Index: webrtc/modules/audio_coding/codecs/audio_encoder.h |
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h |
index c913765fe48fd87ddc9fd45c7eb534dee016c014..8584683576a93bc950e1c6c42978edf74ca0f7ed 100644 |
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h |
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h |
@@ -181,6 +181,13 @@ class AudioEncoder { |
// Provides RTT to this encoder to allow it to adapt. |
virtual void OnReceivedRtt(int rtt_ms); |
+ // Provides overhead to this encoder to adapt. The overhead is the number of |
+ // bytes that will be add to each payload the encoder generates. With this |
kwiberg-webrtc
2016/12/03 00:30:56
"add" -> "added"
And maybe "payload" -> "packet"?
minyue-webrtc
2016/12/06 09:04:19
Done.
|
+ // and the total bandwidth and target bitrate provided by |
+ // |OnReceivedUplinkBandwidth| and |OnReceivedTargetAudioBitrate| |
+ // respectively, the encoder can decide its frame length and payload rate. |
kwiberg-webrtc
2016/12/03 00:30:56
Aren't the round-trip time and the packet loss rat
minyue-webrtc
2016/12/06 09:04:19
Round trip time is not taken into account right no
kwiberg-webrtc
2016/12/06 09:43:25
Acknowledged.
|
+ virtual void OnReceivedOverhead(size_t overhead_bytes_per_packet); |
+ |
// To allow encoder to adapt its frame length, it must be provided the frame |
// length range that receivers can accept. |
virtual void SetReceiverFrameLengthRange(int min_frame_length_ms, |