| Index: webrtc/modules/audio_coding/include/audio_coding_module.h
|
| diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
|
| index cd99579e1d5d7365a14746261f61ee05d4965af7..7843fb876de4b38d716f66eebb214a5d55f20845 100644
|
| --- a/webrtc/modules/audio_coding/include/audio_coding_module.h
|
| +++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
|
| @@ -637,7 +637,8 @@ class AudioCodingModule {
|
| //
|
| virtual int SetMaximumPlayoutDelay(int time_ms) = 0;
|
|
|
| - //
|
| + // TODO(kwiberg): Consider if this is needed anymore, now that voe::Channel
|
| + // doesn't use it.
|
| // The shortest latency, in milliseconds, required by jitter buffer. This
|
| // is computed based on inter-arrival times and playout mode of NetEq. The
|
| // actual delay is the maximum of least-required-delay and the minimum-delay
|
|
|