Adds proper 44.1 support to WebRTC

content/renderer/media/webrtc_audio_renderer.cc

Index: content/renderer/media/webrtc_audio_renderer.cc
===================================================================
--- content/renderer/media/webrtc_audio_renderer.cc	(revision 202522)
+++ content/renderer/media/webrtc_audio_renderer.cc	(working copy)
@@ -46,7 +46,7 @@
 enum AudioFramesPerBuffer {
   k160,
   k320,
-  k440,  // WebRTC works internally with 440 audio frames at 44.1kHz.
+  k440,
   k480,
   k640,
   k880,
@@ -58,11 +58,14 @@
 
 // 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;
@@ -147,24 +150,13 @@
   }
 
   // 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;
+  int buffer_size = (sample_rate / 100);
+  DVLOG(1) << "Using WebRTC output buffer size: " << buffer_size;
 
-  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 channels = ChannelLayoutToChannelCount(channel_layout);
   source_params.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                       channel_layout, channels, 0,