webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc - Issue 1710103004: webrtc::RtpPacket name freed for better RtpPacket

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

Issue 1710103004: webrtc::RtpPacket name freed for better RtpPacket (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: FecRtpPacket -> test::RawRtpPacket Created 4 years, 10 months ago

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

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

index 1eecd0bca175bcd24a4283e3771c52348ff6c849..32ba26f54b88a5e9eaa6d2cf4d2c785aa857405e 100644

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

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

@@ -29,11 +29,6 @@

namespace webrtc {

enum { REDForFECHeaderLength = 1 };

-struct RtpPacket {

- uint16_t rtpHeaderLength;

- ForwardErrorCorrection::Packet* pkt;

-};

RTPSenderVideo::RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender)

: _rtpSender(*rtpSender),

crit_(CriticalSectionWrapper::CreateCriticalSection()),

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