Adding build switch for Opus that supports 120ms ptime.

webrtc/modules/audio_coding/codecs/opus/opus_interface.c

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
 
 #include <stdlib.h>
 #include <string.h>
 
 enum {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+  /* Maximum supported frame size in WebRTC is 120 ms. */
+  kWebRtcOpusMaxEncodeFrameSizeMs = 120,
+#else
   /* Maximum supported frame size in WebRTC is 60 ms. */
   kWebRtcOpusMaxEncodeFrameSizeMs = 60,
+#endif
 
   /* The format allows up to 120 ms frames. Since we don't control the other
    * side, we must allow for packets of that size. NetEq is currently limited
    * to 60 ms on the receive side. */
   kWebRtcOpusMaxDecodeFrameSizeMs = 120,
 
   /* Maximum sample count per channel is 48 kHz * maximum frame size in
    * milliseconds. */
   kWebRtcOpusMaxFrameSizePerChannel = 48 * kWebRtcOpusMaxDecodeFrameSizeMs,
 
@@ skipping 437 matching lines @@
     return 0;
   }
 
   for (n = 0; n < channels; n++) {
     if (frame_data[0][0] & (0x80 >> ((n + 1) * (frames + 1) - 1)))
       return 1;
   }
 
   return 0;
 }