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 a0dafec022db8c55150d706b2761294cf191e87e..898de3466124c3a605274466a1749437e5f3a265 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 |