Index: webrtc/modules/audio_coding/codecs/opus/opus_interface.c |
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c |
index e2b4fdbeee79b83eab36120077e5fabf60ede292..d544d850a9307bf5ef75ec4f121030e8656aac8d 100644 |
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c |
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c |
@@ -17,8 +17,13 @@ |
#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 |