b/pc/peerconnection.h
@@ -317,6 +317,14 @@
// Implements MessageHandler.
void OnMessage(rtc::Message* msg) override;
+ cricket::VoiceMediaChannel* voice_media_channel() const {
+ return voice_channel() ? voice_channel()->media_channel() : nullptr;
+ }
+
+ cricket::VideoMediaChannel* video_media_channel() const {
+ return video_channel() ? video_channel()->media_channel() : nullptr;
+ }
+
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
GetSendersInternal() const;
std::vector<
diff --git a/pc/remoteaudiosource.cc b/pc/remoteaudiosource.cc
index d88fa69..a260be8 100644
--- a/pc/remoteaudiosource.cc
+++ b/pc/remoteaudiosource.cc
@@ -18,6 +18,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/logging.h"
+#include "rtc_base/ptr_util.h"
#include "rtc_base/thread.h"
namespace webrtc {
@@ -38,11 +39,12 @@
};
rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create(
- uint32_t ssrc,
- cricket::VoiceChannel* channel) {
+ rtc::Thread* worker_thread,
+ cricket::VoiceMediaChannel* media_channel,
+ uint32_t ssrc) {
rtc::scoped_refptr<RemoteAudioSource> ret(
new rtc::RefCountedObject<RemoteAudioSource>());
- ret->Initialize(ssrc, channel);
+ ret->Initialize(worker_thread, media_channel, ssrc);
return ret;
}
@@ -58,14 +60,16 @@
RTC_DCHECK(sinks_.empty());
}
-void RemoteAudioSource::Initialize(uint32_t ssrc,
- cricket::VoiceChannel* channel) {
+void RemoteAudioSource::Initialize(rtc::Thread* worker_thread,
+ cricket::VoiceMediaChannel* media_channel,
+ uint32_t ssrc) {
RTC_DCHECK(main_thread_->IsCurrent());
// To make sure we always get notified when the channel goes out of scope,
// we register for callbacks here and not on demand in AddSink.
- if (channel) { // May be null in tests.
- channel->SetRawAudioSink(
- ssrc, std::unique_ptr<AudioSinkInterface>(new Sink(this)));
+ if (media_channel) { // May be null in tests.
+ worker_thread->Invoke<void>(RTC_FROM_HERE, [&] {
+ media_channel->SetRawAudioSink(ssrc, rtc::MakeUnique<Sink>(this));
+ });
}
}
diff --git a/pc/remoteaudiosource.h b/pc/remoteaudiosource.h
index d35fb04..eadbcca 100644
--- a/pc/remoteaudiosource.h
+++ b/pc/remoteaudiosource.h
@@ -33,8 +33,9 @@
public:
// Creates an instance of RemoteAudioSource.
static rtc::scoped_refptr<RemoteAudioSource> Create(
- uint32_t ssrc,
- cricket::VoiceChannel* channel);
+ rtc::Thread* worker_thread,
+ cricket::VoiceMediaChannel* media_channel,
+ uint32_t ssrc);
// MediaSourceInterface implementation.
MediaSourceInterface::SourceState state() const override;
@@ -49,7 +50,9 @@
// Post construction initialize where we can do things like save a reference
// to ourselves (need to be fully constructed).
- void Initialize(uint32_t ssrc, cricket::VoiceChannel* channel);
+ void Initialize(rtc::Thread* worker_thread,
+ cricket::VoiceMediaChannel* media_channel,
+ uint32_t ssrc);
private:
typedef std::list<AudioObserver*> AudioObserverList;
diff --git a/pc/rtpreceiver.cc b/pc/rtpreceiver.cc
index 4968ac1..33ad899 100644
--- a/pc/rtpreceiver.cc
+++ b/pc/rtpreceiver.cc
@@ -22,27 +22,27 @@
namespace webrtc {
AudioRtpReceiver::AudioRtpReceiver(
+ rtc::Thread* worker_thread,
const std::string& receiver_id,
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams,
uint32_t ssrc,
- cricket::VoiceChannel* channel)
- : id_(receiver_id),
+ cricket::VoiceMediaChannel* media_channel)
+ : worker_thread_(worker_thread),
+ id_(receiver_id),
ssrc_(ssrc),
- channel_(channel),
track_(AudioTrackProxy::Create(
rtc::Thread::Current(),
- AudioTrack::Create(receiver_id,
- RemoteAudioSource::Create(ssrc, channel)))),
+ AudioTrack::Create(
+ receiver_id,
+ RemoteAudioSource::Create(worker_thread, media_channel, ssrc)))),
streams_(std::move(streams)),
cached_track_enabled_(track_->enabled()) {
+ RTC_DCHECK(worker_thread_);
RTC_DCHECK(track_->GetSource()->remote());
track_->RegisterObserver(this);
track_->GetSource()->RegisterAudioObserver(this);
+ SetMediaChannel(media_channel);
Reconfigure();
- if (channel_) {
- channel_->SignalFirstPacketReceived.connect(
- this, &AudioRtpReceiver::OnFirstPacketReceived);
- }
}
AudioRtpReceiver::~AudioRtpReceiver() {
@@ -58,11 +58,20 @@
}
}
+bool AudioRtpReceiver::SetOutputVolume(double volume) {
+ RTC_DCHECK_GE(volume, 0.0);
+ RTC_DCHECK_LE(volume, 10.0);
+ RTC_DCHECK(media_channel_);
+ return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
+ return media_channel_->SetOutputVolume(ssrc_, volume);
+ });
+}
+
void AudioRtpReceiver::OnSetVolume(double volume) {
RTC_DCHECK_GE(volume, 0);
RTC_DCHECK_LE(volume, 10);
cached_volume_ = volume;
- if (!channel_) {
+ if (!media_channel_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::OnSetVolume: No audio channel exists.";
return;
@@ -71,25 +80,29 @@
// corresponding WebRtc Voice Engine channel will be 0. So we do not allow
// setting the volume to the source when the track is disabled.
if (!stopped_ && track_->enabled()) {
- if (!channel_->SetOutputVolume(ssrc_, cached_volume_)) {
+ if (!SetOutputVolume(cached_volume_)) {
RTC_NOTREACHED();
}
}
}
RtpParameters AudioRtpReceiver::GetParameters() const {
- if (!channel_ || stopped_) {
+ if (!media_channel_ || stopped_) {
return RtpParameters();
}
- return channel_->GetRtpReceiveParameters(ssrc_);
+ return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
+ return media_channel_->GetRtpReceiveParameters(ssrc_);
+ });
}
bool AudioRtpReceiver::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "AudioRtpReceiver::SetParameters");
- if (!channel_ || stopped_) {
+ if (!media_channel_ || stopped_) {
return false;
}
- return channel_->SetRtpReceiveParameters(ssrc_, parameters);
+ return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
+ return media_channel_->SetRtpReceiveParameters(ssrc_, parameters);
+ });
}
void AudioRtpReceiver::Stop() {
@@ -97,27 +110,27 @@
if (stopped_) {
return;
}
- if (channel_) {
+ if (media_channel_) {
// Allow that SetOutputVolume fail. This is the normal case when the
// underlying media channel has already been deleted.
- channel_->SetOutputVolume(ssrc_, 0);
+ SetOutputVolume(0.0);
}
stopped_ = true;
}
std::vector<RtpSource> AudioRtpReceiver::GetSources() const {
- return channel_->GetSources(ssrc_);
+ return worker_thread_->Invoke<std::vector<RtpSource>>(
+ RTC_FROM_HERE, [&] { return media_channel_->GetSources(ssrc_); });
}
void AudioRtpReceiver::Reconfigure() {
RTC_DCHECK(!stopped_);
- if (!channel_) {
+ if (!media_channel_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::Reconfigure: No audio channel exists.";
return;
}
- if (!channel_->SetOutputVolume(ssrc_,
- track_->enabled() ? cached_volume_ : 0)) {
+ if (!SetOutputVolume(track_->enabled() ? cached_volume_ : 0)) {
RTC_NOTREACHED();
}
}
@@ -130,18 +143,12 @@
}
}
-void AudioRtpReceiver::SetChannel(cricket::VoiceChannel* channel) {
- if (channel_) {
- channel_->SignalFirstPacketReceived.disconnect(this);
- }
- channel_ = channel;
- if (channel_) {
- channel_->SignalFirstPacketReceived.connect(
- this, &AudioRtpReceiver::OnFirstPacketReceived);
- }
+void AudioRtpReceiver::SetMediaChannel(
+ cricket::VoiceMediaChannel* media_channel) {
+ media_channel_ = media_channel;
}
-void AudioRtpReceiver::OnFirstPacketReceived(cricket::BaseChannel* channel) {
+void AudioRtpReceiver::NotifyFirstPacketReceived() {
if (observer_) {
observer_->OnFirstPacketReceived(media_type());
}
@@ -149,14 +156,14 @@
}
VideoRtpReceiver::VideoRtpReceiver(
+ rtc::Thread* worker_thread,
const std::string& track_id,
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams,
- rtc::Thread* worker_thread,
uint32_t ssrc,
- cricket::VideoChannel* channel)
- : id_(track_id),
+ cricket::VideoMediaChannel* media_channel)
+ : worker_thread_(worker_thread),
+ id_(track_id),
ssrc_(ssrc),
- channel_(channel),
source_(new RefCountedObject<VideoTrackSource>(&broadcaster_,
true /* remote */)),
track_(VideoTrackProxy::Create(
@@ -169,19 +176,9 @@
source_),
worker_thread))),
streams_(std::move(streams)) {
+ RTC_DCHECK(worker_thread_);
source_->SetState(MediaSourceInterface::kLive);
- if (!channel_) {
- RTC_LOG(LS_ERROR)
- << "VideoRtpReceiver::VideoRtpReceiver: No video channel exists.";
- } else {
- if (!channel_->SetSink(ssrc_, &broadcaster_)) {
- RTC_NOTREACHED();
- }
- }
- if (channel_) {
- channel_->SignalFirstPacketReceived.connect(
- this, &VideoRtpReceiver::OnFirstPacketReceived);
- }
+ SetMediaChannel(media_channel);
}
VideoRtpReceiver::~VideoRtpReceiver() {
@@ -190,19 +187,29 @@
Stop();
}
+bool VideoRtpReceiver::SetSink(rtc::VideoSinkInterface<VideoFrame>* sink) {
+ RTC_DCHECK(media_channel_);
+ return worker_thread_->Invoke<bool>(
+ RTC_FROM_HERE, [&] { return media_channel_->SetSink(ssrc_, sink); });
+}
+
RtpParameters VideoRtpReceiver::GetParameters() const {
- if (!channel_ || stopped_) {
+ if (!media_channel_ || stopped_) {
return RtpParameters();
}
- return channel_->GetRtpReceiveParameters(ssrc_);
+ return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
+ return media_channel_->GetRtpReceiveParameters(ssrc_);
+ });
}
bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "VideoRtpReceiver::SetParameters");
- if (!channel_ || stopped_) {
+ if (!media_channel_ || stopped_) {
return false;
}
- return channel_->SetRtpReceiveParameters(ssrc_, parameters);
+ return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
+ return media_channel_->SetRtpReceiveParameters(ssrc_, parameters);
+ });
}
void VideoRtpReceiver::Stop() {
@@ -212,12 +219,12 @@
}
source_->SetState(MediaSourceInterface::kEnded);
source_->OnSourceDestroyed();
- if (!channel_) {
+ if (!media_channel_) {
RTC_LOG(LS_WARNING) << "VideoRtpReceiver::Stop: No video channel exists.";
} else {
// Allow that SetSink fail. This is the normal case when the underlying
// media channel has already been deleted.
- channel_->SetSink(ssrc_, nullptr);
+ SetSink(nullptr);
}
stopped_ = true;
}
@@ -230,22 +237,20 @@
}
}
-void VideoRtpReceiver::SetChannel(cricket::VideoChannel* channel) {
- if (channel_) {
- channel_->SignalFirstPacketReceived.disconnect(this);
- channel_->SetSink(ssrc_, nullptr);
+void VideoRtpReceiver::SetMediaChannel(
+ cricket::VideoMediaChannel* media_channel) {
+ if (media_channel_) {
+ SetSink(nullptr);
}
- channel_ = channel;
- if (channel_) {
- if (!channel_->SetSink(ssrc_, &broadcaster_)) {
+ media_channel_ = media_channel;
+ if (media_channel_) {
+ if (!SetSink(&broadcaster_)) {
RTC_NOTREACHED();
}
- channel_->SignalFirstPacketReceived.connect(
- this, &VideoRtpReceiver::OnFirstPacketReceived);
}
}
-void VideoRtpReceiver::OnFirstPacketReceived(cricket::BaseChannel* channel) {
+void VideoRtpReceiver::NotifyFirstPacketReceived() {
if (observer_) {
observer_->OnFirstPacketReceived(media_type());
}
diff --git a/pc/rtpreceiver.h b/pc/rtpreceiver.h
index 6df4642..0ce6373 100644
--- a/pc/rtpreceiver.h
+++ b/pc/rtpreceiver.h
@@ -23,11 +23,9 @@
#include "api/mediastreaminterface.h"
#include "api/rtpreceiverinterface.h"
#include "media/base/videobroadcaster.h"
-#include "pc/channel.h"
#include "pc/remoteaudiosource.h"
#include "pc/videotracksource.h"
#include "rtc_base/basictypes.h"
-#include "rtc_base/sigslot.h"
namespace webrtc {
@@ -38,22 +36,26 @@
// This SSRC is used as an identifier for the receiver between the API layer
// and the WebRtcVideoEngine, WebRtcVoiceEngine layer.
virtual uint32_t ssrc() const = 0;
+
+ // Call this to notify the RtpReceiver when the first packet has been received
+ // on the corresponding channel.
+ virtual void NotifyFirstPacketReceived() = 0;
};
class AudioRtpReceiver : public ObserverInterface,
public AudioSourceInterface::AudioObserver,
- public rtc::RefCountedObject<RtpReceiverInternal>,
- public sigslot::has_slots<> {
+ public rtc::RefCountedObject<RtpReceiverInternal> {
public:
// An SSRC of 0 will create a receiver that will match the first SSRC it
// sees.
// TODO(deadbeef): Use rtc::Optional, or have another constructor that
// doesn't take an SSRC, and make this one DCHECK(ssrc != 0).
AudioRtpReceiver(
+ rtc::Thread* worker_thread,
const std::string& receiver_id,
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams,
uint32_t ssrc,
- cricket::VoiceChannel* channel);
+ cricket::VoiceMediaChannel* media_channel);
virtual ~AudioRtpReceiver();
// ObserverInterface implementation
@@ -87,22 +89,24 @@
// RtpReceiverInternal implementation.
void Stop() override;
uint32_t ssrc() const override { return ssrc_; }
+ void NotifyFirstPacketReceived() override;
void SetObserver(RtpReceiverObserverInterface* observer) override;
// Does not take ownership.
- // Should call SetChannel(nullptr) before |channel| is destroyed.
- void SetChannel(cricket::VoiceChannel* channel);
+ // Should call SetMediaChannel(nullptr) before |media_channel| is destroyed.
+ void SetMediaChannel(cricket::VoiceMediaChannel* media_channel);
std::vector<RtpSource> GetSources() const override;
private:
void Reconfigure();
- void OnFirstPacketReceived(cricket::BaseChannel* channel);
+ bool SetOutputVolume(double volume);
+ rtc::Thread* const worker_thread_;
const std::string id_;
const uint32_t ssrc_;
- cricket::VoiceChannel* channel_;
+ cricket::VoiceMediaChannel* media_channel_ = nullptr;
const rtc::scoped_refptr<AudioTrackInterface> track_;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
bool cached_track_enabled_;
@@ -112,17 +116,16 @@
bool received_first_packet_ = false;
};
-class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal>,
- public sigslot::has_slots<> {
+class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
public:
// An SSRC of 0 will create a receiver that will match the first SSRC it
// sees.
VideoRtpReceiver(
+ rtc::Thread* worker_thread,
const std::string& track_id,
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams,
- rtc::Thread* worker_thread,
uint32_t ssrc,
- cricket::VideoChannel* channel);
+ cricket::VideoMediaChannel* media_channel);
virtual ~VideoRtpReceiver();
@@ -151,19 +154,21 @@
// RtpReceiverInternal implementation.
void Stop() override;
uint32_t ssrc() const override { return ssrc_; }
+ void NotifyFirstPacketReceived() override;
void SetObserver(RtpReceiverObserverInterface* observer) override;
// Does not take ownership.
- // Should call SetChannel(nullptr) before |channel| is destroyed.
- void SetChannel(cricket::VideoChannel* channel);
+ // Should call SetMediaChannel(nullptr) before |media_channel| is destroyed.
+ void SetMediaChannel(cricket::VideoMediaChannel* media_channel);
private:
- void OnFirstPacketReceived(cricket::BaseChannel* channel);
+ bool SetSink(rtc::VideoSinkInterface<VideoFrame>* sink);
+ rtc::Thread* const worker_thread_;
std::string id_;
uint32_t ssrc_;
- cricket::VideoChannel* channel_;
+ cricket::VideoMediaChannel* media_channel_ = nullptr;
// |broadcaster_| is needed since the decoder can only handle one sink.
// It might be better if the decoder can handle multiple sinks and consider
// the VideoSinkWants.
diff --git a/pc/rtpsenderreceiver_unittest.cc b/pc/rtpsenderreceiver_unittest.cc
index 6334f72..f1ba6d8 100644
--- a/pc/rtpsenderreceiver_unittest.cc
+++ b/pc/rtpsenderreceiver_unittest.cc
@@ -169,7 +169,8 @@
void CreateAudioRtpReceiver(
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {}) {
audio_rtp_receiver_ = new AudioRtpReceiver(
- kAudioTrackId, std::move(streams), kAudioSsrc, voice_channel_);
+ rtc::Thread::Current(), kAudioTrackId, std::move(streams), kAudioSsrc,
+ voice_media_channel_);
audio_track_ = audio_rtp_receiver_->audio_track();
VerifyVoiceChannelOutput();
}
@@ -177,8 +178,8 @@
void CreateVideoRtpReceiver(
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {}) {
video_rtp_receiver_ = new VideoRtpReceiver(
- kVideoTrackId, std::move(streams), rtc::Thread::Current(), kVideoSsrc,
- video_channel_);
+ rtc::Thread::Current(), kVideoTrackId, std::move(streams), kVideoSsrc,
+ video_media_channel_);
video_track_ = video_rtp_receiver_->video_track();
VerifyVideoChannelOutput();
}
diff --git a/pc/rtptransceiver.cc b/pc/rtptransceiver.cc
index 61ebdc7..0dfc8b5 100644
--- a/pc/rtptransceiver.cc
+++ b/pc/rtptransceiver.cc
@@ -46,7 +46,18 @@
if (channel) {
RTC_DCHECK_EQ(media_type(), channel->media_type());
}
+
+ if (channel_) {
+ channel_->SignalFirstPacketReceived.disconnect(this);
+ }
+
channel_ = channel;
+
+ if (channel_) {
+ channel_->SignalFirstPacketReceived.connect(
+ this, &RtpTransceiver::OnFirstPacketReceived);
+ }
+
for (auto sender : senders_) {
if (media_type() == cricket::MEDIA_TYPE_AUDIO) {
static_cast<AudioRtpSender*>(sender->internal())
@@ -56,16 +67,21 @@
->SetChannel(static_cast<cricket::VideoChannel*>(channel));
}
}
+
for (auto receiver : receivers_) {
if (!channel) {
receiver->internal()->Stop();
}
if (media_type() == cricket::MEDIA_TYPE_AUDIO) {
+ auto* voice_channel = static_cast<cricket::VoiceChannel*>(channel);
static_cast<AudioRtpReceiver*>(receiver->internal())
- ->SetChannel(static_cast<cricket::VoiceChannel*>(channel));
+ ->SetMediaChannel(voice_channel ? voice_channel->media_channel()
+ : nullptr);
} else {
+ auto* video_channel = static_cast<cricket::VideoChannel*>(channel);
static_cast<VideoRtpReceiver*>(receiver->internal())
- ->SetChannel(static_cast<cricket::VideoChannel*>(channel));
+ ->SetMediaChannel(video_channel ? video_channel->media_channel()
+ : nullptr);
}
}
}
@@ -136,6 +152,12 @@
return mid_;
}
+void RtpTransceiver::OnFirstPacketReceived(cricket::BaseChannel* channel) {
+ for (auto receiver : receivers_) {
+ receiver->internal()->NotifyFirstPacketReceived();
+ }
+}
+
rtc::scoped_refptr<RtpSenderInterface> RtpTransceiver::sender() const {
RTC_DCHECK(unified_plan_);
RTC_CHECK_EQ(1u, senders_.size());
diff --git a/pc/rtptransceiver.h b/pc/rtptransceiver.h
index 9e8565b..40ab2f0 100644
--- a/pc/rtptransceiver.h
+++ b/pc/rtptransceiver.h
@@ -51,7 +51,8 @@
// AudioRtpSenders, AudioRtpReceivers, and a VoiceChannel. Video RtpTransceivers
// will have VideoRtpSenders, VideoRtpReceivers, and a VideoChannel.
class RtpTransceiver final
- : public rtc::RefCountedObject<RtpTransceiverInterface> {
+ : public rtc::RefCountedObject<RtpTransceiverInterface>,
+ public sigslot::has_slots<> {
public:
// Construct a Plan B-style RtpTransceiver with no senders, receivers, or
// channel set.
@@ -167,6 +168,8 @@
void SetCodecPreferences(rtc::ArrayView<RtpCodecCapability> codecs) override;
private:
+ void OnFirstPacketReceived(cricket::BaseChannel* channel);
+
const bool unified_plan_;
const cricket::MediaType media_type_;
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
