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Commit aaf48c8b authored by henrika@chromium.org's avatar henrika@chromium.org
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Re-enables three WebRTC unit tests in content. 

It looks like some disabled unit tests could easily be enabled again just by extending the MockMediaInternals class.

BUG=23533039
TEST=content_unittests

Review URL: https://chromiumcodereview.appspot.com/23533039

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@221784 0039d316-1c4b-4281-b951-d872f2087c98
parent f53194f2
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Inline Side-by-side
Showing with 62 additions and 36 deletions
content/browser/renderer_host/media/mock_media_observer.h
View file @ aaf48c8b
......@@ -10,6 +10,7 @@
#include "base/basictypes.h"
#include "content/browser/media/media_internals.h"
#include "content/public/browser/media_observer.h"
#include "media/audio/audio_parameters.h"
#include "media/base/media_log_event.h"
#include "testing/gmock/include/gmock/gmock.h"
......@@ -47,6 +48,10 @@ class MockMediaInternals : public MediaInternals {
void(void* host, int stream_id));
MOCK_METHOD3(OnSetAudioStreamPlaying,
void(void* host, int stream_id, bool playing));
MOCK_METHOD4(OnAudioStreamCreated,
void(void* host, int stream_id,
const media::AudioParameters& params,
const std::string& input_device_id));
MOCK_METHOD3(OnSetAudioStreamStatus,
void(void* host, int stream_id, const std::string& status));
MOCK_METHOD3(OnSetAudioStreamVolume,
......
content/renderer/media/webrtc_audio_device_impl.cc
View file @ aaf48c8b
......@@ -120,6 +120,7 @@ void WebRtcAudioDeviceImpl::RenderData(uint8* audio_data,
DCHECK_LE(number_of_frames, output_buffer_size());
{
base::AutoLock auto_lock(lock_);
DCHECK(audio_transport_callback_);
// Store the reported audio delay locally.
output_delay_ms_ = audio_delay_milliseconds;
}
......
content/renderer/media/webrtc_audio_device_unittest.cc
View file @ aaf48c8b
......@@ -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();
......@@ -767,10 +774,13 @@ 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(),
......@@ -778,17 +788,13 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, MAYBE_FullDuplexAudioWithAGC) {
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;
......@@ -909,10 +926,13 @@ 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(),
......
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