Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
Additionally:
* Moving all implementation inside RemoteAudioTrack into AudioTrack and remove RemoteAudioTrack.
* AddSink/RemoveSink are now on all audio sources (like they are for video sources).
While doing this I found that some of our tests are broken :) and fixed them. They were broken because AudioTrack didn't previously do much such as updating its state.
BUG=chromium:569526
Review URL: https://codereview.webrtc.org/1522903002
Cr-Commit-Position: refs/heads/master@{#11026}
diff --git a/talk/app/webrtc/audiotrack.cc b/talk/app/webrtc/audiotrack.cc
index 63bd87c..b3223cd 100644
--- a/talk/app/webrtc/audiotrack.cc
+++ b/talk/app/webrtc/audiotrack.cc
@@ -27,27 +27,82 @@
#include "talk/app/webrtc/audiotrack.h"
-#include <string>
+#include "webrtc/base/checks.h"
+
+using rtc::scoped_refptr;
namespace webrtc {
const char MediaStreamTrackInterface::kAudioKind[] = "audio";
+// static
+scoped_refptr<AudioTrack> AudioTrack::Create(
+ const std::string& id,
+ const scoped_refptr<AudioSourceInterface>& source) {
+ return new rtc::RefCountedObject<AudioTrack>(id, source);
+}
+
AudioTrack::AudioTrack(const std::string& label,
- AudioSourceInterface* audio_source)
- : MediaStreamTrack<AudioTrackInterface>(label),
- audio_source_(audio_source) {
+ const scoped_refptr<AudioSourceInterface>& source)
+ : MediaStreamTrack<AudioTrackInterface>(label), audio_source_(source) {
+ if (audio_source_) {
+ audio_source_->RegisterObserver(this);
+ OnChanged();
+ }
+}
+
+AudioTrack::~AudioTrack() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ set_state(MediaStreamTrackInterface::kEnded);
+ if (audio_source_)
+ audio_source_->UnregisterObserver(this);
}
std::string AudioTrack::kind() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
return kAudioKind;
}
-rtc::scoped_refptr<AudioTrack> AudioTrack::Create(
- const std::string& id, AudioSourceInterface* source) {
- rtc::RefCountedObject<AudioTrack>* track =
- new rtc::RefCountedObject<AudioTrack>(id, source);
- return track;
+AudioSourceInterface* AudioTrack::GetSource() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ return audio_source_.get();
+}
+
+void AudioTrack::AddSink(AudioTrackSinkInterface* sink) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (audio_source_)
+ audio_source_->AddSink(sink);
+}
+
+void AudioTrack::RemoveSink(AudioTrackSinkInterface* sink) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (audio_source_)
+ audio_source_->RemoveSink(sink);
+}
+
+void AudioTrack::OnChanged() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (state() == kFailed)
+ return; // We can't recover from this state (do we ever set it?).
+
+ TrackState new_state = kInitializing;
+
+ // |audio_source_| must be non-null if we ever get here.
+ switch (audio_source_->state()) {
+ case MediaSourceInterface::kLive:
+ case MediaSourceInterface::kMuted:
+ new_state = kLive;
+ break;
+ case MediaSourceInterface::kEnded:
+ new_state = kEnded;
+ break;
+ case MediaSourceInterface::kInitializing:
+ default:
+ // use kInitializing.
+ break;
+ }
+
+ set_state(new_state);
}
} // namespace webrtc
diff --git a/talk/app/webrtc/audiotrack.h b/talk/app/webrtc/audiotrack.h
index 750f272..55f4837 100644
--- a/talk/app/webrtc/audiotrack.h
+++ b/talk/app/webrtc/audiotrack.h
@@ -28,40 +28,47 @@
#ifndef TALK_APP_WEBRTC_AUDIOTRACK_H_
#define TALK_APP_WEBRTC_AUDIOTRACK_H_
+#include <string>
+
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/mediastreamtrack.h"
#include "talk/app/webrtc/notifier.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/thread_checker.h"
namespace webrtc {
-class AudioTrack : public MediaStreamTrack<AudioTrackInterface> {
+class AudioTrack : public MediaStreamTrack<AudioTrackInterface>,
+ public ObserverInterface {
+ protected:
+ // Protected ctor to force use of factory method.
+ AudioTrack(const std::string& label,
+ const rtc::scoped_refptr<AudioSourceInterface>& source);
+ ~AudioTrack() override;
+
public:
static rtc::scoped_refptr<AudioTrack> Create(
- const std::string& id, AudioSourceInterface* source);
+ const std::string& id,
+ const rtc::scoped_refptr<AudioSourceInterface>& source);
- // AudioTrackInterface implementation.
- AudioSourceInterface* GetSource() const override {
- return audio_source_.get();
- }
- // TODO(xians): Implement these methods.
- void AddSink(AudioTrackSinkInterface* sink) override {}
- void RemoveSink(AudioTrackSinkInterface* sink) override {}
- bool GetSignalLevel(int* level) override { return false; }
- rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() override {
- return NULL;
- }
- cricket::AudioRenderer* GetRenderer() override { return NULL; }
-
+ private:
// MediaStreamTrack implementation.
std::string kind() const override;
- protected:
- AudioTrack(const std::string& label, AudioSourceInterface* audio_source);
+ // AudioTrackInterface implementation.
+ AudioSourceInterface* GetSource() const override;
+
+ void AddSink(AudioTrackSinkInterface* sink) override;
+ void RemoveSink(AudioTrackSinkInterface* sink) override;
+
+ // ObserverInterface implementation.
+ void OnChanged() override;
private:
- rtc::scoped_refptr<AudioSourceInterface> audio_source_;
+ const rtc::scoped_refptr<AudioSourceInterface> audio_source_;
+ rtc::ThreadChecker thread_checker_;
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTrack);
};
} // namespace webrtc
diff --git a/talk/app/webrtc/localaudiosource.h b/talk/app/webrtc/localaudiosource.h
index 557745b..5158eb1 100644
--- a/talk/app/webrtc/localaudiosource.h
+++ b/talk/app/webrtc/localaudiosource.h
@@ -48,16 +48,17 @@
const PeerConnectionFactoryInterface::Options& options,
const MediaConstraintsInterface* constraints);
- virtual SourceState state() const { return source_state_; }
+ SourceState state() const override { return source_state_; }
+ bool remote() const override { return false; }
+
virtual const cricket::AudioOptions& options() const { return options_; }
- protected:
- LocalAudioSource()
- : source_state_(kInitializing) {
- }
+ void AddSink(AudioTrackSinkInterface* sink) override {}
+ void RemoveSink(AudioTrackSinkInterface* sink) override {}
- ~LocalAudioSource() {
- }
+ protected:
+ LocalAudioSource() : source_state_(kInitializing) {}
+ ~LocalAudioSource() override {}
private:
void Initialize(const PeerConnectionFactoryInterface::Options& options,
diff --git a/talk/app/webrtc/mediastream_unittest.cc b/talk/app/webrtc/mediastream_unittest.cc
index 2cf930c..f19b945 100644
--- a/talk/app/webrtc/mediastream_unittest.cc
+++ b/talk/app/webrtc/mediastream_unittest.cc
@@ -48,9 +48,23 @@
// Helper class to test Observer.
class MockObserver : public ObserverInterface {
public:
- MockObserver() {}
+ explicit MockObserver(NotifierInterface* notifier) : notifier_(notifier) {
+ notifier_->RegisterObserver(this);
+ }
+
+ ~MockObserver() { Unregister(); }
+
+ void Unregister() {
+ if (notifier_) {
+ notifier_->UnregisterObserver(this);
+ notifier_ = nullptr;
+ }
+ }
MOCK_METHOD0(OnChanged, void());
+
+ private:
+ NotifierInterface* notifier_;
};
class MediaStreamTest: public testing::Test {
@@ -75,8 +89,7 @@
}
void ChangeTrack(MediaStreamTrackInterface* track) {
- MockObserver observer;
- track->RegisterObserver(&observer);
+ MockObserver observer(track);
EXPECT_CALL(observer, OnChanged())
.Times(Exactly(1));
@@ -127,8 +140,7 @@
}
TEST_F(MediaStreamTest, RemoveTrack) {
- MockObserver observer;
- stream_->RegisterObserver(&observer);
+ MockObserver observer(stream_);
EXPECT_CALL(observer, OnChanged())
.Times(Exactly(2));
diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h
index 5327bd2..a1c0c80 100644
--- a/talk/app/webrtc/mediastreaminterface.h
+++ b/talk/app/webrtc/mediastreaminterface.h
@@ -71,8 +71,6 @@
// Base class for sources. A MediaStreamTrack have an underlying source that
// provide media. A source can be shared with multiple tracks.
-// TODO(perkj): Implement sources for local and remote audio tracks and
-// remote video tracks.
class MediaSourceInterface : public rtc::RefCountInterface,
public NotifierInterface {
public:
@@ -85,6 +83,8 @@
virtual SourceState state() const = 0;
+ virtual bool remote() const = 0;
+
protected:
virtual ~MediaSourceInterface() {}
};
@@ -152,6 +152,19 @@
virtual ~VideoTrackInterface() {}
};
+// Interface for receiving audio data from a AudioTrack.
+class AudioTrackSinkInterface {
+ public:
+ virtual void OnData(const void* audio_data,
+ int bits_per_sample,
+ int sample_rate,
+ int number_of_channels,
+ size_t number_of_frames) = 0;
+
+ protected:
+ virtual ~AudioTrackSinkInterface() {}
+};
+
// AudioSourceInterface is a reference counted source used for AudioTracks.
// The same source can be used in multiple AudioTracks.
class AudioSourceInterface : public MediaSourceInterface {
@@ -174,18 +187,10 @@
// Registers/unregisters observer to the audio source.
virtual void RegisterAudioObserver(AudioObserver* observer) {}
virtual void UnregisterAudioObserver(AudioObserver* observer) {}
-};
-// Interface for receiving audio data from a AudioTrack.
-class AudioTrackSinkInterface {
- public:
- virtual void OnData(const void* audio_data,
- int bits_per_sample,
- int sample_rate,
- int number_of_channels,
- size_t number_of_frames) = 0;
- protected:
- virtual ~AudioTrackSinkInterface() {}
+ // TODO(tommi): Make pure virtual.
+ virtual void AddSink(AudioTrackSinkInterface* sink) {}
+ virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
};
// Interface of the audio processor used by the audio track to collect
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index 83380e1..6cca166 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -39,7 +39,6 @@
#include "talk/app/webrtc/mediastreamproxy.h"
#include "talk/app/webrtc/mediastreamtrackproxy.h"
#include "talk/app/webrtc/remoteaudiosource.h"
-#include "talk/app/webrtc/remoteaudiotrack.h"
#include "talk/app/webrtc/remotevideocapturer.h"
#include "talk/app/webrtc/rtpreceiver.h"
#include "talk/app/webrtc/rtpsender.h"
@@ -454,7 +453,7 @@
AudioProviderInterface* provider,
webrtc::MediaStreamInterface* stream,
const std::string& track_id) {
- return AddTrack<AudioTrackInterface, RemoteAudioTrack, AudioTrackProxy>(
+ return AddTrack<AudioTrackInterface, AudioTrack, AudioTrackProxy>(
stream, track_id, RemoteAudioSource::Create(ssrc, provider));
}
@@ -463,7 +462,7 @@
return AddTrack<VideoTrackInterface, VideoTrack, VideoTrackProxy>(
stream, track_id,
VideoSource::Create(channel_manager_, new RemoteVideoCapturer(),
- nullptr)
+ nullptr, true)
.get());
}
diff --git a/talk/app/webrtc/peerconnectionfactory.cc b/talk/app/webrtc/peerconnectionfactory.cc
index 6723550..d01f542 100644
--- a/talk/app/webrtc/peerconnectionfactory.cc
+++ b/talk/app/webrtc/peerconnectionfactory.cc
@@ -225,8 +225,8 @@
cricket::VideoCapturer* capturer,
const MediaConstraintsInterface* constraints) {
RTC_DCHECK(signaling_thread_->IsCurrent());
- rtc::scoped_refptr<VideoSource> source(
- VideoSource::Create(channel_manager_.get(), capturer, constraints));
+ rtc::scoped_refptr<VideoSource> source(VideoSource::Create(
+ channel_manager_.get(), capturer, constraints, false));
return VideoSourceProxy::Create(signaling_thread_, source);
}
@@ -339,8 +339,7 @@
PeerConnectionFactory::CreateAudioTrack(const std::string& id,
AudioSourceInterface* source) {
RTC_DCHECK(signaling_thread_->IsCurrent());
- rtc::scoped_refptr<AudioTrackInterface> track(
- AudioTrack::Create(id, source));
+ rtc::scoped_refptr<AudioTrackInterface> track(AudioTrack::Create(id, source));
return AudioTrackProxy::Create(signaling_thread_, track);
}
diff --git a/talk/app/webrtc/remoteaudiosource.cc b/talk/app/webrtc/remoteaudiosource.cc
index a35f0bb..e904dd9 100644
--- a/talk/app/webrtc/remoteaudiosource.cc
+++ b/talk/app/webrtc/remoteaudiosource.cc
@@ -106,6 +106,11 @@
return state_;
}
+bool RemoteAudioSource::remote() const {
+ RTC_DCHECK(main_thread_->IsCurrent());
+ return true;
+}
+
void RemoteAudioSource::SetVolume(double volume) {
RTC_DCHECK(volume >= 0 && volume <= 10);
for (auto* observer : audio_observers_)
diff --git a/talk/app/webrtc/remoteaudiosource.h b/talk/app/webrtc/remoteaudiosource.h
index f518d9b..d648ba4 100644
--- a/talk/app/webrtc/remoteaudiosource.h
+++ b/talk/app/webrtc/remoteaudiosource.h
@@ -56,9 +56,10 @@
// MediaSourceInterface implementation.
MediaSourceInterface::SourceState state() const override;
+ bool remote() const override;
- void AddSink(AudioTrackSinkInterface* sink);
- void RemoveSink(AudioTrackSinkInterface* sink);
+ void AddSink(AudioTrackSinkInterface* sink) override;
+ void RemoveSink(AudioTrackSinkInterface* sink) override;
protected:
RemoteAudioSource();
diff --git a/talk/app/webrtc/remoteaudiotrack.cc b/talk/app/webrtc/remoteaudiotrack.cc
index 2c4481c..5f0b23e 100644
--- a/talk/app/webrtc/remoteaudiotrack.cc
+++ b/talk/app/webrtc/remoteaudiotrack.cc
@@ -25,71 +25,4 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/app/webrtc/remoteaudiotrack.h"
-
-#include "talk/app/webrtc/remoteaudiosource.h"
-
-using rtc::scoped_refptr;
-
-namespace webrtc {
-
-// static
-scoped_refptr<RemoteAudioTrack> RemoteAudioTrack::Create(
- const std::string& id,
- const scoped_refptr<RemoteAudioSource>& source) {
- return new rtc::RefCountedObject<RemoteAudioTrack>(id, source);
-}
-
-RemoteAudioTrack::RemoteAudioTrack(
- const std::string& label,
- const scoped_refptr<RemoteAudioSource>& source)
- : MediaStreamTrack<AudioTrackInterface>(label), audio_source_(source) {
- audio_source_->RegisterObserver(this);
- TrackState new_state = kInitializing;
- switch (audio_source_->state()) {
- case MediaSourceInterface::kLive:
- case MediaSourceInterface::kMuted:
- new_state = kLive;
- break;
- case MediaSourceInterface::kEnded:
- new_state = kEnded;
- break;
- case MediaSourceInterface::kInitializing:
- default:
- // kInitializing;
- break;
- }
- set_state(new_state);
-}
-
-RemoteAudioTrack::~RemoteAudioTrack() {
- set_state(MediaStreamTrackInterface::kEnded);
- audio_source_->UnregisterObserver(this);
-}
-
-std::string RemoteAudioTrack::kind() const {
- return MediaStreamTrackInterface::kAudioKind;
-}
-
-AudioSourceInterface* RemoteAudioTrack::GetSource() const {
- return audio_source_.get();
-}
-
-void RemoteAudioTrack::AddSink(AudioTrackSinkInterface* sink) {
- audio_source_->AddSink(sink);
-}
-
-void RemoteAudioTrack::RemoveSink(AudioTrackSinkInterface* sink) {
- audio_source_->RemoveSink(sink);
-}
-
-bool RemoteAudioTrack::GetSignalLevel(int* level) {
- return false;
-}
-
-void RemoteAudioTrack::OnChanged() {
- if (audio_source_->state() == MediaSourceInterface::kEnded)
- set_state(MediaStreamTrackInterface::kEnded);
-}
-
-} // namespace webrtc
+// TODO(tommi): Delete this file when removed from build files in Chromium.
diff --git a/talk/app/webrtc/remoteaudiotrack.h b/talk/app/webrtc/remoteaudiotrack.h
index c9240eb..5f0b23e 100644
--- a/talk/app/webrtc/remoteaudiotrack.h
+++ b/talk/app/webrtc/remoteaudiotrack.h
@@ -25,53 +25,4 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_APP_WEBRTC_REMOTEAUDIOTRACK_H_
-#define TALK_APP_WEBRTC_REMOTEAUDIOTRACK_H_
-
-#include <string>
-
-#include "talk/app/webrtc/mediastreaminterface.h"
-#include "talk/app/webrtc/mediastreamtrack.h"
-#include "talk/app/webrtc/notifier.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/scoped_ref_ptr.h"
-
-namespace webrtc {
-
-class RemoteAudioSource;
-
-class RemoteAudioTrack : public MediaStreamTrack<AudioTrackInterface>,
- public ObserverInterface {
- protected:
- // Protected ctor to force use of factory method.
- RemoteAudioTrack(const std::string& label,
- const rtc::scoped_refptr<RemoteAudioSource>& source);
- ~RemoteAudioTrack() override;
-
- public:
- static rtc::scoped_refptr<RemoteAudioTrack> Create(
- const std::string& id,
- const rtc::scoped_refptr<RemoteAudioSource>& source);
-
- private:
- // MediaStreamTrack implementation.
- std::string kind() const override;
-
- // AudioTrackInterface implementation.
- AudioSourceInterface* GetSource() const override;
-
- void AddSink(AudioTrackSinkInterface* sink) override;
- void RemoveSink(AudioTrackSinkInterface* sink) override;
- bool GetSignalLevel(int* level) override;
-
- // ObserverInterface implementation.
- void OnChanged() override;
-
- private:
- const rtc::scoped_refptr<RemoteAudioSource> audio_source_;
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RemoteAudioTrack);
-};
-
-} // namespace webrtc
-
-#endif // TALK_APP_WEBRTC_REMOTEAUDIOTRACK_H_
+// TODO(tommi): Delete this file when removed from build files in Chromium.
diff --git a/talk/app/webrtc/rtpreceiver.cc b/talk/app/webrtc/rtpreceiver.cc
index b88554f..9540f36 100644
--- a/talk/app/webrtc/rtpreceiver.cc
+++ b/talk/app/webrtc/rtpreceiver.cc
@@ -39,6 +39,7 @@
ssrc_(ssrc),
provider_(provider),
cached_track_enabled_(track->enabled()) {
+ RTC_DCHECK(track_->GetSource()->remote());
track_->RegisterObserver(this);
track_->GetSource()->RegisterAudioObserver(this);
Reconfigure();
@@ -85,6 +86,7 @@
uint32_t ssrc,
VideoProviderInterface* provider)
: id_(track->id()), track_(track), ssrc_(ssrc), provider_(provider) {
+ RTC_DCHECK(track_->GetSource()->remote());
provider_->SetVideoPlayout(ssrc_, true, track_->GetSource()->FrameInput());
}
diff --git a/talk/app/webrtc/rtpsender.cc b/talk/app/webrtc/rtpsender.cc
index ea10b7b..c0d23a0 100644
--- a/talk/app/webrtc/rtpsender.cc
+++ b/talk/app/webrtc/rtpsender.cc
@@ -184,7 +184,8 @@
void AudioRtpSender::SetAudioSend() {
RTC_DCHECK(!stopped_ && can_send_track());
cricket::AudioOptions options;
- if (track_->enabled() && track_->GetSource()) {
+ if (track_->enabled() && track_->GetSource() &&
+ !track_->GetSource()->remote()) {
// TODO(xians): Remove this static_cast since we should be able to connect
// a remote audio track to a peer connection.
options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
diff --git a/talk/app/webrtc/rtpsenderreceiver_unittest.cc b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
index 3f61504..a590e1d 100644
--- a/talk/app/webrtc/rtpsenderreceiver_unittest.cc
+++ b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
@@ -31,7 +31,6 @@
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/mediastream.h"
#include "talk/app/webrtc/remoteaudiosource.h"
-#include "talk/app/webrtc/remoteaudiotrack.h"
#include "talk/app/webrtc/rtpreceiver.h"
#include "talk/app/webrtc/rtpsender.h"
#include "talk/app/webrtc/streamcollection.h"
@@ -98,8 +97,8 @@
class FakeVideoSource : public Notifier<VideoSourceInterface> {
public:
- static rtc::scoped_refptr<FakeVideoSource> Create() {
- return new rtc::RefCountedObject<FakeVideoSource>();
+ static rtc::scoped_refptr<FakeVideoSource> Create(bool remote) {
+ return new rtc::RefCountedObject<FakeVideoSource>(remote);
}
virtual cricket::VideoCapturer* GetVideoCapturer() { return &fake_capturer_; }
virtual void Stop() {}
@@ -107,16 +106,18 @@
virtual void AddSink(cricket::VideoRenderer* output) {}
virtual void RemoveSink(cricket::VideoRenderer* output) {}
virtual SourceState state() const { return state_; }
+ virtual bool remote() const { return remote_; }
virtual const cricket::VideoOptions* options() const { return &options_; }
virtual cricket::VideoRenderer* FrameInput() { return NULL; }
protected:
- FakeVideoSource() : state_(kLive) {}
+ explicit FakeVideoSource(bool remote) : state_(kLive), remote_(remote) {}
~FakeVideoSource() {}
private:
cricket::FakeVideoCapturer fake_capturer_;
SourceState state_;
+ bool remote_;
cricket::VideoOptions options_;
};
@@ -124,7 +125,11 @@
public:
virtual void SetUp() {
stream_ = MediaStream::Create(kStreamLabel1);
- rtc::scoped_refptr<VideoSourceInterface> source(FakeVideoSource::Create());
+ }
+
+ void AddVideoTrack(bool remote) {
+ rtc::scoped_refptr<VideoSourceInterface> source(
+ FakeVideoSource::Create(remote));
video_track_ = VideoTrack::Create(kVideoTrackId, source);
EXPECT_TRUE(stream_->AddTrack(video_track_));
}
@@ -140,6 +145,7 @@
}
void CreateVideoRtpSender() {
+ AddVideoTrack(false);
EXPECT_CALL(video_provider_,
SetCaptureDevice(
kVideoSsrc, video_track_->GetSource()->GetVideoCapturer()));
@@ -162,7 +168,7 @@
}
void CreateAudioRtpReceiver() {
- audio_track_ = RemoteAudioTrack::Create(
+ audio_track_ = AudioTrack::Create(
kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, NULL));
EXPECT_TRUE(stream_->AddTrack(audio_track_));
EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true));
@@ -171,6 +177,7 @@
}
void CreateVideoRtpReceiver() {
+ AddVideoTrack(true);
EXPECT_CALL(video_provider_,
SetVideoPlayout(kVideoSsrc, true,
video_track_->GetSource()->FrameInput()));
@@ -355,6 +362,7 @@
// Test that a video sender calls the expected methods on the provider once
// it has a track and SSRC, when the SSRC is set first.
TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack) {
+ AddVideoTrack(false);
rtc::scoped_refptr<VideoRtpSender> sender =
new VideoRtpSender(&video_provider_);
sender->SetSsrc(kVideoSsrc);
@@ -372,6 +380,7 @@
// Test that a video sender calls the expected methods on the provider once
// it has a track and SSRC, when the SSRC is set last.
TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupTrackThenSsrc) {
+ AddVideoTrack(false);
rtc::scoped_refptr<VideoRtpSender> sender =
new VideoRtpSender(&video_provider_);
sender->SetTrack(video_track_);
@@ -407,6 +416,7 @@
// Test that the sender is disconnected from the provider when its SSRC is
// set to 0.
TEST_F(RtpSenderReceiverTest, VideoSenderSsrcSetToZero) {
+ AddVideoTrack(false);
EXPECT_CALL(video_provider_,
SetCaptureDevice(kVideoSsrc,
video_track_->GetSource()->GetVideoCapturer()));
@@ -442,6 +452,7 @@
}
TEST_F(RtpSenderReceiverTest, VideoSenderTrackSetToNull) {
+ AddVideoTrack(false);
EXPECT_CALL(video_provider_,
SetCaptureDevice(kVideoSsrc,
video_track_->GetSource()->GetVideoCapturer()));
@@ -461,6 +472,7 @@
}
TEST_F(RtpSenderReceiverTest, AudioSenderSsrcChanged) {
+ AddVideoTrack(false);
rtc::scoped_refptr<AudioTrackInterface> track =
AudioTrack::Create(kAudioTrackId, nullptr);
EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
@@ -477,6 +489,7 @@
}
TEST_F(RtpSenderReceiverTest, VideoSenderSsrcChanged) {
+ AddVideoTrack(false);
EXPECT_CALL(video_provider_,
SetCaptureDevice(kVideoSsrc,
video_track_->GetSource()->GetVideoCapturer()));
diff --git a/talk/app/webrtc/videosource.cc b/talk/app/webrtc/videosource.cc
index 5cef7be..5bd735a 100644
--- a/talk/app/webrtc/videosource.cc
+++ b/talk/app/webrtc/videosource.cc
@@ -331,21 +331,23 @@
rtc::scoped_refptr<VideoSource> VideoSource::Create(
cricket::ChannelManager* channel_manager,
cricket::VideoCapturer* capturer,
- const webrtc::MediaConstraintsInterface* constraints) {
+ const webrtc::MediaConstraintsInterface* constraints,
+ bool remote) {
ASSERT(channel_manager != NULL);
ASSERT(capturer != NULL);
- rtc::scoped_refptr<VideoSource> source(
- new rtc::RefCountedObject<VideoSource>(channel_manager,
- capturer));
+ rtc::scoped_refptr<VideoSource> source(new rtc::RefCountedObject<VideoSource>(
+ channel_manager, capturer, remote));
source->Initialize(constraints);
return source;
}
VideoSource::VideoSource(cricket::ChannelManager* channel_manager,
- cricket::VideoCapturer* capturer)
+ cricket::VideoCapturer* capturer,
+ bool remote)
: channel_manager_(channel_manager),
video_capturer_(capturer),
- state_(kInitializing) {
+ state_(kInitializing),
+ remote_(remote) {
channel_manager_->SignalVideoCaptureStateChange.connect(
this, &VideoSource::OnStateChange);
}
diff --git a/talk/app/webrtc/videosource.h b/talk/app/webrtc/videosource.h
index 8253cba..98c1e08 100644
--- a/talk/app/webrtc/videosource.h
+++ b/talk/app/webrtc/videosource.h
@@ -66,9 +66,12 @@
static rtc::scoped_refptr<VideoSource> Create(
cricket::ChannelManager* channel_manager,
cricket::VideoCapturer* capturer,
- const webrtc::MediaConstraintsInterface* constraints);
+ const webrtc::MediaConstraintsInterface* constraints,
+ bool remote);
- virtual SourceState state() const { return state_; }
+ SourceState state() const override { return state_; }
+ bool remote() const override { return remote_; }
+
virtual const cricket::VideoOptions* options() const { return &options_; }
virtual cricket::VideoRenderer* FrameInput();
@@ -86,7 +89,8 @@
protected:
VideoSource(cricket::ChannelManager* channel_manager,
- cricket::VideoCapturer* capturer);
+ cricket::VideoCapturer* capturer,
+ bool remote);
virtual ~VideoSource();
void Initialize(const webrtc::MediaConstraintsInterface* constraints);
@@ -104,6 +108,7 @@
cricket::VideoFormat format_;
cricket::VideoOptions options_;
SourceState state_;
+ const bool remote_;
};
} // namespace webrtc
diff --git a/talk/app/webrtc/videosource_unittest.cc b/talk/app/webrtc/videosource_unittest.cc
index 8d6d6ac..6f1df34 100644
--- a/talk/app/webrtc/videosource_unittest.cc
+++ b/talk/app/webrtc/videosource_unittest.cc
@@ -144,9 +144,9 @@
void CreateVideoSource(
const webrtc::MediaConstraintsInterface* constraints) {
// VideoSource take ownership of |capturer_|
- source_ = VideoSource::Create(channel_manager_.get(),
- capturer_cleanup_.release(),
- constraints);
+ source_ =
+ VideoSource::Create(channel_manager_.get(), capturer_cleanup_.release(),
+ constraints, false);
ASSERT_TRUE(source_.get() != NULL);
EXPECT_EQ(capturer_, source_->GetVideoCapturer());
@@ -210,8 +210,7 @@
// RemoteVideoCapturer and takes video frames from FrameInput.
TEST_F(VideoSourceTest, StartStopRemote) {
source_ = VideoSource::Create(channel_manager_.get(),
- new webrtc::RemoteVideoCapturer(),
- NULL);
+ new webrtc::RemoteVideoCapturer(), NULL, true);
ASSERT_TRUE(source_.get() != NULL);
EXPECT_TRUE(NULL != source_->GetVideoCapturer());
diff --git a/talk/app/webrtc/videosourceproxy.h b/talk/app/webrtc/videosourceproxy.h
index 677fa9c..ce96e8e 100644
--- a/talk/app/webrtc/videosourceproxy.h
+++ b/talk/app/webrtc/videosourceproxy.h
@@ -38,6 +38,7 @@
// signaling thread.
BEGIN_PROXY_MAP(VideoSource)
PROXY_CONSTMETHOD0(SourceState, state)
+ PROXY_CONSTMETHOD0(bool, remote)
PROXY_METHOD0(cricket::VideoCapturer*, GetVideoCapturer)
PROXY_METHOD0(void, Stop)
PROXY_METHOD0(void, Restart)
diff --git a/talk/app/webrtc/videotrack_unittest.cc b/talk/app/webrtc/videotrack_unittest.cc
index 609ee80..013d925 100644
--- a/talk/app/webrtc/videotrack_unittest.cc
+++ b/talk/app/webrtc/videotrack_unittest.cc
@@ -62,7 +62,7 @@
video_track_ = VideoTrack::Create(
kVideoTrackId,
VideoSource::Create(channel_manager_.get(),
- new webrtc::RemoteVideoCapturer(), NULL));
+ new webrtc::RemoteVideoCapturer(), NULL, true));
}
protected:
diff --git a/talk/libjingle.gyp b/talk/libjingle.gyp
index 4ad1e39..e8035fe 100755
--- a/talk/libjingle.gyp
+++ b/talk/libjingle.gyp
@@ -763,8 +763,6 @@
'app/webrtc/proxy.h',
'app/webrtc/remoteaudiosource.cc',
'app/webrtc/remoteaudiosource.h',
- 'app/webrtc/remoteaudiotrack.cc',
- 'app/webrtc/remoteaudiotrack.h',
'app/webrtc/remotevideocapturer.cc',
'app/webrtc/remotevideocapturer.h',
'app/webrtc/rtpreceiver.cc',
