webrtc/modules/rtp_rtcp/source/rtp_sender.cc - Issue 1803923003: Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.

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

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.cc

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

index 3197d60f92ecf9248b0ba7fd1e2dc5bba237c0e6..36d7eb573cbe43db51fe13a25183f41d0a974923 100644

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

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

@@ -108,7 +108,6 @@ RTPSender::RTPSender(

bool audio,

Clock* clock,

Transport* transport,

- RtpAudioFeedback* audio_feedback,

RtpPacketSender* paced_sender,

TransportSequenceNumberAllocator* sequence_number_allocator,

TransportFeedbackObserver* transport_feedback_observer,

@@ -125,7 +124,7 @@ RTPSender::RTPSender(

bitrates_(bitrate_callback),

total_bitrate_sent_(clock, bitrates_.total_bitrate_observer()),

audio_configured_(audio),

- audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr),

+ audio_(audio ? new RTPSenderAudio(clock, this) : nullptr),

video_(audio ? nullptr : new RTPSenderVideo(clock, this)),

paced_sender_(paced_sender),

transport_sequence_number_allocator_(sequence_number_allocator),

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