content/renderer/media/webrtc_audio_device_unittest.cc - Issue 10537121: Adds AudioDevice factory for all audio clients in Chrome

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

Issue 10537121: Adds AudioDevice factory for all audio clients in Chrome (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src

Patch Set: Minor cleanup Created 8 years, 6 months ago

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« content/renderer/media/audio_device.cc ('K') | « content/renderer/media/webrtc_audio_device_impl.cc ('k') | content/renderer/render_view_impl.cc » ('j') | content/renderer/renderer_webkitplatformsupport_impl.cc » ('J')
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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 cf3a45364f5ee6642c1da1e46a7342b4536e97a9..944741d02eb6f8b5cd51ccabd9505916e30cc774 100644

--- a/content/renderer/media/webrtc_audio_device_unittest.cc

+++ b/content/renderer/media/webrtc_audio_device_unittest.cc

@@ -232,16 +232,16 @@ TEST_F(WebRTCAudioDeviceTest, TestValidOutputRates) {

TEST_F(WebRTCAudioDeviceTest, Construct) {

AudioUtilNoHardware audio_util(48000, 48000, CHANNEL_LAYOUT_MONO);

SetAudioUtilCallback(&audio_util);

- scoped_refptr<WebRtcAudioDeviceImpl> audio_device(

+ scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(

new WebRtcAudioDeviceImpl());

- audio_device->SetSessionId(1);

+ webrtc_audio_device->SetSessionId(1);

WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());

ASSERT_TRUE(engine.valid());

ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());

- int err = base->Init(audio_device);

+ int err = base->Init(webrtc_audio_device);

EXPECT_EQ(0, err);

EXPECT_EQ(0, base->Terminate());

}

@@ -273,15 +273,15 @@ TEST_F(WebRTCAudioDeviceTest, StartPlayout) {

EXPECT_CALL(media_observer(),

OnDeleteAudioStream(_, 1)).Times(AnyNumber());

- scoped_refptr<WebRtcAudioDeviceImpl> audio_device(

+ scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(

new WebRtcAudioDeviceImpl());

- audio_device->SetSessionId(1);

+ webrtc_audio_device->SetSessionId(1);

WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());

ASSERT_TRUE(engine.valid());

ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());

ASSERT_TRUE(base.valid());

- int err = base->Init(audio_device);

+ int err = base->Init(webrtc_audio_device);

ASSERT_EQ(0, err);

int ch = base->CreateChannel();

@@ -302,8 +302,8 @@ TEST_F(WebRTCAudioDeviceTest, StartPlayout) {

base::TimeDelta::FromMilliseconds(TestTimeouts::action_timeout_ms())));

WaitForIOThreadCompletion();

- EXPECT_TRUE(audio_device->playing());

- EXPECT_FALSE(audio_device->recording());

+ EXPECT_TRUE(webrtc_audio_device->playing());

+ EXPECT_FALSE(webrtc_audio_device->recording());

EXPECT_EQ(ch, media_process->channel_id());

EXPECT_EQ(webrtc::kPlaybackPerChannel, media_process->type());

EXPECT_EQ(80, media_process->packet_size());

@@ -342,15 +342,15 @@ TEST_F(WebRTCAudioDeviceTest, StartRecording) {

// for new interfaces, like OnSetAudioStreamRecording(). When done, add

// EXPECT_CALL() macros here.

- scoped_refptr<WebRtcAudioDeviceImpl> audio_device(

+ scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(

new WebRtcAudioDeviceImpl());

- audio_device->SetSessionId(1);

+ webrtc_audio_device->SetSessionId(1);

WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());

ASSERT_TRUE(engine.valid());

ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());

ASSERT_TRUE(base.valid());

- int err = base->Init(audio_device);

+ int err = base->Init(webrtc_audio_device);

ASSERT_EQ(0, err);

int ch = base->CreateChannel();

@@ -378,8 +378,8 @@ TEST_F(WebRTCAudioDeviceTest, StartRecording) {

base::TimeDelta::FromMilliseconds(TestTimeouts::action_timeout_ms())));

WaitForIOThreadCompletion();

- EXPECT_FALSE(audio_device->playing());

- EXPECT_TRUE(audio_device->recording());

+ EXPECT_FALSE(webrtc_audio_device->playing());

+ EXPECT_TRUE(webrtc_audio_device->recording());

EXPECT_EQ(ch, media_process->channel_id());

EXPECT_EQ(webrtc::kRecordingPerChannel, media_process->type());

EXPECT_EQ(80, media_process->packet_size());

@@ -419,16 +419,16 @@ TEST_F(WebRTCAudioDeviceTest, PlayLocalFile) {

EXPECT_CALL(media_observer(),

OnDeleteAudioStream(_, 1)).Times(AnyNumber());

- scoped_refptr<WebRtcAudioDeviceImpl> audio_device(

+ scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(

new WebRtcAudioDeviceImpl());

- audio_device->SetSessionId(1);

+ webrtc_audio_device->SetSessionId(1);

WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());

ASSERT_TRUE(engine.valid());

ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());

ASSERT_TRUE(base.valid());

- int err = base->Init(audio_device);

+ int err = base->Init(webrtc_audio_device);

ASSERT_EQ(0, err);

int ch = base->CreateChannel();

@@ -487,15 +487,15 @@ TEST_F(WebRTCAudioDeviceTest, FullDuplexAudioWithAGC) {

EXPECT_CALL(media_observer(),

OnDeleteAudioStream(_, 1)).Times(AnyNumber());

- scoped_refptr<WebRtcAudioDeviceImpl> audio_device(

+ scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(

new WebRtcAudioDeviceImpl());

- audio_device->SetSessionId(1);

+ webrtc_audio_device->SetSessionId(1);

WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());

ASSERT_TRUE(engine.valid());

ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());

ASSERT_TRUE(base.valid());

- int err = base->Init(audio_device);

+ int err = base->Init(webrtc_audio_device);

ASSERT_EQ(0, err);

ScopedWebRTCPtr<webrtc::VoEAudioProcessing> audio_processing(engine.get());