Bug 1247574: Force webrtc audio input processing to resample to target rate to fix 16KHz mics. r=padenot

MozReview-Commit-ID: BBZcX03Z6Kn

media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_core_win.cc

@@ -2236,7 +2236,7 @@ int32_t AudioDeviceWindowsCore::InitPlayout()
     Wfx.wBitsPerSample = 16;
     Wfx.cbSize = 0;
 
-    const int freqs[] = {48000, 44100, 16000, 96000, 32000, 8000};
+    const int freqs[] = {48000, 44100, 32000, 96000, 16000, 8000};
     hr = S_FALSE;
 
     // Iterate over frequencies and channels, in order of priority
@@ -2573,7 +2573,7 @@ int32_t AudioDeviceWindowsCore::InitRecording()
     Wfx.wBitsPerSample = 16;
     Wfx.cbSize = 0;
 
-    const int freqs[6] = {48000, 44100, 16000, 96000, 32000, 8000};
+    const int freqs[6] = {48000, 44100, 32000, 96000, 16000, 8000};
     hr = S_FALSE;
 
     // Iterate over frequencies and channels, in order of priority

media/webrtc/trunk/webrtc/voice_engine/utility.cc

@@ -86,7 +86,11 @@ void DownConvertToCodecFormat(const int16_t* src_data,
 
   // Never upsample the capture signal here. This should be done at the
   // end of the send chain.
-  int destination_rate = std::min(codec_rate_hz, sample_rate_hz);
+  // XXX bug 1247574 temporary hack until we switch to full-duplex
+  // We need to know the final audio rate before starting the audio channels,
+  // and this means we can get called back in Process() with the input
+  // rate if it's less than the codec rate.
+  int destination_rate = codec_rate_hz;
 
   // If no stereo codecs are in use, we downmix a stereo stream from the
   // device early in the chain, before resampling.