Adds proper 44.1 support to 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/string_util.h"
 #include "content/renderer/media/audio_device_factory.h"
@@ skipping 28 matching lines @@
 // Android devices.
 const int kValidOutputRates[] = {48000, 44100, 16000};
 #else
 const int kValidOutputRates[] = {44100};
 #endif
 
 // TODO(xians): Merge the following code to WebRtcAudioCapturer, or remove.
 enum AudioFramesPerBuffer {
   k160,
   k320,
-  k440,  // WebRTC works internally with 440 audio frames at 44.1kHz.
+  k440,
   k480,
   k640,
   k880,
   k960,
   k1440,
   k1920,
   kUnexpectedAudioBufferSize  // Must always be last!
 };
 
 // Helper method to convert integral values to their respective enum values
 // above, or kUnexpectedAudioBufferSize if no match exists.
+// We map 441 to k440 to avoid changes in the XML part for histograms.
+// It is still possible to map the histogram result to the actual buffer size.
+// See http://crbug.com/243450 for details.
 AudioFramesPerBuffer AsAudioFramesPerBuffer(int frames_per_buffer) {
   switch (frames_per_buffer) {
     case 160: return k160;
     case 320: return k320;
-    case 440: return k440;
+    case 441: return k440;
     case 480: return k480;
     case 640: return k640;
     case 880: return k880;
     case 960: return k960;
     case 1440: return k1440;
     case 1920: return k1920;
   }
   return kUnexpectedAudioBufferSize;
 }
 
@@ skipping 64 matching lines @@
   // 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;
-  int buffer_size = 0;
-
-  if (sample_rate % 8000 == 0) {

[ajm, 2013/05/28 15:13:05]

Since webrtc needs 10 ms buffers it only supports sample rates divisible by 100
(e.g. not 22050 or 11025 Hz). Should you still have a check on that here?

[henrika (OOO until Aug 14), 2013/05/28 16:40:30]

You will catch that in the lines above where we ensure that the sample rate is
among the supported rates. I.e., you will never end up here using e.g. 11025.

Sound OK?

[ajm, 2013/05/28 16:43:22]

Sounds good, thanks.

-    buffer_size = (sample_rate / 100);
-  } else if (sample_rate == 44100) {
-    // The resampler in WebRTC does not support 441 as input. We hard code
-    // the size to 440 (~0.9977ms) instead and rely on the internal jitter
-    // buffer in WebRTC to deal with the resulting drift.
-    // TODO(henrika): ensure that WebRTC supports 44100Hz and use 441 instead.
-    buffer_size = 440;
-  } else {
-    return false;
-  }
+  int buffer_size = (sample_rate / 100);
+  DVLOG(1) << "Using WebRTC output buffer size: " << buffer_size;
 
   int channels = ChannelLayoutToChannelCount(channel_layout);
   source_params.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                       channel_layout, channels, 0,
                       sample_rate, 16, buffer_size);
 
   // Set up audio parameters for the sink, i.e., the native audio output stream.
   // We strive to open up using native parameters to achieve best possible
   // performance and to ensure that no FIFO is needed on the browser side to
   // match the client request. Any mismatch between the source and the sink is
@@ skipping 176 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