Added 24k sample rate support and removed the validation check in webrtc

content/renderer/media/webrtc_audio_capturer.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_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
 #include "content/child/child_process.h"
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 #include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "content/renderer/media/webrtc_logging.h"
 #include "media/audio/sample_rates.h"
 
 namespace content {
 
 namespace {
 
-// Supported hardware sample rates for input and output sides.
-#if defined(OS_WIN) || defined(OS_MACOSX)
-// media::GetAudioInputHardwareSampleRate() asks the audio layer
-// for its current sample rate (set by the user) on Windows and Mac OS X.
-// The listed rates below adds restrictions and WebRtcAudioDeviceImpl::Init()
-// will fail if the user selects any rate outside these ranges.
-const int kValidInputRates[] =
-    {192000, 96000, 48000, 44100, 32000, 16000, 8000};
-#elif defined(OS_LINUX) || defined(OS_OPENBSD)
-const int kValidInputRates[] = {48000, 44100};
-#elif defined(OS_ANDROID)
-const int kValidInputRates[] = {48000, 44100};
-#else
-const int kValidInputRates[] = {44100};
-#endif
-
 // Time constant for AudioPowerMonitor.  See AudioPowerMonitor ctor comments
 // for semantics.  This value was arbitrarily chosen, but seems to work well.
 const int kPowerMonitorTimeConstantMs = 10;
 
 // The time between two audio power level samples.
 const int kPowerMonitorLogIntervalSeconds = 10;
 
 }  // namespace
 
 // Reference counted container of WebRtcLocalAudioTrack delegate.
@@ skipping 129 matching lines @@
       channel_layout != media::CHANNEL_LAYOUT_STEREO &&
       channel_layout != media::CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC) {
     DLOG(ERROR) << channel_layout
                 << " is not a supported input channel configuration.";
     return false;
   }
 
   DVLOG(1) << "Audio input hardware sample rate: "
            << device_info_.device.input.sample_rate;
   media::AudioSampleRate asr;
   if (media::ToAudioSampleRate(device_info_.device.input.sample_rate, &asr)) {

[henrika (OOO until Aug 14), 2014/09/05 13:08:28]

Still valid to populate this histogram, right?

[no longer working on chromium, 2014/09/05 13:11:57]

Yes, we would like to keep the uma.

     UMA_HISTOGRAM_ENUMERATION(
         "WebRTC.AudioInputSampleRate", asr, media::kAudioSampleRateMax + 1);
   } else {
     UMA_HISTOGRAM_COUNTS("WebRTC.AudioInputSampleRateUnexpected",
                          device_info_.device.input.sample_rate);
   }
 
-  // Verify that the reported input hardware sample rate is supported
-  // on the current platform.
-  if (std::find(&kValidInputRates[0],
-                &kValidInputRates[0] + arraysize(kValidInputRates),
-                device_info_.device.input.sample_rate) ==
-          &kValidInputRates[arraysize(kValidInputRates)]) {
-    DLOG(ERROR) << device_info_.device.input.sample_rate
-                << " is not a supported input rate.";
-    return false;
-  }
-
   // Create and configure the default audio capturing source.
   SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id_),
                     channel_layout,
                     static_cast<float>(device_info_.device.input.sample_rate));
 
   // Add the capturer to the WebRtcAudioDeviceImpl since it needs some hardware
   // information from the capturer.
   if (audio_device_)
     audio_device_->AddAudioCapturer(this);
 
@@ skipping 389 matching lines @@
 
 void WebRtcAudioCapturer::SetCapturerSourceForTesting(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::AudioParameters params) {
   // Create a new audio stream as source which uses the new source.
   SetCapturerSource(source, params.channel_layout(),
                     static_cast<float>(params.sample_rate()));
 }
 
 }  // namespace content