trunk/src/content/renderer/media/webrtc_local_audio_track_unittest.cc - Issue 22871007: Revert 217768 "Adding key press detection in the browser process."

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

Issue 22871007: Revert 217768 "Adding key press detection in the browser process." (Closed) Base URL: svn://svn.chromium.org/chrome/

Patch Set: Created 7 years, 4 months ago

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Index: trunk/src/content/renderer/media/webrtc_local_audio_track_unittest.cc

===================================================================

--- trunk/src/content/renderer/media/webrtc_local_audio_track_unittest.cc (revision 217773)

+++ trunk/src/content/renderer/media/webrtc_local_audio_track_unittest.cc (working copy)

@@ -50,7 +50,7 @@

static_cast<media::AudioCapturerSource::CaptureCallback*>(

capturer_.get());

audio_bus_->Zero();

- callback->Capture(audio_bus_.get(), 0, 0, false);

+ callback->Capture(audio_bus_.get(), 0, 0);

// Sleep 1ms to yield the resource for the main thread.

base::PlatformThread::Sleep(base::TimeDelta::FromMilliseconds(1));

@@ -103,27 +103,20 @@

int number_of_frames,

int audio_delay_milliseconds,

int current_volume,

- 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);

+ bool need_audio_processing) OVERRIDE {

+ CaptureData(channels.size(), sample_rate, number_of_channels,

+ number_of_frames, audio_delay_milliseconds, current_volume,

+ need_audio_processing);

return 0;

}

- 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_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_METHOD1(SetCaptureFormat, void(const media::AudioParameters& params));

};

@@ -180,16 +173,10 @@

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,

- false,

- false)).Times(AtLeast(1))

- .WillRepeatedly(SignalEvent(&event));

+ EXPECT_CALL(*sink, CaptureData(

+ kNumberOfNetworkChannels, params.sample_rate(), params.channels(),

+ params.frames_per_buffer(), 0, 0, false))

+ .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));

track->AddSink(sink.get());

EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));

@@ -220,29 +207,18 @@

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,

- false,

- false)).Times(0);

+ EXPECT_CALL(*sink, CaptureData(

+ 1, params.sample_rate(), params.channels(),

+ params.frames_per_buffer(), 0, 0, 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,

- false,

- false)).Times(AtLeast(1))

- .WillRepeatedly(SignalEvent(&event));

+ EXPECT_CALL(*sink, CaptureData(

+ 1, params.sample_rate(), params.channels(),

+ params.frames_per_buffer(), 0, 0, false))

+ .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));

EXPECT_TRUE(track->set_enabled(true));

EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));

track->RemoveSink(sink.get());

@@ -267,16 +243,10 @@

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,

- false,

- 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))

+ .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_1));

track_1->AddSink(sink_1.get());

EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));

@@ -294,26 +264,14 @@

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,

- 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,

- false,

- 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))

+ .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));

track_2->AddSink(sink_2.get());

EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));

EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));

@@ -361,7 +319,7 @@

scoped_ptr<MockWebRtcAudioCapturerSink> sink(

new MockWebRtcAudioCapturerSink());

event.Reset();

- EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))

+ EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false))

.Times(AnyNumber()).WillRepeatedly(Return());

EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);

track_1->AddSink(sink.get());

@@ -381,7 +339,7 @@

track_1->Stop();

track_1 = NULL;

- EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))

+ EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false))

.Times(AnyNumber()).WillRepeatedly(Return());

EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);

track_2->AddSink(sink.get());

@@ -438,10 +396,8 @@

// 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, false))

+ EXPECT_CALL(*sink_1.get(), CaptureData(kNumberOfNetworkChannelsForTrack1,

+ 48000, 2, _, 0, 0, false))

.Times(AnyNumber()).WillRepeatedly(Return());

EXPECT_CALL(*sink_1.get(), SetCaptureFormat(_)).Times(1);

track_1->AddSink(sink_1.get());

@@ -477,10 +433,8 @@

// 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, false))

+ EXPECT_CALL(*sink_2, CaptureData(kNumberOfNetworkChannelsForTrack2,

+ 44100, 1, _, 0, 0, false))

.Times(AnyNumber()).WillRepeatedly(Return());

EXPECT_CALL(*sink_2, SetCaptureFormat(_)).Times(1);

track_2->AddSink(sink_2.get());

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