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Unified Diff: content/renderer/media/webrtc/processed_local_audio_track_unittest.cc

Issue 1647773002: MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: NOT FOR REVIEW -- This will be broken-up across multiple CLs. Created 4 years, 10 months ago
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Index: content/renderer/media/webrtc/processed_local_audio_track_unittest.cc
diff --git a/content/renderer/media/webrtc_local_audio_track_unittest.cc b/content/renderer/media/webrtc/processed_local_audio_track_unittest.cc
similarity index 54%
rename from content/renderer/media/webrtc_local_audio_track_unittest.cc
rename to content/renderer/media/webrtc/processed_local_audio_track_unittest.cc
index 986a536af913afcfa42c745020fab826639b66c1..989c3c3664327ab2cdb892bc0775b5eb80930d6b 100644
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc
+++ b/content/renderer/media/webrtc/processed_local_audio_track_unittest.cc
@@ -7,8 +7,9 @@
#include "base/test/test_timeouts.h"
#include "build/build_config.h"
#include "content/public/renderer/media_stream_audio_sink.h"
-#include "content/renderer/media/media_stream_audio_source.h"
#include "content/renderer/media/mock_media_constraint_factory.h"
+#include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h"
+#include "content/renderer/media/webrtc/processed_local_audio_source.h"
#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
#include "content/renderer/media/webrtc_audio_capturer.h"
#include "content/renderer/media/webrtc_local_audio_track.h"
@@ -95,7 +96,7 @@ class MockCapturerSource : public media::AudioCapturerSource {
int session_id));
MOCK_METHOD0(OnStart, void());
MOCK_METHOD0(OnStop, void());
- MOCK_METHOD1(SetVolume, void(double volume));
+ void SetVolume(double volume) final {} // Ignored.
MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));
void Initialize(const media::AudioParameters& params,
@@ -154,38 +155,50 @@ class MockMediaStreamAudioSink : public MediaStreamAudioSink {
class WebRtcLocalAudioTrackTest : public ::testing::Test {
protected:
void SetUp() override {
- params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
- media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480);
+ CreateSourceWithParams(media::AudioParameters(
+ media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+ media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480));
+ }
+
+ void TearDown() override {
+ blink_source_.reset();
+ blink::WebHeap::collectAllGarbageForTesting();
+ }
+
+ void CreateSourceWithParams(const media::AudioParameters& params) {
+ params_ = params;
MockMediaConstraintFactory constraint_factory;
blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio,
"dummy",
false /* remote */, true /* readonly */);
- MediaStreamAudioSource* audio_source = new MediaStreamAudioSource();
+ ProcessedLocalAudioSource* const audio_source =
+ new ProcessedLocalAudioSource(
+ -1,
+ StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, std::string(),
+ std::string(), params_.sample_rate(),
+ params_.channel_layout(),
+ params_.frames_per_buffer()),
+ &mock_dependency_factory_);
+ audio_source->SetAllowInvalidRenderFrameIdForTesting(true);
+ audio_source->SetSourceConstraints(
+ MockMediaConstraintFactory().CreateWebMediaConstraints());
blink_source_.setExtraData(audio_source);
- StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
- std::string(), std::string());
- 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))
+ audio_source->StartSourceForTesting();
+ capturer_ = audio_source->audio_capturer();
+ capturer_source_ = new MockCapturerSource(capturer_);
+ EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_, -1))
.WillOnce(Return());
EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
EXPECT_CALL(*capturer_source_.get(), OnStart());
capturer_->SetCapturerSource(capturer_source_, params_);
}
- void TearDown() override {
- blink_source_.reset();
- blink::WebHeap::collectAllGarbageForTesting();
- }
-
+ MockPeerConnectionDependencyFactory mock_dependency_factory_;
media::AudioParameters params_;
blink::WebMediaStreamSource blink_source_;
scoped_refptr<MockCapturerSource> capturer_source_;
- scoped_refptr<WebRtcAudioCapturer> capturer_;
+ WebRtcAudioCapturer* capturer_; // Owned by ProcessedLocalAudioSource.
};
// Creates a capturer and audio track, fakes its audio thread, and
@@ -193,12 +206,15 @@ class WebRtcLocalAudioTrackTest : public ::testing::Test {
// 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) {
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
- scoped_ptr<WebRtcLocalAudioTrack> track(
- new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
- track->Start();
- EXPECT_TRUE(track->GetAudioAdapter()->enabled());
+ blink::WebMediaStreamTrack blink_track;
+ blink_track.initialize(blink::WebString::fromUTF8("dummy_track"),
+ blink_source_);
+ ASSERT_TRUE(
+ MediaStreamAudioSource::Get(blink_source_)->ConnectToTrack(blink_track));
+
+ WebRtcLocalAudioTrack* const track = WebRtcLocalAudioTrack::From(
+ MediaStreamAudioTrack::Get(blink_track));
+ EXPECT_TRUE(track->adapter()->enabled());
scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
base::WaitableEvent event(false, false);
@@ -221,15 +237,16 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
// TODO(xians): Enable this test after resolving the racing issue that TSAN
// reports on MediaStreamTrack::enabled();
TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
- 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(
- new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
- track->Start();
- EXPECT_TRUE(track->GetAudioAdapter()->enabled());
- EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false));
+ blink::WebMediaStreamTrack blink_track;
+ blink_track.initialize(blink::WebString::fromUTF8("dummy_track"),
+ blink_source_);
+ ASSERT_TRUE(
+ MediaStreamAudioSource::Get(blink_source_)->ConnectToTrack(blink_track));
+
+ WebRtcLocalAudioTrack* const track = WebRtcLocalAudioTrack::From(
+ MediaStreamAudioTrack::Get(blink_track));
+ EXPECT_TRUE(track->adapter()->enabled());
+
scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
const media::AudioParameters params = capturer_->source_audio_parameters();
base::WaitableEvent event(false, false);
@@ -243,25 +260,28 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
event.Reset();
EXPECT_CALL(*sink, CaptureData()).Times(AtLeast(1))
.WillRepeatedly(SignalEvent(&event));
- EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(true));
+ EXPECT_TRUE(track->adapter()->set_enabled(true));
EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
track->RemoveSink(sink.get());
EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
capturer_->Stop();
- track.reset();
}
// Create multiple audio tracks and enable/disable them, verify that the audio
// callbacks appear/disappear.
// Flaky due to a data race, see http://crbug.com/295418
TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
- scoped_ptr<WebRtcLocalAudioTrack> track_1(
- new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
- track_1->Start();
- EXPECT_TRUE(track_1->GetAudioAdapter()->enabled());
+ blink::WebMediaStreamTrack blink_track_1;
+ blink_track_1.initialize(blink::WebString::fromUTF8("dummy_track"),
+ blink_source_);
+ ASSERT_TRUE(MediaStreamAudioSource::Get(blink_source_)
+ ->ConnectToTrack(blink_track_1));
+
+ WebRtcLocalAudioTrack* const track_1 = WebRtcLocalAudioTrack::From(
+ MediaStreamAudioTrack::Get(blink_track_1));
+ EXPECT_TRUE(track_1->adapter()->enabled());
+
scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
const media::AudioParameters params = capturer_->source_audio_parameters();
base::WaitableEvent event_1(false, false);
@@ -273,12 +293,15 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
track_1->AddSink(sink_1.get());
EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
- scoped_ptr<WebRtcLocalAudioTrack> track_2(
- new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL));
- track_2->Start();
- EXPECT_TRUE(track_2->GetAudioAdapter()->enabled());
+ blink::WebMediaStreamTrack blink_track_2;
+ blink_track_2.initialize(blink::WebString::fromUTF8("dummy_track"),
+ blink_source_);
+ ASSERT_TRUE(MediaStreamAudioSource::Get(blink_source_)
+ ->ConnectToTrack(blink_track_2));
+
+ WebRtcLocalAudioTrack* const track_2 = WebRtcLocalAudioTrack::From(
+ MediaStreamAudioTrack::Get(blink_track_2));
+ EXPECT_TRUE(track_2->adapter()->enabled());
// Verify both |sink_1| and |sink_2| get data.
event_1.Reset();
@@ -300,207 +323,86 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
track_1->RemoveSink(sink_1.get());
track_1->Stop();
- track_1.reset();
+ blink_track_1.reset();
EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
track_2->RemoveSink(sink_2.get());
track_2->Stop();
- track_2.reset();
+ blink_track_2.reset();
}
// 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) {
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
- scoped_ptr<WebRtcLocalAudioTrack> track(
- new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
- track->Start();
+ blink::WebMediaStreamTrack blink_track;
+ blink_track.initialize(blink::WebString::fromUTF8("dummy_track"),
+ blink_source_);
+ ASSERT_TRUE(
+ MediaStreamAudioSource::Get(blink_source_)->ConnectToTrack(blink_track));
+
+ WebRtcLocalAudioTrack* const track = WebRtcLocalAudioTrack::From(
+ MediaStreamAudioTrack::Get(blink_track));
+ EXPECT_TRUE(track->adapter()->enabled());
// When the track goes away, it will automatically stop the
// |capturer_source_|.
EXPECT_CALL(*capturer_source_.get(), OnStop());
- 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.get(), capturer_, NULL));
- track1->Start();
-
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2(
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
- scoped_ptr<WebRtcLocalAudioTrack> track2(
- new WebRtcLocalAudioTrack(adapter2.get(), 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) {
- base::WaitableEvent event(false, false);
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
- scoped_ptr<WebRtcLocalAudioTrack> track_1(
- 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());
- event.Reset();
- EXPECT_CALL(*sink, FormatIsSet()).WillOnce(SignalEvent(&event));
- EXPECT_CALL(*sink, CaptureData())
- .Times(AnyNumber()).WillRepeatedly(Return());
- track_1->AddSink(sink.get());
- EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
-
- // Start the second audio track will not start the |capturer_source_|
+ blink::WebMediaStreamTrack blink_track_1;
+ blink_track_1.initialize(blink::WebString::fromUTF8("dummy_track"),
+ blink_source_);
+ ASSERT_TRUE(MediaStreamAudioSource::Get(blink_source_)
+ ->ConnectToTrack(blink_track_1));
+
+ blink::WebMediaStreamTrack blink_track_2;
+ blink_track_2.initialize(blink::WebString::fromUTF8("dummy_track"),
+ blink_source_);
+ // Starting the second audio track will not start the |capturer_source_|
// since it has been started.
EXPECT_CALL(*capturer_source_.get(), OnStart()).Times(0);
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
- scoped_ptr<WebRtcLocalAudioTrack> track_2(
- 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());
- capturer_->Stop();
-
- // Adding a new track to the capturer.
- track_2->AddSink(sink.get());
- EXPECT_CALL(*sink, FormatIsSet()).Times(0);
-
- // Stop the capturer again will not trigger stopping the source of the
- // capturer again..
- event.Reset();
- EXPECT_CALL(*capturer_source_.get(), OnStop()).Times(0);
- capturer_->Stop();
-}
-
-// Create a new capturer with new source, connect it to a new audio track.
-#if defined(THREAD_SANITIZER)
-// Fails under TSan, see https://crbug.com/576634.
-#define MAYBE_ConnectTracksToDifferentCapturers \
- DISABLED_ConnectTracksToDifferentCapturers
-#else
-#define MAYBE_ConnectTracksToDifferentCapturers \
- ConnectTracksToDifferentCapturers
-#endif
-TEST_F(WebRtcLocalAudioTrackTest, MAYBE_ConnectTracksToDifferentCapturers) {
- // Setup the first audio track and start it.
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
- scoped_ptr<WebRtcLocalAudioTrack> track_1(
- 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());
- EXPECT_CALL(*sink_1.get(), CaptureData())
- .Times(AnyNumber()).WillRepeatedly(Return());
- EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber());
- track_1->AddSink(sink_1.get());
-
- // Create a new capturer with new source with different audio format.
- MockMediaConstraintFactory constraint_factory;
- StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
- std::string(), std::string());
- scoped_refptr<WebRtcAudioCapturer> new_capturer(
- WebRtcAudioCapturer::CreateCapturer(
- -1, device, constraint_factory.CreateWebMediaConstraints(), 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->SetCapturerSource(new_source, new_param);
-
- // Setup the second audio track, connect it to the new capturer and start it.
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
- scoped_ptr<WebRtcLocalAudioTrack> track_2(
- new WebRtcLocalAudioTrack(adapter_2.get(), new_capturer, NULL));
- track_2->Start();
+ ASSERT_TRUE(MediaStreamAudioSource::Get(blink_source_)
+ ->ConnectToTrack(blink_track_2));
- // Verify the data flow by connecting the |sink_2| to |track_2|.
- scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
- base::WaitableEvent event(false, false);
- EXPECT_CALL(*sink_2, CaptureData())
- .Times(AnyNumber()).WillRepeatedly(Return());
- EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event));
- track_2->AddSink(sink_2.get());
- EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
-
- // Stopping the new source will stop the second track.
- event.Reset();
- EXPECT_CALL(*new_source.get(), OnStop())
- .Times(1).WillOnce(SignalEvent(&event));
- new_capturer->Stop();
- EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
+ WebRtcLocalAudioTrack::From(MediaStreamAudioTrack::Get(blink_track_1))
+ ->Stop();
- // Stop the capturer of the first audio track.
+ // When the last track is stopped, it will automatically stop the
+ // |capturer_source_|.
EXPECT_CALL(*capturer_source_.get(), OnStop());
- capturer_->Stop();
+ WebRtcLocalAudioTrack::From(MediaStreamAudioTrack::Get(blink_track_2))
+ ->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) {
// Setup a capturer which works with a buffer size smaller than 10ms.
- media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
- media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128);
-
- // Create a capturer with new source which works with the format above.
- MockMediaConstraintFactory factory;
- factory.DisableDefaultAudioConstraints();
- 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));
- EXPECT_CALL(*source.get(), SetAutomaticGainControl(true));
- EXPECT_CALL(*source.get(), OnStart());
- capturer->SetCapturerSource(source, params);
+ CreateSourceWithParams(media::AudioParameters(
+ media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+ media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128));
// Setup a audio track, connect it to the capturer and start it.
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
- scoped_ptr<WebRtcLocalAudioTrack> track(
- new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
- track->Start();
+ blink::WebMediaStreamTrack blink_track;
+ blink_track.initialize(blink::WebString::fromUTF8("dummy_track"),
+ blink_source_);
+ ASSERT_TRUE(
+ MediaStreamAudioSource::Get(blink_source_)->ConnectToTrack(blink_track));
+ WebRtcLocalAudioTrack* const track = WebRtcLocalAudioTrack::From(
+ MediaStreamAudioTrack::Get(blink_track));
+ EXPECT_TRUE(track->adapter()->enabled());
// Verify the data flow by connecting the |sink| to |track|.
scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
base::WaitableEvent event(false, false);
EXPECT_CALL(*sink, FormatIsSet()).Times(1);
// Verify the sinks are getting the packets with an expecting buffer size.
-#if defined(OS_ANDROID)
- const int expected_buffer_size = params.sample_rate() / 100;
-#else
- const int expected_buffer_size = params.frames_per_buffer();
-#endif
+ const int expected_buffer_size = params_.sample_rate() / 100;
EXPECT_CALL(*sink, CaptureData())
.Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
track->AddSink(sink.get());
@@ -508,12 +410,8 @@ TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer());
// Stopping the new source will stop the second track.
- EXPECT_CALL(*source.get(), 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());
+ EXPECT_CALL(*capturer_source_, OnStop()).Times(AtLeast(1));
+ capturer_->Stop();
}
} // namespace content
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