Add support for 24kHz audio sample rate and remove the validation check

in webrtc. Our resampler is capable of handling any sample rates.

This has been reviewed here:
https://codereview.chromium.org/548503002/

BUG=411241
TBR=xians,dalecurtis
NOTRY=true

Review URL: https://codereview.chromium.org/554733002

Cr-Commit-Position: refs/heads/master@{#293773}

content/renderer/media/webrtc_audio_capturer.cc

@@ -23,22 +23,6 @@ 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;
@@ -196,17 +180,6 @@ bool WebRtcAudioCapturer::Initialize() {
                          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,

content/renderer/media/webrtc_audio_renderer.cc

@@ -29,24 +29,6 @@ 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
@@ -254,16 +236,6 @@ bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) {
                          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

media/audio/sample_rates.cc

@@ -44,6 +44,9 @@ bool ToAudioSampleRate(int sample_rate, AudioSampleRate* asr) {
     case 192000:
       *asr = k192000Hz;
       return true;
+    case 24000:
+      *asr = k24000Hz;
+      return true;
   }
   return false;
 }

media/audio/sample_rates.h

@@ -23,8 +23,9 @@ enum AudioSampleRate {
   k88200Hz = 8,
   k176400Hz = 9,
   k192000Hz = 10,
+  k24000Hz = 11,
   // Must always equal the largest value ever reported:
-  kAudioSampleRateMax = k192000Hz,
+  kAudioSampleRateMax = k24000Hz,
 };
 
 // Helper method to convert integral values to their respective enum values,