Add sample rates checking in MediaStreamAudioProcessor

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 <vector>
 
 #include "base/files/file_path.h"
 #include "base/files/file_util.h"
 #include "base/logging.h"
 #include "base/memory/aligned_memory.h"
@@ skipping 438 matching lines @@
   MockMediaConstraintFactory constraint_factory;
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
       new rtc::RefCountedObject<MediaStreamAudioProcessor>(
           constraint_factory.CreateWebMediaConstraints(), input_device_params_,
           webrtc_audio_device.get()));
   EXPECT_TRUE(audio_processor->has_audio_processing());
 
   static const int kSupportedSampleRates[] =
-      { 8000, 16000, 22050, 32000, 44100, 48000, 88200, 96000 };
+      { 8000, 16000, 22050, 32000, 44100, 48000 };
   for (size_t i = 0; i < arraysize(kSupportedSampleRates); ++i) {
     int buffer_size = (kSupportedSampleRates[i] / 100)  < 128 ?
         kSupportedSampleRates[i] / 100 : 128;
     media::AudioParameters params(
         media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
         media::CHANNEL_LAYOUT_STEREO, kSupportedSampleRates[i], 16,
         buffer_size);
     audio_processor->OnCaptureFormatChanged(params);
     VerifyDefaultComponents(audio_processor.get());
 
@@ skipping 118 matching lines @@
   ProcessDataAndVerifyFormat(audio_processor.get(),
                              kAudioProcessingSampleRate,
                              kAudioProcessingNumberOfChannel,
                              kAudioProcessingSampleRate / 100);
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
   // |audio_processor|.
   audio_processor = NULL;
 }
 
 }  // namespace content