| Index: content/renderer/media/webrtc_audio_device_unittest.cc
|
| diff --git a/content/renderer/media/webrtc_audio_device_unittest.cc b/content/renderer/media/webrtc_audio_device_unittest.cc
|
| index 37d5401c558b9e1aadfed78d7aadc2ddf2e87135..4dac1f8f2831f2e19f34a502e7fad0391e09b903 100644
|
| --- a/content/renderer/media/webrtc_audio_device_unittest.cc
|
| +++ b/content/renderer/media/webrtc_audio_device_unittest.cc
|
| @@ -11,6 +11,7 @@
|
| #include "content/renderer/media/webrtc_audio_capturer.h"
|
| #include "content/renderer/media/webrtc_audio_device_impl.h"
|
| #include "content/renderer/media/webrtc_audio_renderer.h"
|
| +#include "content/renderer/media/webrtc_local_audio_track.h"
|
| #include "content/renderer/render_thread_impl.h"
|
| #include "content/test/webrtc_audio_device_test.h"
|
| #include "media/audio/audio_manager_base.h"
|
| @@ -582,7 +583,14 @@ TEST_F(WebRTCAudioDeviceTest, MAYBE_StartRecording) {
|
| ASSERT_EQ(0, err);
|
|
|
| EXPECT_TRUE(InitializeCapturer(webrtc_audio_device.get()));
|
| - webrtc_audio_device->capturer()->Start();
|
| +
|
| + // Create and start a local audio track. Starting the audio track will connect
|
| + // the audio track to the capturer and also start the source of the capturer.
|
| + scoped_refptr<WebRtcLocalAudioTrack> local_audio_track(
|
| + WebRtcLocalAudioTrack::Create(std::string(),
|
| + webrtc_audio_device->capturer(),
|
| + NULL));
|
| + local_audio_track->Start();
|
|
|
| int ch = base->CreateChannel();
|
| EXPECT_NE(-1, ch);
|
| @@ -619,7 +627,7 @@ TEST_F(WebRTCAudioDeviceTest, MAYBE_StartRecording) {
|
| ch, webrtc::kRecordingPerChannel));
|
| EXPECT_EQ(0, base->StopSend(ch));
|
|
|
| - webrtc_audio_device->capturer()->Stop();
|
| + local_audio_track->Stop();
|
| EXPECT_EQ(0, base->DeleteChannel(ch));
|
| EXPECT_EQ(0, base->Terminate());
|
| }
|
| @@ -745,7 +753,13 @@ TEST_F(WebRTCAudioDeviceTest, MAYBE_FullDuplexAudioWithAGC) {
|
| ASSERT_EQ(0, err);
|
|
|
| EXPECT_TRUE(InitializeCapturer(webrtc_audio_device.get()));
|
| - webrtc_audio_device->capturer()->Start();
|
| + // Create and start a local audio track. Starting the audio track will connect
|
| + // the audio track to the capturer and also start the source of the capturer.
|
| + scoped_refptr<WebRtcLocalAudioTrack> local_audio_track(
|
| + WebRtcLocalAudioTrack::Create(std::string(),
|
| + webrtc_audio_device->capturer(),
|
| + NULL));
|
| + local_audio_track->Start();
|
|
|
| ScopedWebRTCPtr<webrtc::VoEAudioProcessing> audio_processing(engine.get());
|
| ASSERT_TRUE(audio_processing.valid());
|
| @@ -781,7 +795,7 @@ TEST_F(WebRTCAudioDeviceTest, MAYBE_FullDuplexAudioWithAGC) {
|
| base::TimeDelta::FromSeconds(2));
|
| message_loop_.Run();
|
|
|
| - webrtc_audio_device->capturer()->Stop();
|
| + local_audio_track->Stop();
|
| renderer->Stop();
|
| EXPECT_EQ(0, base->StopSend(ch));
|
| EXPECT_EQ(0, base->StopPlayout(ch));
|
| @@ -815,7 +829,13 @@ TEST_F(WebRTCAudioDeviceTest, WebRtcRecordingSetupTime) {
|
| ASSERT_EQ(0, err);
|
|
|
| EXPECT_TRUE(InitializeCapturer(webrtc_audio_device.get()));
|
| - webrtc_audio_device->capturer()->Start();
|
| + // Create and start a local audio track. Starting the audio track will connect
|
| + // the audio track to the capturer and also start the source of the capturer.
|
| + scoped_refptr<WebRtcLocalAudioTrack> local_audio_track(
|
| + WebRtcLocalAudioTrack::Create(std::string(),
|
| + webrtc_audio_device->capturer(),
|
| + NULL));
|
| + local_audio_track->Start();
|
|
|
| base::WaitableEvent event(false, false);
|
| scoped_ptr<MockWebRtcAudioCapturerSink> capturer_sink(
|
| @@ -834,7 +854,7 @@ TEST_F(WebRTCAudioDeviceTest, WebRtcRecordingSetupTime) {
|
| PrintPerfResultMs("webrtc_recording_setup_c", "t", delay);
|
|
|
| capturer->RemoveSink(capturer_sink.get());
|
| - webrtc_audio_device->capturer()->Stop();
|
| + local_audio_track->Stop();
|
| EXPECT_EQ(0, base->StopSend(ch));
|
| EXPECT_EQ(0, base->DeleteChannel(ch));
|
| EXPECT_EQ(0, base->Terminate());
|
|
|