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

Issue 23533039: Re-enables three WebRTC unit tests in content (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Disabled Android and improved tests for all other platforms Created 7 years, 3 months ago
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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 7074c95bb09c3e28966169cb62e91300918c89ec..ac41e4f4b00d9e88eb39d179ba0126a14df84a1a 100644
--- a/content/renderer/media/webrtc_audio_device_unittest.cc
+++ b/content/renderer/media/webrtc_audio_device_unittest.cc
@@ -496,9 +496,14 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, Construct) {
// be utilized to implement the actual audio path. The test registers a
// webrtc::VoEExternalMedia implementation to hijack the output audio and
// verify that streaming starts correctly.
-// Disabled when running headless since the bots don't have the required config.
-// Flaky, http://crbug.com/167299 .
-TEST_F(MAYBE_WebRTCAudioDeviceTest, DISABLED_StartPlayout) {
+// TODO(henrika): include on Android as well as soon as alla race conditions
+// in OpenSLES are resolved.
+#if defined(OS_ANDROID)
+#define MAYBE_StartPlayout DISABLED_StartPlayout
+#else
+#define MAYBE_StartPlayout StartPlayout
+#endif
+TEST_F(MAYBE_WebRTCAudioDeviceTest, MAYBE_StartPlayout) {
if (!has_output_devices_) {
LOG(WARNING) << "No output device detected.";
return;
@@ -507,12 +512,13 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, DISABLED_StartPlayout) {
scoped_ptr<media::AudioHardwareConfig> config =
CreateRealHardwareConfig(audio_manager_.get());
SetAudioHardwareConfig(config.get());
+ media::AudioParameters params(config->GetOutputConfig());
if (!HardwareSampleRatesAreValid())
return;
EXPECT_CALL(media_observer(),
- OnSetAudioStreamStatus(_, 1, StrEq("created"))).Times(1);
+ OnAudioStreamCreated(_, 1, params, StrEq(""))).Times(1);
EXPECT_CALL(media_observer(),
OnSetAudioStreamPlaying(_, 1, true)).Times(1);
EXPECT_CALL(media_observer(),
@@ -520,33 +526,30 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, DISABLED_StartPlayout) {
EXPECT_CALL(media_observer(),
OnDeleteAudioStream(_, 1)).Times(AnyNumber());
- scoped_refptr<WebRtcAudioRenderer> renderer =
- new WebRtcAudioRenderer(kRenderViewId);
- scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
- new WebRtcAudioDeviceImpl());
- EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
-
WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
ASSERT_TRUE(engine.valid());
-
ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
ASSERT_TRUE(base.valid());
+
+ scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
+ new WebRtcAudioDeviceImpl());
int err = base->Init(webrtc_audio_device.get());
ASSERT_EQ(0, err);
- int ch = base->CreateChannel();
- EXPECT_NE(-1, ch);
-
ScopedWebRTCPtr<webrtc::VoEExternalMedia> external_media(engine.get());
ASSERT_TRUE(external_media.valid());
-
base::WaitableEvent event(false, false);
scoped_ptr<WebRTCMediaProcessImpl> media_process(
new WebRTCMediaProcessImpl(&event));
+ int ch = base->CreateChannel();
+ EXPECT_NE(-1, ch);
EXPECT_EQ(0, external_media->RegisterExternalMediaProcessing(
ch, webrtc::kPlaybackPerChannel, *media_process.get()));
EXPECT_EQ(0, base->StartPlayout(ch));
+ scoped_refptr<WebRtcAudioRenderer> renderer =
+ new WebRtcAudioRenderer(kRenderViewId);
+ EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
renderer->Play();
EXPECT_TRUE(event.TimedWait(TestTimeouts::action_timeout()));
@@ -674,9 +677,14 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, MAYBE_StartRecording) {
}
// Uses WebRtcAudioDeviceImpl to play a local wave file.
-// Disabled when running headless since the bots don't have the required config.
-// Flaky, http://crbug.com/167298 .
-TEST_F(MAYBE_WebRTCAudioDeviceTest, DISABLED_PlayLocalFile) {
+// TODO(henrika): include on Android as well as soon as alla race conditions
+// in OpenSLES are resolved.
+#if defined(OS_ANDROID)
+#define MAYBE_PlayLocalFile DISABLED_PlayLocalFile
+#else
+#define MAYBE_PlayLocalFile PlayLocalFile
+#endif
+TEST_F(MAYBE_WebRTCAudioDeviceTest, MAYBE_PlayLocalFile) {
if (!has_output_devices_) {
LOG(WARNING) << "No output device detected.";
return;
@@ -688,12 +696,13 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, DISABLED_PlayLocalFile) {
scoped_ptr<media::AudioHardwareConfig> config =
CreateRealHardwareConfig(audio_manager_.get());
SetAudioHardwareConfig(config.get());
+ media::AudioParameters params(config->GetOutputConfig());
if (!HardwareSampleRatesAreValid())
return;
EXPECT_CALL(media_observer(),
- OnSetAudioStreamStatus(_, 1, StrEq("created"))).Times(1);
+ OnAudioStreamCreated(_, 1, params, StrEq(""))).Times(1);
EXPECT_CALL(media_observer(),
OnSetAudioStreamPlaying(_, 1, true)).Times(1);
EXPECT_CALL(media_observer(),
@@ -701,23 +710,21 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, DISABLED_PlayLocalFile) {
EXPECT_CALL(media_observer(),
OnDeleteAudioStream(_, 1)).Times(AnyNumber());
- scoped_refptr<WebRtcAudioRenderer> renderer =
- new WebRtcAudioRenderer(kRenderViewId);
- scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
- new WebRtcAudioDeviceImpl());
- EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
-
WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
ASSERT_TRUE(engine.valid());
-
ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
ASSERT_TRUE(base.valid());
+
+ scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
+ new WebRtcAudioDeviceImpl());
int err = base->Init(webrtc_audio_device.get());
ASSERT_EQ(0, err);
-
int ch = base->CreateChannel();
EXPECT_NE(-1, ch);
EXPECT_EQ(0, base->StartPlayout(ch));
+ scoped_refptr<WebRtcAudioRenderer> renderer =
+ new WebRtcAudioRenderer(kRenderViewId);
+ EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
renderer->Play();
ScopedWebRTCPtr<webrtc::VoEFile> file(engine.get());
@@ -733,7 +740,7 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, DISABLED_PlayLocalFile) {
// Play 2 seconds worth of audio and then quit.
message_loop_.PostDelayedTask(FROM_HERE,
base::MessageLoop::QuitClosure(),
- base::TimeDelta::FromSeconds(6));
+ base::TimeDelta::FromSeconds(2));
message_loop_.Run();
renderer->Stop();
@@ -767,28 +774,27 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, MAYBE_FullDuplexAudioWithAGC) {
scoped_ptr<media::AudioHardwareConfig> config =
CreateRealHardwareConfig(audio_manager_.get());
SetAudioHardwareConfig(config.get());
+ media::AudioParameters params(config->GetOutputConfig());
if (!HardwareSampleRatesAreValid())
return;
EXPECT_CALL(media_observer(),
+ OnAudioStreamCreated(_, 1, params, StrEq(""))).Times(1);
+ EXPECT_CALL(media_observer(),
OnSetAudioStreamPlaying(_, 1, true));
EXPECT_CALL(media_observer(),
OnSetAudioStreamStatus(_, 1, StrEq("closed")));
EXPECT_CALL(media_observer(),
OnDeleteAudioStream(_, 1)).Times(AnyNumber());
- scoped_refptr<WebRtcAudioRenderer> renderer =
- new WebRtcAudioRenderer(kRenderViewId);
- scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
- new WebRtcAudioDeviceImpl());
- EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
-
WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
ASSERT_TRUE(engine.valid());
-
ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
ASSERT_TRUE(base.valid());
+
+ scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
+ new WebRtcAudioDeviceImpl());
int err = base->Init(webrtc_audio_device.get());
ASSERT_EQ(0, err);
@@ -826,6 +832,9 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, MAYBE_FullDuplexAudioWithAGC) {
EXPECT_EQ(0, network->RegisterExternalTransport(ch, *transport.get()));
EXPECT_EQ(0, base->StartPlayout(ch));
EXPECT_EQ(0, base->StartSend(ch));
+ scoped_refptr<WebRtcAudioRenderer> renderer =
+ new WebRtcAudioRenderer(kRenderViewId);
+ EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
renderer->Play();
LOG(INFO) << ">> You should now be able to hear yourself in loopback...";
@@ -900,7 +909,15 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, DISABLED_WebRtcRecordingSetupTime) {
EXPECT_EQ(0, base->Terminate());
}
-TEST_F(MAYBE_WebRTCAudioDeviceTest, WebRtcPlayoutSetupTime) {
+
+// TODO(henrika): include on Android as well as soon as alla race conditions
+// in OpenSLES are resolved.
+#if defined(OS_ANDROID)
+#define MAYBE_WebRtcPlayoutSetupTime DISABLED_WebRtcPlayoutSetupTime
+#else
+#define MAYBE_WebRtcPlayoutSetupTime WebRtcPlayoutSetupTime
+#endif
+TEST_F(MAYBE_WebRTCAudioDeviceTest, MAYBE_WebRtcPlayoutSetupTime) {
if (!has_output_devices_) {
LOG(WARNING) << "No output device detected.";
return;
@@ -909,11 +926,14 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, WebRtcPlayoutSetupTime) {
scoped_ptr<media::AudioHardwareConfig> config =
CreateRealHardwareConfig(audio_manager_.get());
SetAudioHardwareConfig(config.get());
+ media::AudioParameters params(config->GetOutputConfig());
if (!HardwareSampleRatesAreValid())
return;
EXPECT_CALL(media_observer(),
+ OnAudioStreamCreated(_, 1, params, StrEq(""))).Times(1);
+ EXPECT_CALL(media_observer(),
OnSetAudioStreamStatus(_, 1, _)).Times(AnyNumber());
EXPECT_CALL(media_observer(),
OnSetAudioStreamPlaying(_, 1, true));
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