Revert 187936 "Pass more detailed audio hardware configuration i..."

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/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_audio_renderer.h"
 #include "content/renderer/render_thread_impl.h"
 #include "content/test/webrtc_audio_device_test.h"
 #include "media/audio/audio_manager_base.h"
 #include "media/base/audio_hardware_config.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"
 #include "third_party/webrtc/voice_engine/include/voe_network.h"
 
-using media::AudioParameters;
 using testing::_;
 using testing::AnyNumber;
 using testing::InvokeWithoutArgs;
 using testing::Return;
 using testing::StrEq;
 
 namespace content {
 
 namespace {
 
 const int kRenderViewId = 1;
 
 scoped_ptr<media::AudioHardwareConfig> CreateRealHardwareConfig(
     media::AudioManager* manager) {
-  const AudioParameters output_parameters =
+  const media::AudioParameters output_parameters =
       manager->GetDefaultOutputStreamParameters();
-  const AudioParameters input_parameters =
+  const media::AudioParameters input_parameters =
       manager->GetInputStreamParameters(
           media::AudioManagerBase::kDefaultDeviceId);
   return make_scoped_ptr(new media::AudioHardwareConfig(
-      input_parameters, output_parameters));
+      output_parameters.frames_per_buffer(), output_parameters.sample_rate(),
+      input_parameters.sample_rate(), input_parameters.channel_layout()));
 }
 
 // Return true if at least one element in the array matches |value|.
 bool FindElementInArray(const int* array, int size, int value) {
   return (std::find(&array[0], &array[0] + size, value) != &array[size]);
 }
 
 // This method returns false if a non-supported rate is detected on the
 // input or output side.
 // TODO(henrika): add support for automatic fallback to Windows Wave audio
@@ skipping 158 matching lines @@
   int invalid_rates[] = {-1, 0, 8000, 11025, 22050, 32000, 192000};
   for (size_t i = 0; i < arraysize(invalid_rates); ++i) {
     EXPECT_FALSE(FindElementInArray(valid_rates, arraysize(valid_rates),
         invalid_rates[i]));
   }
 }
 
 // Basic test that instantiates and initializes an instance of
 // WebRtcAudioDeviceImpl.
 TEST_F(WebRTCAudioDeviceTest, Construct) {
-  AudioParameters input_params(
-      AudioParameters::AUDIO_PCM_LOW_LATENCY,
-      media::CHANNEL_LAYOUT_MONO,
-      48000,
-      16,
-      480);
-
-  AudioParameters output_params(
-      AudioParameters::AUDIO_PCM_LOW_LATENCY,
-      media::CHANNEL_LAYOUT_STEREO,
-      48000,
-      16,
-      480);
-
-  media::AudioHardwareConfig audio_config(input_params, output_params);
+  media::AudioHardwareConfig audio_config(
+      480, 48000, 48000, media::CHANNEL_LAYOUT_MONO);
   SetAudioHardwareConfig(&audio_config);
 
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
 
   // 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());
@@ skipping 326 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