Chromium Code Reviews| Index: webrtc/voice_engine/channel.h |
| diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h |
| index 01907d269b1fa2eab5ce5396a1e09f16ce3e326b..a4ba8d7bd253f0ec67d0f5df0278cf5dd9e3dfcb 100644 |
| --- a/webrtc/voice_engine/channel.h |
| +++ b/webrtc/voice_engine/channel.h |
| @@ -443,6 +443,8 @@ class Channel |
| RTPExtensionType type, |
| unsigned char id); |
| + void UpdateOverheadForEncoder(); |
| + |
| int GetRtpTimestampRateHz() const; |
| int64_t GetRTT(bool allow_associate_channel) const; |
| @@ -529,6 +531,8 @@ class Channel |
| uint32_t _lastLocalTimeStamp; |
| int8_t _lastPayloadType; |
| bool _includeAudioLevelIndication; |
| + int _transport_overhead_per_packet; |
|
the sun
2016/12/21 10:33:02
How about we stick to size_t here as well?
nisse-webrtc
2017/01/09 16:02:24
Done (for both this one and _rtp_overhead_per_pack
|
| + int _rtp_overhead_per_packet; |
| // VoENetwork |
| AudioFrame::SpeechType _outputSpeechType; |
| // VoEVideoSync |