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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender.h

Issue 1803923003: Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 9 months ago
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Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 5b6d4f7af7409f3db6309f2c6d589c480ea06009..b376c380d05919af17a8107be25eb665a27020a8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -90,7 +90,6 @@ class RTPSender : public RTPSenderInterface {
RTPSender(bool audio,
Clock* clock,
Transport* transport,
- RtpAudioFeedback* audio_feedback,
RtpPacketSender* paced_sender,
TransportSequenceNumberAllocator* sequence_number_allocator,
TransportFeedbackObserver* transport_feedback_callback,
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