#include "base/synchronization/waitable_event.h"
#include "base/test/test_timeouts.h"
+#include "content/renderer/media/media_stream_audio_source.h"
#include "content/renderer/media/mock_media_constraint_factory.h"
#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
#include "content/renderer/media/webrtc_audio_capturer.h"
// the |WebRtcAudioCapturer|.
class FakeAudioThread : public base::PlatformThread::Delegate {
public:
- FakeAudioThread(const scoped_refptr<WebRtcAudioCapturer>& capturer,
+ FakeAudioThread(WebRtcAudioCapturer* capturer,
const media::AudioParameters& params)
: capturer_(capturer),
thread_(),
closure_(false, false) {
- DCHECK(capturer.get());
+ DCHECK(capturer);
audio_bus_ = media::AudioBus::Create(params);
}
media::AudioCapturerSource::CaptureCallback* callback =
static_cast<media::AudioCapturerSource::CaptureCallback*>(
- capturer_.get());
+ capturer_);
audio_bus_->Zero();
callback->Capture(audio_bus_.get(), 0, 0, false);
private:
scoped_ptr<media::AudioBus> audio_bus_;
- scoped_refptr<WebRtcAudioCapturer> capturer_;
+ WebRtcAudioCapturer* capturer_;
base::PlatformThreadHandle thread_;
base::WaitableEvent closure_;
DISALLOW_COPY_AND_ASSIGN(FakeAudioThread);
params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
media::CHANNEL_LAYOUT_STEREO, 2, 0, 48000, 16, 480);
blink::WebMediaConstraints constraints;
+ blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio,
+ "dummy");
+ MediaStreamAudioSource* audio_source = new MediaStreamAudioSource();
+ blink_source_.setExtraData(audio_source);
+
StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
std::string(), std::string());
capturer_ = WebRtcAudioCapturer::CreateCapturer(-1, device,
- constraints, NULL);
+ constraints, NULL,
+ audio_source);
+ audio_source->SetAudioCapturer(capturer_);
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());
capturer_->SetCapturerSourceForTesting(capturer_source_, params_);
}
media::AudioParameters params_;
+ blink::WebMediaStreamSource blink_source_;
scoped_refptr<MockCapturerSource> capturer_source_;
scoped_refptr<WebRtcAudioCapturer> capturer_;
};
// get data callback when the track is connected to the capturer but not when
// the track is disconnected from the capturer.
TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
- EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
- EXPECT_CALL(*capturer_source_.get(), OnStart());
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track(
// callbacks appear/disappear.
// Flaky due to a data race, see http://crbug.com/295418
TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
- EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
- EXPECT_CALL(*capturer_source_.get(), OnStart());
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track_1(
track_2->RemoveSink(sink_2.get());
track_2->Stop();
track_2.reset();
-
- capturer_->Stop();
}
// Start one track and verify the capturer is correctly starting its source.
// And it should be fine to not to call Stop() explicitly.
TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
- EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
- EXPECT_CALL(*capturer_source_.get(), OnStart());
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track(
// When the track goes away, it will automatically stop the
// |capturer_source_|.
EXPECT_CALL(*capturer_source_.get(), OnStop());
- capturer_->Stop();
track.reset();
}
+// Start two tracks and verify the capturer is correctly starting its source.
+// When the last track connected to the capturer is stopped, the source is
+// stopped.
+TEST_F(WebRtcLocalAudioTrackTest, StartTwoAudioTracks) {
+ scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter1(
+ WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
+ scoped_ptr<WebRtcLocalAudioTrack> track1(
+ new WebRtcLocalAudioTrack(adapter1, capturer_, NULL));
+ track1->Start();
+
+ scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2(
+ WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
+ scoped_ptr<WebRtcLocalAudioTrack> track2(
+ new WebRtcLocalAudioTrack(adapter2, capturer_, NULL));
+ track2->Start();
+
+ track1->Stop();
+ // When the last track is stopped, it will automatically stop the
+ // |capturer_source_|.
+ EXPECT_CALL(*capturer_source_.get(), OnStop());
+ track2->Stop();
+}
+
// Start/Stop tracks and verify the capturer is correctly starting/stopping
// its source.
TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
- // Starting the first audio track will start the |capturer_source_|.
base::WaitableEvent event(false, false);
- EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
- EXPECT_CALL(*capturer_source_.get(), OnStart()).WillOnce(SignalEvent(&event));
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track_1(
static_cast<webrtc::AudioTrackInterface*>(
adapter_1.get())->GetRenderer()->AddChannel(0);
track_1->Start();
- EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
// Verify the data flow by connecting the sink to |track_1|.
scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
// Create a new capturer with new source, connect it to a new audio track.
TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
// Setup the first audio track and start it.
- EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
- EXPECT_CALL(*capturer_source_.get(), OnStart());
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track_1(
StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
std::string(), std::string());
scoped_refptr<WebRtcAudioCapturer> new_capturer(
- WebRtcAudioCapturer::CreateCapturer(-1, device, constraints, NULL));
+ WebRtcAudioCapturer::CreateCapturer(-1, device, constraints, NULL, NULL));
scoped_refptr<MockCapturerSource> new_source(
new MockCapturerSource(new_capturer.get()));
EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1));
+ EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
+ EXPECT_CALL(*new_source.get(), OnStart());
+
media::AudioParameters new_param(
media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
media::CHANNEL_LAYOUT_MONO, 44100, 16, 441);
new_capturer->SetCapturerSourceForTesting(new_source, new_param);
// Setup the second audio track, connect it to the new capturer and start it.
- EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
- EXPECT_CALL(*new_source.get(), OnStart());
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track_2(
capturer_->Stop();
}
-
// Make sure a audio track can deliver packets with a buffer size smaller than
// 10ms when it is not connected with a peer connection.
TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
params.channel_layout(),
params.frames_per_buffer()),
factory.CreateWebMediaConstraints(),
- NULL));
+ NULL, NULL));
scoped_refptr<MockCapturerSource> source(
new MockCapturerSource(capturer.get()));
EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1));
+ EXPECT_CALL(*source.get(), SetAutomaticGainControl(true));
+ EXPECT_CALL(*source.get(), OnStart());
capturer->SetCapturerSourceForTesting(source, params);
// Setup a audio track, connect it to the capturer and start it.
- EXPECT_CALL(*source.get(), SetAutomaticGainControl(true));
- EXPECT_CALL(*source.get(), OnStart());
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
scoped_ptr<WebRtcLocalAudioTrack> track(
// Stopping the new source will stop the second track.
EXPECT_CALL(*source, OnStop()).Times(1);
capturer->Stop();
+
+ // Even though this test don't use |capturer_source_| it will be stopped
+ // during teardown of the test harness.
+ EXPECT_CALL(*capturer_source_.get(), OnStop());
}
} // namespace content
projects / platform / framework / web / crosswalk.git / blobdiff