Webrtc deps roll.

content/renderer/media/media_stream_audio_processor_unittest.cc

 // Copyright 2013 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/command_line.h"
 #include "base/file_util.h"
 #include "base/files/file_path.h"
 #include "base/logging.h"
 #include "base/memory/aligned_memory.h"
 #include "base/path_service.h"
@@ skipping 144 matching lines @@
 };
 
 TEST_F(MediaStreamAudioProcessorTest, WithoutAudioProcessing) {
   // Setup the audio processor with disabled flag on.
   CommandLine::ForCurrentProcess()->AppendSwitch(
       switches::kDisableAudioTrackProcessing);
   MockMediaConstraintFactory constraint_factory;
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
-      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
+      new rtc::RefCountedObject<MediaStreamAudioProcessor>(
           constraint_factory.CreateWebMediaConstraints(), 0,
           webrtc_audio_device.get()));
   EXPECT_FALSE(audio_processor->has_audio_processing());
   audio_processor->OnCaptureFormatChanged(params_);
 
   ProcessDataAndVerifyFormat(audio_processor,
                              params_.sample_rate(),
                              params_.channels(),
                              params_.sample_rate() / 100);
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
   // |audio_processor|.
   audio_processor = NULL;
 }
 
 TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) {
   MockMediaConstraintFactory constraint_factory;
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
-      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
+      new rtc::RefCountedObject<MediaStreamAudioProcessor>(
           constraint_factory.CreateWebMediaConstraints(), 0,
           webrtc_audio_device.get()));
   EXPECT_TRUE(audio_processor->has_audio_processing());
   audio_processor->OnCaptureFormatChanged(params_);
   VerifyDefaultComponents(audio_processor);
 
   ProcessDataAndVerifyFormat(audio_processor,
                              kAudioProcessingSampleRate,
                              kAudioProcessingNumberOfChannel,
                              kAudioProcessingSampleRate / 100);
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
   // |audio_processor|.
   audio_processor = NULL;
 }
 
 TEST_F(MediaStreamAudioProcessorTest, VerifyTabCaptureWithoutAudioProcessing) {
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   // Create MediaStreamAudioProcessor instance for kMediaStreamSourceTab source.
   MockMediaConstraintFactory tab_constraint_factory;
   const std::string tab_string = kMediaStreamSourceTab;
   tab_constraint_factory.AddMandatory(kMediaStreamSource,
                                       tab_string);
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
-      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
+      new rtc::RefCountedObject<MediaStreamAudioProcessor>(
           tab_constraint_factory.CreateWebMediaConstraints(), 0,
           webrtc_audio_device.get()));
   EXPECT_FALSE(audio_processor->has_audio_processing());
   audio_processor->OnCaptureFormatChanged(params_);
 
   ProcessDataAndVerifyFormat(audio_processor,
                              params_.sample_rate(),
                              params_.channels(),
                              params_.sample_rate() / 100);
 
   // Create MediaStreamAudioProcessor instance for kMediaStreamSourceSystem
   // source.
   MockMediaConstraintFactory system_constraint_factory;
   const std::string system_string = kMediaStreamSourceSystem;
   system_constraint_factory.AddMandatory(kMediaStreamSource,
                                          system_string);
-  audio_processor = new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
+  audio_processor = new rtc::RefCountedObject<MediaStreamAudioProcessor>(
       system_constraint_factory.CreateWebMediaConstraints(), 0,
       webrtc_audio_device.get());
   EXPECT_FALSE(audio_processor->has_audio_processing());
 
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
   // |audio_processor|.
   audio_processor = NULL;
 }
 
 TEST_F(MediaStreamAudioProcessorTest, TurnOffDefaultConstraints) {
   // Turn off the default constraints and pass it to MediaStreamAudioProcessor.
   MockMediaConstraintFactory constraint_factory;
   constraint_factory.DisableDefaultAudioConstraints();
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
-      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
+      new rtc::RefCountedObject<MediaStreamAudioProcessor>(
           constraint_factory.CreateWebMediaConstraints(), 0,
           webrtc_audio_device.get()));
   EXPECT_FALSE(audio_processor->has_audio_processing());
   audio_processor->OnCaptureFormatChanged(params_);
 
   ProcessDataAndVerifyFormat(audio_processor,
                              params_.sample_rate(),
                              params_.channels(),
                              params_.sample_rate() / 100);
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
@@ skipping 95 matching lines @@
   MediaAudioConstraints audio_constraints(
       constraint_factory.CreateWebMediaConstraints(), 0);
   EXPECT_FALSE(audio_constraints.IsValid());
 }
 
 TEST_F(MediaStreamAudioProcessorTest, TestAllSampleRates) {
   MockMediaConstraintFactory constraint_factory;
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
-      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
+      new rtc::RefCountedObject<MediaStreamAudioProcessor>(
           constraint_factory.CreateWebMediaConstraints(), 0,
           webrtc_audio_device.get()));
   EXPECT_TRUE(audio_processor->has_audio_processing());
 
   static const int kSupportedSampleRates[] =
       { 8000, 16000, 22050, 32000, 44100, 48000, 88200, 96000 };
   for (size_t i = 0; i < arraysize(kSupportedSampleRates); ++i) {
     int buffer_size = (kSupportedSampleRates[i] / 100)  < 128 ?
         kSupportedSampleRates[i] / 100 : 128;
     media::AudioParameters params(
@@ skipping 20 matching lines @@
 TEST_F(MediaStreamAudioProcessorTest, GetAecDumpMessageFilter) {
   base::MessageLoopForUI message_loop;
   scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_(
       new AecDumpMessageFilter(message_loop.message_loop_proxy(),
                                message_loop.message_loop_proxy()));
 
   MockMediaConstraintFactory constraint_factory;
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
-      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
+      new rtc::RefCountedObject<MediaStreamAudioProcessor>(
           constraint_factory.CreateWebMediaConstraints(), 0,
           webrtc_audio_device.get()));
 
   EXPECT_TRUE(audio_processor->aec_dump_message_filter_);
 
   audio_processor = NULL;
 }
 
 TEST_F(MediaStreamAudioProcessorTest, TestStereoAudio) {
   // Set up the correct constraints to turn off the audio processing and turn
   // on the stereo channels mirroring.
   MockMediaConstraintFactory constraint_factory;
   constraint_factory.AddMandatory(MediaAudioConstraints::kEchoCancellation,
                                   false);
   constraint_factory.AddMandatory(MediaAudioConstraints::kGoogAudioMirroring,
                                   true);
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
-      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
+      new rtc::RefCountedObject<MediaStreamAudioProcessor>(
           constraint_factory.CreateWebMediaConstraints(), 0,
           webrtc_audio_device.get()));
   EXPECT_FALSE(audio_processor->has_audio_processing());
   const media::AudioParameters source_params(
       media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
       media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480);
   audio_processor->OnCaptureFormatChanged(source_params);
   EXPECT_EQ(audio_processor->OutputFormat().channels(), 2);
 
   // Construct left and right channels, and assign different values to the
@@ skipping 31 matching lines @@
     EXPECT_EQ(output_bus->channel(0)[0], 0);
     EXPECT_NE(output_bus->channel(1)[0], 0);
   }
 
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
   // |audio_processor|.
   audio_processor = NULL;
 }
 
 }  // namespace content