Remove MockMediaInternals and incestuous usage.

content/renderer/media/webrtc_audio_device_unittest.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include <vector>
 
 #include "base/environment.h"
 #include "base/file_util.h"
 #include "base/files/file_path.h"
 #include "base/path_service.h"
@@ skipping 494 matching lines @@
 #endif
 TEST_F(MAYBE_WebRTCAudioDeviceTest, MAYBE_StartPlayout) {
   if (!has_output_devices_) {
     LOG(WARNING) << "No output device detected.";
     return;
   }
 
   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)).Times(1);
-  EXPECT_CALL(media_observer(),
-      OnSetAudioStreamStatus(_, 1, StrEq("closed"))).Times(1);
-  EXPECT_CALL(media_observer(),
-      OnDeleteAudioStream(_, 1)).Times(AnyNumber());
-
   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);
 
@@ skipping 152 matching lines @@
     LOG(WARNING) << "No output device detected.";
     return;
   }
 
   std::string file_path(
       GetTestDataPath(FILE_PATH_LITERAL("speechmusic_mono_16kHz.pcm")));
 
   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)).Times(1);
-  EXPECT_CALL(media_observer(),
-      OnSetAudioStreamStatus(_, 1, StrEq("closed"))).Times(1);
-  EXPECT_CALL(media_observer(),
-      OnDeleteAudioStream(_, 1)).Times(AnyNumber());
-
   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();
@@ skipping 47 matching lines @@
 #endif
 TEST_F(MAYBE_WebRTCAudioDeviceTest, MAYBE_FullDuplexAudioWithAGC) {
   if (!has_output_devices_ || !has_input_devices_) {
     LOG(WARNING) << "Missing audio devices.";
     return;
   }
 
   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());
-
   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);
 
@@ skipping 120 matching lines @@
 #endif
 TEST_F(MAYBE_WebRTCAudioDeviceTest, MAYBE_WebRtcPlayoutSetupTime) {
   if (!has_output_devices_) {
     LOG(WARNING) << "No output device detected.";
     return;
   }
 
   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));
-  EXPECT_CALL(media_observer(),
-              OnDeleteAudioStream(_, 1)).Times(AnyNumber());
-
   base::WaitableEvent event(false, false);
   scoped_ptr<MockWebRtcAudioRendererSource> renderer_source(
       new MockWebRtcAudioRendererSource(&event));
 
   scoped_refptr<WebRtcAudioRenderer> renderer(
       CreateDefaultWebRtcAudioRenderer(kRenderViewId));
   renderer->Initialize(renderer_source.get());
   scoped_refptr<MediaStreamAudioRenderer> proxy(
       renderer->CreateSharedAudioRendererProxy());
   proxy->Start();
@@ skipping 48 matching lines @@
     LOG(WARNING) << "Test disabled due to the test hangs on WinXP.";
     return;
   }
 #endif
   int latency = RunWebRtcLoopbackTimeTest(audio_manager_.get(), true);
   PrintPerfResultMs("webrtc_loopback_with_signal_processing (100 packets)",
                     "t", latency);
 }
 
 }  // namespace content