webrtc/api/rtpsender.h - Issue 1788583004: Enable setting the maximum bitrate limit in RtpSender.

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Unified Diff: webrtc/api/rtpsender.h

Issue 1788583004: Enable setting the maximum bitrate limit in RtpSender. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Removed support for bitrate limits for audio streams; corrected code review issues Created 4 years, 9 months ago

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Index: webrtc/api/rtpsender.h

diff --git a/webrtc/api/rtpsender.h b/webrtc/api/rtpsender.h

index 25cc4ed14a97bc8f73595c1f544d0dc6c1072bd4..879a332209b3bdef484da7094178005b36286107 100644

--- a/webrtc/api/rtpsender.h

+++ b/webrtc/api/rtpsender.h

@@ -97,6 +97,9 @@ class AudioRtpSender : public ObserverInterface,

void Stop() override;

+ RtpParameters GetParameters() const;

+ bool SetParameters(const RtpParameters& parameters);

private:

bool can_send_track() const { return track_ && ssrc_; }

// Helper function to construct options for

@@ -158,6 +161,9 @@ class VideoRtpSender : public ObserverInterface,