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

Issue 1834323002: MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: REBASE + Workaround to ensure MediaStreamAudioProcessor is destroyed on the main thread. Created 4 years, 7 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
deleted file mode 100644
index 51af138563639080e6770681332ea88ed4b5508a..0000000000000000000000000000000000000000
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc
+++ /dev/null
@@ -1,579 +0,0 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file.
-
-#include "base/macros.h"
-#include "base/synchronization/waitable_event.h"
-#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_constraint_factory.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"
-#include "media/base/audio_bus.h"
-#include "media/base/audio_capturer_source.h"
-#include "media/base/audio_parameters.h"
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "third_party/WebKit/public/platform/WebMediaConstraints.h"
-#include "third_party/WebKit/public/web/WebHeap.h"
-#include "third_party/webrtc/api/mediastreaminterface.h"
-
-using ::testing::_;
-using ::testing::AnyNumber;
-using ::testing::AtLeast;
-using ::testing::Return;
-
-namespace content {
-
-namespace {
-
-ACTION_P(SignalEvent, event) {
- event->Signal();
-}
-
-// A simple thread that we use to fake the audio thread which provides data to
-// the |WebRtcAudioCapturer|.
-class FakeAudioThread : public base::PlatformThread::Delegate {
- public:
- FakeAudioThread(WebRtcAudioCapturer* capturer,
- const media::AudioParameters& params)
- : capturer_(capturer),
- thread_(),
- closure_(false, false) {
- DCHECK(capturer);
- audio_bus_ = media::AudioBus::Create(params);
- }
-
- ~FakeAudioThread() override { DCHECK(thread_.is_null()); }
-
- // base::PlatformThread::Delegate:
- void ThreadMain() override {
- while (true) {
- if (closure_.IsSignaled())
- return;
-
- media::AudioCapturerSource::CaptureCallback* callback =
- static_cast<media::AudioCapturerSource::CaptureCallback*>(
- capturer_);
- audio_bus_->Zero();
- callback->Capture(audio_bus_.get(), 0, 0, false);
-
- // Sleep 1ms to yield the resource for the main thread.
- base::PlatformThread::Sleep(base::TimeDelta::FromMilliseconds(1));
- }
- }
-
- void Start() {
- base::PlatformThread::CreateWithPriority(
- 0, this, &thread_, base::ThreadPriority::REALTIME_AUDIO);
- CHECK(!thread_.is_null());
- }
-
- void Stop() {
- closure_.Signal();
- base::PlatformThread::Join(thread_);
- thread_ = base::PlatformThreadHandle();
- }
-
- private:
- std::unique_ptr<media::AudioBus> audio_bus_;
- WebRtcAudioCapturer* capturer_;
- base::PlatformThreadHandle thread_;
- base::WaitableEvent closure_;
- DISALLOW_COPY_AND_ASSIGN(FakeAudioThread);
-};
-
-class MockCapturerSource : public media::AudioCapturerSource {
- public:
- explicit MockCapturerSource(WebRtcAudioCapturer* capturer)
- : capturer_(capturer) {}
- MOCK_METHOD3(OnInitialize, void(const media::AudioParameters& params,
- CaptureCallback* callback,
- int session_id));
- MOCK_METHOD0(OnStart, void());
- MOCK_METHOD0(OnStop, void());
- void SetVolume(double volume) final {}
- MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));
-
- void Initialize(const media::AudioParameters& params,
- CaptureCallback* callback,
- int session_id) override {
- DCHECK(params.IsValid());
- params_ = params;
- OnInitialize(params, callback, session_id);
- }
- void Start() override {
- audio_thread_.reset(new FakeAudioThread(capturer_, params_));
- audio_thread_->Start();
- OnStart();
- }
- void Stop() override {
- audio_thread_->Stop();
- audio_thread_.reset();
- OnStop();
- }
-
- protected:
- ~MockCapturerSource() override {}
-
- private:
- std::unique_ptr<FakeAudioThread> audio_thread_;
- WebRtcAudioCapturer* capturer_;
- media::AudioParameters params_;
-};
-
-class MockMediaStreamAudioSink : public MediaStreamAudioSink {
- public:
- MockMediaStreamAudioSink() {}
- ~MockMediaStreamAudioSink() {}
- void OnData(const media::AudioBus& audio_bus,
- base::TimeTicks estimated_capture_time) override {
- EXPECT_EQ(params_.channels(), audio_bus.channels());
- EXPECT_EQ(params_.frames_per_buffer(), audio_bus.frames());
- EXPECT_FALSE(estimated_capture_time.is_null());
- CaptureData();
- }
- MOCK_METHOD0(CaptureData, void());
- void OnSetFormat(const media::AudioParameters& params) {
- params_ = params;
- FormatIsSet();
- }
- MOCK_METHOD0(FormatIsSet, void());
-
- const media::AudioParameters& audio_params() const { return params_; }
-
- private:
- media::AudioParameters params_;
-};
-
-} // namespace
-
-class WebRtcLocalAudioTrackTest : public ::testing::Test {
- protected:
- void SetUp() override {
- params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
- media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480);
- MockConstraintFactory constraint_factory;
- blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio,
- "dummy",
- false /* remote */);
- MediaStreamAudioSource* audio_source = new MediaStreamAudioSource();
- blink_source_.setExtraData(audio_source);
-
- StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
- std::string(), std::string());
- {
- std::unique_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))
- .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();
- }
-
- media::AudioParameters params_;
- blink::WebMediaStreamSource blink_source_;
- WebRtcAudioCapturer* capturer_; // Owned by |blink_source_|.
- scoped_refptr<MockCapturerSource> capturer_source_;
-};
-
-// Creates a capturer and audio track, fakes its audio thread, and
-// connect/disconnect the sink to the audio track on the fly, the sink should
-// 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));
- std::unique_ptr<WebRtcLocalAudioTrack> track(
- new WebRtcLocalAudioTrack(adapter.get()));
- track->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track.get()));
- capturer_->AddTrack(track.get());
- EXPECT_TRUE(track->GetAudioAdapter()->enabled());
-
- std::unique_ptr<MockMediaStreamAudioSink> sink(
- new MockMediaStreamAudioSink());
- base::WaitableEvent event(false, false);
- EXPECT_CALL(*sink, FormatIsSet());
- EXPECT_CALL(*sink,
- CaptureData()).Times(AtLeast(1))
- .WillRepeatedly(SignalEvent(&event));
- track->AddSink(sink.get());
- EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
- track->RemoveSink(sink.get());
-
- EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
- capturer_->Stop();
-}
-
-// The same setup as ConnectAndDisconnectOneSink, but enable and disable the
-// audio track on the fly. When the audio track is disabled, there is no data
-// callback to the sink; when the audio track is enabled, there comes data
-// callback.
-// 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));
- std::unique_ptr<WebRtcLocalAudioTrack> track(
- new WebRtcLocalAudioTrack(adapter.get()));
- track->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track.get()));
- capturer_->AddTrack(track.get());
- EXPECT_TRUE(track->GetAudioAdapter()->enabled());
- EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false));
- std::unique_ptr<MockMediaStreamAudioSink> sink(
- new MockMediaStreamAudioSink());
- const media::AudioParameters params = capturer_->GetInputFormat();
- base::WaitableEvent event(false, false);
- EXPECT_CALL(*sink, FormatIsSet()).Times(1);
- EXPECT_CALL(*sink, CaptureData()).Times(0);
- EXPECT_EQ(sink->audio_params().frames_per_buffer(),
- params.sample_rate() / 100);
- track->AddSink(sink.get());
- EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
-
- event.Reset();
- EXPECT_CALL(*sink, CaptureData()).Times(AtLeast(1))
- .WillRepeatedly(SignalEvent(&event));
- EXPECT_TRUE(track->GetAudioAdapter()->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));
- std::unique_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());
- EXPECT_TRUE(track_1->GetAudioAdapter()->enabled());
- std::unique_ptr<MockMediaStreamAudioSink> sink_1(
- new MockMediaStreamAudioSink());
- const media::AudioParameters params = capturer_->GetInputFormat();
- base::WaitableEvent event_1(false, false);
- EXPECT_CALL(*sink_1, FormatIsSet()).WillOnce(Return());
- EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1))
- .WillRepeatedly(SignalEvent(&event_1));
- EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
- params.sample_rate() / 100);
- track_1->AddSink(sink_1.get());
- EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
-
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
- std::unique_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());
- EXPECT_TRUE(track_2->GetAudioAdapter()->enabled());
-
- // Verify both |sink_1| and |sink_2| get data.
- event_1.Reset();
- base::WaitableEvent event_2(false, false);
-
- std::unique_ptr<MockMediaStreamAudioSink> sink_2(
- new MockMediaStreamAudioSink());
- EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(Return());
- EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1))
- .WillRepeatedly(SignalEvent(&event_1));
- EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
- params.sample_rate() / 100);
- EXPECT_CALL(*sink_2, CaptureData()).Times(AtLeast(1))
- .WillRepeatedly(SignalEvent(&event_2));
- EXPECT_EQ(sink_2->audio_params().frames_per_buffer(),
- params.sample_rate() / 100);
- track_2->AddSink(sink_2.get());
- EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
- EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));
-
- track_1->RemoveSink(sink_1.get());
- track_1->Stop();
- track_1.reset();
-
- EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
- track_2->RemoveSink(sink_2.get());
- track_2->Stop();
- 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));
- std::unique_ptr<WebRtcLocalAudioTrack> track(
- new WebRtcLocalAudioTrack(adapter.get()));
- track->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track.get()));
- capturer_->AddTrack(track.get());
-
- // 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));
- std::unique_ptr<WebRtcLocalAudioTrack> track1(
- new WebRtcLocalAudioTrack(adapter1.get()));
- track1->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track1.get()));
- capturer_->AddTrack(track1.get());
-
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2(
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
- std::unique_ptr<WebRtcLocalAudioTrack> track2(
- new WebRtcLocalAudioTrack(adapter2.get()));
- track2->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track2.get()));
- capturer_->AddTrack(track2.get());
-
- 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));
- std::unique_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());
-
- // Verify the data flow by connecting the sink to |track_1|.
- std::unique_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_|
- // since it has been started.
- EXPECT_CALL(*capturer_source_.get(), OnStart()).Times(0);
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
- std::unique_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());
-
- // 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));
- std::unique_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());
-
- // Verify the data flow by connecting the |sink_1| to |track_1|.
- std::unique_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.
- MockConstraintFactory constraint_factory;
- StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
- std::string(), std::string());
- std::unique_ptr<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));
- std::unique_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());
-
- // Verify the data flow by connecting the |sink_2| to |track_2|.
- std::unique_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()));
-
- // Stop the capturer of the first audio track.
- EXPECT_CALL(*capturer_source_.get(), OnStop());
- 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) {
- // 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.
- MockConstraintFactory factory;
- factory.DisableDefaultAudioConstraints();
- std::unique_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<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);
-
- // Setup a audio track, connect it to the capturer and start it.
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
- std::unique_ptr<WebRtcLocalAudioTrack> track(
- new WebRtcLocalAudioTrack(adapter.get()));
- track->Start(
- base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
- MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
- track.get()));
- capturer->AddTrack(track.get());
-
- // Verify the data flow by connecting the |sink| to |track|.
- std::unique_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
- EXPECT_CALL(*sink, CaptureData())
- .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
- track->AddSink(sink.get());
- EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
- 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());
-}
-
-} // namespace content
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