Ensures that WebRTC works for device selection using a different sample rate than default

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 "base/environment.h"
 #include "base/test/test_timeouts.h"
 #include "content/renderer/media/audio_hardware.h"
+#include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_audio_renderer.h"
 #include "content/test/webrtc_audio_device_test.h"
 #include "media/audio/audio_manager.h"
 #include "media/audio/audio_util.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "third_party/webrtc/voice_engine/include/voe_audio_processing.h"
 #include "third_party/webrtc/voice_engine/include/voe_base.h"
 #include "third_party/webrtc/voice_engine/include/voe_external_media.h"
 #include "third_party/webrtc/voice_engine/include/voe_file.h"
@@ skipping 89 matching lines @@
   int output_sample_rate = GetAudioOutputSampleRate();
   if (!FindElementInArray(valid_output_rates, arraysize(valid_output_rates),
                           output_sample_rate)) {
     LOG(WARNING) << "Non-supported output sample rate detected.";
     return false;
   }
 
   return true;
 }
 
+// Utility method which initializes the audio capturer contained in the
+// WebRTC audio device. This method should be used in tests where
+// HardwareSampleRatesAreValid() has been called and returned true.
+bool InitializeCapturer(WebRtcAudioDeviceImpl* webrtc_audio_device) {
+  // Access the capturer owned and created by the audio device.
+  WebRtcAudioCapturer* capturer = webrtc_audio_device->capturer();
+  if (!capturer)
+    return false;
+
+  // Use native capture sample rate and channel configuration to get some
+  // action in this test.
+  int sample_rate = GetAudioInputSampleRate();
+  media::ChannelLayout channel_layout = GetAudioInputChannelLayout();
+  if (!capturer->Initialize(channel_layout, sample_rate))
+    return false;
+
+  // Ensures that the default capture device is utilized.
+  webrtc_audio_device->capturer()->SetDevice(1);
+  return true;
+}
+
 
 class WebRTCMediaProcessImpl : public webrtc::VoEMediaProcess {
  public:
   explicit WebRTCMediaProcessImpl(base::WaitableEvent* event)
       : event_(event),
         channel_id_(-1),
         type_(webrtc::kPlaybackPerChannel),
         packet_size_(0),
         sample_rate_(0),
         channels_(0) {
@@ skipping 95 matching lines @@
 }
 
 // Basic test that instantiates and initializes an instance of
 // WebRtcAudioDeviceImpl.
 TEST_F(WebRTCAudioDeviceTest, Construct) {
   AudioUtilNoHardware audio_util(48000, 48000, media::CHANNEL_LAYOUT_MONO);
   SetAudioUtilCallback(&audio_util);
 
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
-  webrtc_audio_device->SetSessionId(1);
+
+  // The capturer is not created until after the WebRtcAudioDeviceImpl has
+  // been initialized.
+  EXPECT_FALSE(InitializeCapturer(webrtc_audio_device.get()));
 
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   int err = base->Init(webrtc_audio_device);
+  EXPECT_TRUE(InitializeCapturer(webrtc_audio_device.get()));
   EXPECT_EQ(0, err);
   EXPECT_EQ(0, base->Terminate());
 }
 
 // Verify that a call to webrtc::VoEBase::StartPlayout() starts audio output
 // with the correct set of parameters. A WebRtcAudioDeviceImpl instance will
 // 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.
@@ skipping 16 matching lines @@
       OnSetAudioStreamPlaying(_, 1, true)).Times(1);
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("closed"))).Times(1);
   EXPECT_CALL(media_observer(),
       OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
   scoped_refptr<WebRtcAudioRenderer> renderer =
       new WebRtcAudioRenderer(kRenderViewId);
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
-  webrtc_audio_device->SetSessionId(1);
   EXPECT_TRUE(webrtc_audio_device->SetRenderer(renderer));
 
   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(webrtc_audio_device);
   ASSERT_EQ(0, err);
 
@@ skipping 50 matching lines @@
   SetAudioUtilCallback(&audio_util);
 
   if (!HardwareSampleRatesAreValid())
     return;
 
   // TODO(tommi): extend MediaObserver and MockMediaObserver with support
   // for new interfaces, like OnSetAudioStreamRecording(). When done, add
   // EXPECT_CALL() macros here.
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
-  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(webrtc_audio_device);
   ASSERT_EQ(0, err);
 
+  EXPECT_TRUE(InitializeCapturer(webrtc_audio_device.get()));
+
   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));
   EXPECT_EQ(0, external_media->RegisterExternalMediaProcessing(
@@ skipping 50 matching lines @@
       OnSetAudioStreamPlaying(_, 1, true)).Times(1);
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("closed"))).Times(1);
   EXPECT_CALL(media_observer(),
       OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
   scoped_refptr<WebRtcAudioRenderer> renderer =
       new WebRtcAudioRenderer(kRenderViewId);
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
-  webrtc_audio_device->SetSessionId(1);
   EXPECT_TRUE(webrtc_audio_device->SetRenderer(renderer));
 
   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(webrtc_audio_device);
   ASSERT_EQ(0, err);
 
@@ skipping 51 matching lines @@
       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());
-  webrtc_audio_device->SetSessionId(1);
   EXPECT_TRUE(webrtc_audio_device->SetRenderer(renderer));
 
   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(webrtc_audio_device);
   ASSERT_EQ(0, err);
 
+  EXPECT_TRUE(InitializeCapturer(webrtc_audio_device.get()));
+
   ScopedWebRTCPtr<webrtc::VoEAudioProcessing> audio_processing(engine.get());
   ASSERT_TRUE(audio_processing.valid());
   bool enabled = false;
   webrtc::AgcModes agc_mode =  webrtc::kAgcDefault;
   EXPECT_EQ(0, audio_processing->GetAgcStatus(enabled, agc_mode));
   EXPECT_TRUE(enabled);
   EXPECT_EQ(agc_mode, webrtc::kAgcAdaptiveAnalog);
 
   int ch = base->CreateChannel();
   EXPECT_NE(-1, ch);
@@ skipping 15 matching lines @@
 
   renderer->Stop();
   EXPECT_EQ(0, base->StopSend(ch));
   EXPECT_EQ(0, base->StopPlayout(ch));
 
   EXPECT_EQ(0, base->DeleteChannel(ch));
   EXPECT_EQ(0, base->Terminate());
 }
 
 }  // namespace content