Adding three frame sizes to Opus

Adding support for 10, 40 and 60 ms packet sizes for Opus.

BUG=issue1015

Review URL: https://webrtc-codereview.appspot.com/1086004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3454 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/main/source/acm_codec_database.cc

@@ -286,7 +286,7 @@ const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
     // Opus supports frames shorter than 10ms,
     // but it doesn't help us to use them.
     // Mono and stereo.
-    {1, {960}, 0, 2},
+    {4, {480, 960, 1920, 2880}, 0, 2},
 #endif
 #ifdef WEBRTC_CODEC_SPEEX
     {3, {160, 320, 480}, 0, 1},

webrtc/modules/audio_coding/main/source/acm_common_defs.h

@@ -36,8 +36,8 @@ namespace webrtc {
 
 // 60 ms is the maximum block size we support. An extra 20 ms is considered
 // for safety if process() method is not called when it should be, i.e. we
-// accept 20 ms of jitter. 80 ms @ 32 kHz (super wide-band) is 2560 samples.
-#define AUDIO_BUFFER_SIZE_W16  2560
+// accept 20 ms of jitter. 80 ms @ 48 kHz (full-band) stereo is 7680 samples.
+#define AUDIO_BUFFER_SIZE_W16  7680
 
 // There is one timestamp per each 10 ms of audio
 // the audio buffer, at max, may contain 32 blocks of 10ms

webrtc/modules/audio_coding/main/source/acm_opus.cc

@@ -108,7 +108,8 @@ ACMOpus::ACMOpus(int16_t codec_id)
       bitrate_(20000),  // Default bit-rate.
       channels_(1) {  // Default mono
   codec_id_ = codec_id;
-  // Opus has internal DTX, but we dont use it for now.
+
+  // Opus has internal DTX, but we don't use it for now.
   has_internal_dtx_ = false;
 
   if (codec_id_ != ACMCodecDB::kOpus) {