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

Issue 21183002: Adding key press detection in the browser process. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 7 years, 4 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 4720e45bf01ac95f9029642dd1b6008fe2c672a3..b75ca79b920f230ac68f68b1658808df5daba2b3 100644
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc
+++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc
@@ -50,7 +50,7 @@ class FakeAudioThread : public base::PlatformThread::Delegate {
static_cast<media::AudioCapturerSource::CaptureCallback*>(
capturer_.get());
audio_bus_->Zero();
- callback->Capture(audio_bus_.get(), 0, 0);
+ 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));
@@ -103,20 +103,27 @@ class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink {
int number_of_frames,
int audio_delay_milliseconds,
int current_volume,
- bool need_audio_processing) OVERRIDE {
- CaptureData(channels.size(), sample_rate, number_of_channels,
- number_of_frames, audio_delay_milliseconds, current_volume,
- need_audio_processing);
+ bool need_audio_processing,
+ bool key_pressed) OVERRIDE {
+ CaptureData(channels.size(),
+ sample_rate,
+ number_of_channels,
+ number_of_frames,
+ audio_delay_milliseconds,
+ current_volume,
+ need_audio_processing,
+ key_pressed);
return 0;
}
- MOCK_METHOD7(CaptureData, void(int number_of_network_channels,
- int sample_rate,
- int number_of_channels,
- int number_of_frames,
- int audio_delay_milliseconds,
- int current_volume,
- bool need_audio_processing));
-
+ MOCK_METHOD8(CaptureData,
+ void(int number_of_network_channels,
+ int sample_rate,
+ int number_of_channels,
+ int number_of_frames,
+ int audio_delay_milliseconds,
+ int current_volume,
+ bool need_audio_processing,
+ bool key_pressed));
MOCK_METHOD1(SetCaptureFormat, void(const media::AudioParameters& params));
};
@@ -173,10 +180,16 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
const media::AudioParameters params = capturer_->audio_parameters();
base::WaitableEvent event(false, false);
EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
- EXPECT_CALL(*sink, CaptureData(
- kNumberOfNetworkChannels, params.sample_rate(), params.channels(),
- params.frames_per_buffer(), 0, 0, false))
- .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
+ EXPECT_CALL(*sink,
+ CaptureData(kNumberOfNetworkChannels,
+ params.sample_rate(),
+ params.channels(),
+ params.frames_per_buffer(),
+ 0,
+ 0,
+ false,
+ false)).Times(AtLeast(1))
+ .WillRepeatedly(SignalEvent(&event));
track->AddSink(sink.get());
EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
@@ -207,18 +220,29 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
const media::AudioParameters params = capturer_->audio_parameters();
base::WaitableEvent event(false, false);
EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
- EXPECT_CALL(*sink, CaptureData(
- 1, params.sample_rate(), params.channels(),
- params.frames_per_buffer(), 0, 0, false))
- .Times(0);
+ EXPECT_CALL(*sink,
+ CaptureData(1,
+ params.sample_rate(),
+ params.channels(),
+ params.frames_per_buffer(),
+ 0,
+ 0,
+ false,
+ false)).Times(0);
track->AddSink(sink.get());
EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
event.Reset();
- EXPECT_CALL(*sink, CaptureData(
- 1, params.sample_rate(), params.channels(),
- params.frames_per_buffer(), 0, 0, false))
- .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
+ EXPECT_CALL(*sink,
+ CaptureData(1,
+ params.sample_rate(),
+ params.channels(),
+ params.frames_per_buffer(),
+ 0,
+ 0,
+ false,
+ false)).Times(AtLeast(1))
+ .WillRepeatedly(SignalEvent(&event));
EXPECT_TRUE(track->set_enabled(true));
EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
track->RemoveSink(sink.get());
@@ -243,10 +267,16 @@ TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
const media::AudioParameters params = capturer_->audio_parameters();
base::WaitableEvent event_1(false, false);
EXPECT_CALL(*sink_1, SetCaptureFormat(_)).WillOnce(Return());
- EXPECT_CALL(*sink_1, CaptureData(
- 1, params.sample_rate(), params.channels(),
- params.frames_per_buffer(), 0, 0, false))
- .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_1));
+ EXPECT_CALL(*sink_1,
+ CaptureData(1,
+ params.sample_rate(),
+ params.channels(),
+ params.frames_per_buffer(),
+ 0,
+ 0,
+ false,
+ false)).Times(AtLeast(1))
+ .WillRepeatedly(SignalEvent(&event_1));
track_1->AddSink(sink_1.get());
EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
@@ -264,14 +294,26 @@ TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
new MockWebRtcAudioCapturerSink());
EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(Return());
- EXPECT_CALL(*sink_1, CaptureData(
- 1, params.sample_rate(), params.channels(),
- params.frames_per_buffer(), 0, 0, false))
- .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_1));
- EXPECT_CALL(*sink_2, CaptureData(
- 1, params.sample_rate(), params.channels(),
- params.frames_per_buffer(), 0, 0, false))
- .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_2));
+ EXPECT_CALL(*sink_1,
+ CaptureData(1,
+ params.sample_rate(),
+ params.channels(),
+ params.frames_per_buffer(),
+ 0,
+ 0,
+ false,
+ false)).Times(AtLeast(1))
+ .WillRepeatedly(SignalEvent(&event_1));
+ EXPECT_CALL(*sink_2,
+ CaptureData(1,
+ params.sample_rate(),
+ params.channels(),
+ params.frames_per_buffer(),
+ 0,
+ 0,
+ false,
+ false)).Times(AtLeast(1))
+ .WillRepeatedly(SignalEvent(&event_2));
track_2->AddSink(sink_2.get());
EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));
@@ -319,7 +361,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
scoped_ptr<MockWebRtcAudioCapturerSink> sink(
new MockWebRtcAudioCapturerSink());
event.Reset();
- EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false))
+ EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))
.Times(AnyNumber()).WillRepeatedly(Return());
EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
track_1->AddSink(sink.get());
@@ -339,7 +381,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
track_1->Stop();
track_1 = NULL;
- EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false))
+ EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))
.Times(AnyNumber()).WillRepeatedly(Return());
EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
track_2->AddSink(sink.get());
@@ -396,8 +438,10 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
// Verify the data flow by connecting the |sink_1| to |track_1|.
scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
new MockWebRtcAudioCapturerSink());
- EXPECT_CALL(*sink_1.get(), CaptureData(kNumberOfNetworkChannelsForTrack1,
- 48000, 2, _, 0, 0, false))
+ EXPECT_CALL(
+ *sink_1.get(),
+ CaptureData(
+ kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, false, false))
.Times(AnyNumber()).WillRepeatedly(Return());
EXPECT_CALL(*sink_1.get(), SetCaptureFormat(_)).Times(1);
track_1->AddSink(sink_1.get());
@@ -433,8 +477,10 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
// Verify the data flow by connecting the |sink_2| to |track_2|.
scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
new MockWebRtcAudioCapturerSink());
- EXPECT_CALL(*sink_2, CaptureData(kNumberOfNetworkChannelsForTrack2,
- 44100, 1, _, 0, 0, false))
+ EXPECT_CALL(
+ *sink_2,
+ CaptureData(
+ kNumberOfNetworkChannelsForTrack2, 44100, 1, _, 0, 0, false, false))
.Times(AnyNumber()).WillRepeatedly(Return());
EXPECT_CALL(*sink_2, SetCaptureFormat(_)).Times(1);
track_2->AddSink(sink_2.get());
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