Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc |
index 20f57460a68ffc1cb30d84b43e46dd2361400c7f..f0c5c99bda93606d3802527dae11c4fb4d006dbb 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc |
@@ -165,6 +165,12 @@ bool RtpReceiverImpl::IncomingRtpPacket( |
UpdateSources(); |
+ auto source = ssrc_sources_.rbegin(); |
danilchap
2017/08/15 08:18:29
though it is not annotated, it seems ssrc_sources_
Zach Stein
2017/08/15 21:44:59
Done.
|
+ source->update_audio_level( |
danilchap
2017/08/15 08:18:29
minor suggestion to use back() instead of rbegin()
Zach Stein
2017/08/15 21:44:59
Done.
|
+ rtp_header.extension.hasAudioLevel |
+ ? rtc::Optional<uint8_t>(rtp_header.extension.audioLevel) |
+ : rtc::Optional<uint8_t>()); |
+ |
size_t payload_data_length = payload_length - rtp_header.paddingLength; |
bool is_first_packet_in_frame = false; |