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

content/browser/renderer_host/media/mock_media_observer.h

@@ -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

@@ -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

@@ -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(),