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Unified Diff: content/renderer/media/webrtc_local_audio_track_unittest.cc

Issue 1780653002: Revert of MediaStream audio object graph untangling and clean-ups. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 9 months ago
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Index: content/renderer/media/webrtc_local_audio_track_unittest.cc
diff --git a/content/renderer/media/webrtc_local_audio_track_unittest.cc b/content/renderer/media/webrtc_local_audio_track_unittest.cc
index 1e9868c802f12e967341545901614f24b883af7a..7c478c81031c279e79ec7472cacc605b619525cf 100644
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc
+++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc
@@ -95,7 +95,7 @@
int session_id));
MOCK_METHOD0(OnStart, void());
MOCK_METHOD0(OnStop, void());
- void SetVolume(double volume) final {}
+ MOCK_METHOD1(SetVolume, void(double volume));
MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));
void Initialize(const media::AudioParameters& params,
@@ -165,16 +165,12 @@
StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
std::string(), std::string());
- {
- scoped_ptr<WebRtcAudioCapturer> capturer =
- WebRtcAudioCapturer::CreateCapturer(
- -1, device, constraint_factory.CreateWebMediaConstraints(),
- nullptr, audio_source);
- capturer_ = capturer.get();
- audio_source->SetAudioCapturer(std::move(capturer));
- }
- capturer_source_ = new MockCapturerSource(capturer_);
- EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_, -1))
+ capturer_ = WebRtcAudioCapturer::CreateCapturer(
+ -1, device, constraint_factory.CreateWebMediaConstraints(), NULL,
+ audio_source);
+ audio_source->SetAudioCapturer(capturer_.get());
+ capturer_source_ = new MockCapturerSource(capturer_.get());
+ EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1))
.WillOnce(Return());
EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
EXPECT_CALL(*capturer_source_.get(), OnStart());
@@ -188,8 +184,8 @@
media::AudioParameters params_;
blink::WebMediaStreamSource blink_source_;
- WebRtcAudioCapturer* capturer_; // Owned by |blink_source_|.
scoped_refptr<MockCapturerSource> capturer_source_;
+ scoped_refptr<WebRtcAudioCapturer> capturer_;
};
// Creates a capturer and audio track, fakes its audio thread, and
@@ -200,12 +196,8 @@
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track(
- new WebRtcLocalAudioTrack(adapter.get()));
- track->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track.get()));
- capturer_->AddTrack(track.get());
+ new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
+ track->Start();
EXPECT_TRUE(track->GetAudioAdapter()->enabled());
scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
@@ -234,16 +226,12 @@
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track(
- new WebRtcLocalAudioTrack(adapter.get()));
- track->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track.get()));
- capturer_->AddTrack(track.get());
+ new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
+ track->Start();
EXPECT_TRUE(track->GetAudioAdapter()->enabled());
EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false));
scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
- const media::AudioParameters params = capturer_->GetInputFormat();
+ const media::AudioParameters params = capturer_->source_audio_parameters();
base::WaitableEvent event(false, false);
EXPECT_CALL(*sink, FormatIsSet()).Times(1);
EXPECT_CALL(*sink, CaptureData()).Times(0);
@@ -271,15 +259,11 @@
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track_1(
- new WebRtcLocalAudioTrack(adapter_1.get()));
- track_1->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track_1.get()));
- capturer_->AddTrack(track_1.get());
+ new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
+ track_1->Start();
EXPECT_TRUE(track_1->GetAudioAdapter()->enabled());
scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
- const media::AudioParameters params = capturer_->GetInputFormat();
+ const media::AudioParameters params = capturer_->source_audio_parameters();
base::WaitableEvent event_1(false, false);
EXPECT_CALL(*sink_1, FormatIsSet()).WillOnce(Return());
EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1))
@@ -292,12 +276,8 @@
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track_2(
- new WebRtcLocalAudioTrack(adapter_2.get()));
- track_2->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track_2.get()));
- capturer_->AddTrack(track_2.get());
+ new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL));
+ track_2->Start();
EXPECT_TRUE(track_2->GetAudioAdapter()->enabled());
// Verify both |sink_1| and |sink_2| get data.
@@ -335,12 +315,8 @@
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track(
- new WebRtcLocalAudioTrack(adapter.get()));
- track->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track.get()));
- capturer_->AddTrack(track.get());
+ new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
+ track->Start();
// When the track goes away, it will automatically stop the
// |capturer_source_|.
@@ -355,22 +331,14 @@
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter1(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track1(
- new WebRtcLocalAudioTrack(adapter1.get()));
- track1->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track1.get()));
- capturer_->AddTrack(track1.get());
+ new WebRtcLocalAudioTrack(adapter1.get(), capturer_, NULL));
+ track1->Start();
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track2(
- new WebRtcLocalAudioTrack(adapter2.get()));
- track2->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track2.get()));
- capturer_->AddTrack(track2.get());
+ new WebRtcLocalAudioTrack(adapter2.get(), capturer_, NULL));
+ track2->Start();
track1->Stop();
// When the last track is stopped, it will automatically stop the
@@ -386,12 +354,8 @@
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track_1(
- new WebRtcLocalAudioTrack(adapter_1.get()));
- track_1->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track_1.get()));
- capturer_->AddTrack(track_1.get());
+ new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
+ track_1->Start();
// Verify the data flow by connecting the sink to |track_1|.
scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
@@ -408,12 +372,8 @@
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track_2(
- new WebRtcLocalAudioTrack(adapter_2.get()));
- track_2->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track_2.get()));
- capturer_->AddTrack(track_2.get());
+ new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL));
+ track_2->Start();
// Stop the capturer will clear up the track lists in the capturer.
EXPECT_CALL(*capturer_source_.get(), OnStop());
@@ -444,12 +404,8 @@
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track_1(
- new WebRtcLocalAudioTrack(adapter_1.get()));
- track_1->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track_1.get()));
- capturer_->AddTrack(track_1.get());
+ new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
+ track_1->Start();
// Verify the data flow by connecting the |sink_1| to |track_1|.
scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
@@ -462,7 +418,7 @@
MockConstraintFactory constraint_factory;
StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
std::string(), std::string());
- scoped_ptr<WebRtcAudioCapturer> new_capturer(
+ scoped_refptr<WebRtcAudioCapturer> new_capturer(
WebRtcAudioCapturer::CreateCapturer(
-1, device, constraint_factory.CreateWebMediaConstraints(), NULL,
NULL));
@@ -481,12 +437,8 @@
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track_2(
- new WebRtcLocalAudioTrack(adapter_2.get()));
- track_2->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track_2.get()));
- new_capturer->AddTrack(track_2.get());
+ new WebRtcLocalAudioTrack(adapter_2.get(), new_capturer, NULL));
+ track_2->Start();
// Verify the data flow by connecting the |sink_2| to |track_2|.
scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
@@ -519,11 +471,12 @@
// Create a capturer with new source which works with the format above.
MockConstraintFactory factory;
factory.DisableDefaultAudioConstraints();
- scoped_ptr<WebRtcAudioCapturer> capturer(WebRtcAudioCapturer::CreateCapturer(
- -1,
- StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "", params.sample_rate(),
- params.channel_layout(), params.frames_per_buffer()),
- factory.CreateWebMediaConstraints(), NULL, NULL));
+ scoped_refptr<WebRtcAudioCapturer> capturer(
+ WebRtcAudioCapturer::CreateCapturer(
+ -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "",
+ params.sample_rate(), params.channel_layout(),
+ params.frames_per_buffer()),
+ factory.CreateWebMediaConstraints(), NULL, NULL));
scoped_refptr<MockCapturerSource> source(
new MockCapturerSource(capturer.get()));
EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1));
@@ -535,12 +488,8 @@
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track(
- new WebRtcLocalAudioTrack(adapter.get()));
- track->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track.get()));
- capturer->AddTrack(track.get());
+ new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
+ track->Start();
// Verify the data flow by connecting the |sink| to |track|.
scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
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