webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc - Issue 1539423003: [rtp_rtcp] Lint errors cleaned from rtp_utility

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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

Issue 1539423003: [rtp_rtcp] Lint errors cleaned from rtp_utility (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Created 5 years ago

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Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

index 86407f9f40617555c4d2ba9162795f590311d9ee..d361443cee3ddbb498fcc24fe1e3f34bd0b816af 100644

--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

@@ -350,7 +350,7 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,

size_t packetSize = payloadSize + rtpHeaderLength;

RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);

RTPHeader rtp_header;

- rtp_parser.Parse(rtp_header);

+ rtp_parser.Parse(&rtp_header);

_rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,

(frameType == kAudioFrameSpeech),

audio_level_dbov);