Adds support for 192kHz native output sample rate for WebRTC clients

content/renderer/media/webrtc_audio_renderer.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 "content/renderer/media/webrtc_audio_renderer.h"
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
 #include "content/renderer/media/audio_device_factory.h"
@@ skipping 105 matching lines @@
   media::ChannelLayout channel_layout = media::CHANNEL_LAYOUT_MONO;
 #if defined(OS_WIN)
   channel_layout = media::CHANNEL_LAYOUT_STEREO;
 #endif
 
   // Ask the renderer for the default audio output hardware sample-rate.
   media::AudioHardwareConfig* hardware_config =
       RenderThreadImpl::current()->GetAudioHardwareConfig();
   int sample_rate = hardware_config->GetOutputSampleRate();
   DVLOG(1) << "Audio output hardware sample rate: " << sample_rate;
+
+  // WebRTC does not yet support higher rates than 96000 on the client side
+  // and 48000 is the preferred sample rate. Therefore, if 192000 is detected,
+  // we change the rate to 48000 instead. The consequence is that the native
+  // layer will be opened up at 192kHz but WebRTC will provide data at 48kHz
+  // which will then be resampled by the audio converted on the browser side
+  // to match the native audio layer.
+  if (sample_rate == 192000) {
+    DVLOG(1) << "Resampling from 48000 to 192000 is required";
+    sample_rate = 48000;
+  }
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputSampleRate",
                             sample_rate, media::kUnexpectedAudioSampleRate);
 
   // Verify that the reported output hardware sample rate is supported
   // on the current platform.
   if (std::find(&kValidOutputRates[0],
                 &kValidOutputRates[0] + arraysize(kValidOutputRates),
                 sample_rate) ==
                     &kValidOutputRates[arraysize(kValidOutputRates)]) {
     DLOG(ERROR) << sample_rate << " is not a supported output rate.";
@@ skipping 207 matching lines @@
   }
 
   // De-interleave each channel and convert to 32-bit floating-point
   // with nominal range -1.0 -> +1.0 to match the callback format.
   audio_bus->FromInterleaved(buffer_.get(),
                              audio_bus->frames(),
                              sizeof(buffer_[0]));
 }
 
 }  // namespace content