Delete RtpVideoStreamReceiver methods GetRtpReceiver and rtp_rtcp

Replaced by new method GetSyncInfo.

Bug: webrtc:7135
Change-Id: I541567a5ca173dc334fd85e83f15b25a3120b8aa
Reviewed-on: https://webrtc-review.googlesource.com/91123
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24148}
This commit is contained in:
Niels Möller
2018-07-31 08:29:53 +02:00
committed by Commit Bot
parent 168b4977f3
commit df9e9ae9ee
3 changed files with 24 additions and 19 deletions

View File

@ -18,6 +18,7 @@
#include <vector>
#include "call/rtp_packet_sink_interface.h"
#include "call/syncable.h"
#include "call/video_receive_stream.h"
#include "modules/include/module_common_types.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
@ -75,12 +76,13 @@ class RtpVideoStreamReceiver : public RtpData,
bool AddReceiveCodec(const VideoCodec& video_codec,
const std::map<std::string, std::string>& codec_params);
RtpReceiver* GetRtpReceiver() const;
RtpRtcp* rtp_rtcp() const { return rtp_rtcp_.get(); }
void StartReceive();
void StopReceive();
// Produces the transport-related timestamps; current_delay_ms is left unset.
absl::optional<Syncable::Info> GetSyncInfo() const;
bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length);
void FrameContinuous(int64_t seq_num);