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

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/strings/string_util.h"
 #include "base/strings/stringprintf.h"
@@ skipping 11 matching lines @@
 
 #if defined(OS_WIN)
 #include "base/win/windows_version.h"
 #include "media/audio/win/core_audio_util_win.h"
 #endif
 
 namespace content {
 
 namespace {
 
-// Supported hardware sample rates for output sides.
-#if defined(OS_WIN) || defined(OS_MACOSX)
-// AudioHardwareConfig::GetOutputSampleRate() 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 Initialize() will fail if the user selects
-// any rate outside these ranges.
-const int kValidOutputRates[] = {96000, 48000, 44100, 32000, 16000};
-#elif defined(OS_LINUX) || defined(OS_OPENBSD)
-const int kValidOutputRates[] = {48000, 44100};
-#elif defined(OS_ANDROID)
-// TODO(leozwang): We want to use native sampling rate on Android to achieve
-// low latency, currently 16000 is used to work around audio problem on some
-// Android devices.
-const int kValidOutputRates[] = {48000, 44100, 16000};
-#else
-const int kValidOutputRates[] = {44100};
-#endif
-
 // This is a simple wrapper class that's handed out to users of a shared
 // WebRtcAudioRenderer instance.  This class maintains the per-user 'playing'
 // and 'started' states to avoid problems related to incorrect usage which
 // might violate the implementation assumptions inside WebRtcAudioRenderer
 // (see the play reference count).
 class SharedAudioRenderer : public MediaStreamAudioRenderer {
  public:
   // Callback definition for a callback that is called when when Play(), Pause()
   // or SetVolume are called (whenever the internal |playing_state_| changes).
   typedef base::Callback<
@@ skipping 187 matching lines @@
   }
   media::AudioSampleRate asr;
   if (media::ToAudioSampleRate(sample_rate, &asr)) {
     UMA_HISTOGRAM_ENUMERATION(
         "WebRTC.AudioOutputSampleRate", asr, media::kAudioSampleRateMax + 1);
   } else {
     UMA_HISTOGRAM_COUNTS("WebRTC.AudioOutputSampleRateUnexpected",
                          sample_rate);
   }
 
-  // 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.";
-    return false;
-  }
-
   // Set up audio parameters for the source, i.e., the WebRTC client.
 
   // The WebRTC client only supports multiples of 10ms as buffer size where
   // 10ms is preferred for lowest possible delay.
   media::AudioParameters source_params;
   const int frames_per_10ms = (sample_rate / 100);
   DVLOG(1) << "Using WebRTC output buffer size: " << frames_per_10ms;
 
   source_params.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                       sink_params_.channel_layout(), sink_params_.channels(),
@@ skipping 289 matching lines @@
       if (RemovePlayingState(source, state))
         EnterPauseState();
     } else if (AddPlayingState(source, state)) {
       EnterPlayState();
     }
     UpdateSourceVolume(source);
   }
 }
 
 }  // namespace content