Modifies size of ring buffer in shared memory on the audio capture side

media/audio/audio_input_device.cc

Index: media/audio/audio_input_device.cc
diff --git a/media/audio/audio_input_device.cc b/media/audio/audio_input_device.cc
index 9f46b55a1b9f365facd573b5f0783cd15f9923af..07eb11c2b77b16c2ba02efda82c5c7491b87693c 100644
--- a/media/audio/audio_input_device.cc
+++ b/media/audio/audio_input_device.cc
@@ -17,7 +17,13 @@ namespace media {
 // in order to avoid data overwriting. This number can be any positive number,
 // dependent how fast the renderer process can pick up captured data from
 // shared memory.
-static const int kRequestedSharedMemoryCount = 10;
+// TODO(henrika): figure out a suitable size of this ring buffer.
+// We have seen reports in Chrome where segments of repeated input audio has
+// damaged AEC performance in WebRTC clients. By setting its value to 1, we
+// reduce the number of places in Chrome where such a patteren could possibly
+// be created. The original value of kRequestedSharedMemoryCount was 10.
+// See b/13976602 for details.
+static const int kRequestedSharedMemoryCount = 1;
 
 // Takes care of invoking the capture callback on the audio thread.
 // An instance of this class is created for each capture stream in