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Unified Diff: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h

Issue 1864313003: Move Ownership of RtpModules to VideoSendStream. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 4 years, 8 months ago
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Index: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
index d01465b9f8fd3f6625a7f6612bb839558e9d48f4..50fb35e0e4fb17102cc0b639e805ca2c6d9f03a9 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -210,10 +210,10 @@ class RtpRtcp : public Module {
*/
virtual void SetSequenceNumber(uint16_t seq) = 0;
- // Returns true if the ssrc matched this module, false otherwise.
- virtual bool SetRtpStateForSsrc(uint32_t ssrc,
- const RtpState& rtp_state) = 0;
- virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0;
+ virtual void SetRtpState(const RtpState& rtp_state) = 0;
+ virtual void SetRtxState(const RtpState& rtp_state) = 0;
+ virtual RtpState GetRtpState() const = 0;
+ virtual RtpState GetRtxState() const = 0;
/*
* Get SSRC
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