70e4053844
Reason for revert:
Unexpected perf regressions.
Original issue's description:
> Parse FlexFEC RTP headers in Call and add integration with BWE.
>
> BUG=webrtc:5654
>
> Review-Url: https://codereview.webrtc.org/2553863003
> Cr-Commit-Position: refs/heads/master@{#15709}
> Committed: ab2ffa3b28
TBR=philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2589393002
Cr-Commit-Position: refs/heads/master@{#15727}
81 lines
2.3 KiB
C++
81 lines
2.3 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_H_
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#define WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_H_
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#include <string>
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#include <vector>
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#include "webrtc/api/call/transport.h"
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#include "webrtc/config.h"
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namespace webrtc {
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class FlexfecReceiveStream {
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public:
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struct Stats {
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std::string ToString(int64_t time_ms) const;
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// TODO(brandtr): Add appropriate stats here.
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int flexfec_bitrate_bps;
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};
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struct Config {
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std::string ToString() const;
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// Payload type for FlexFEC.
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int payload_type = -1;
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// SSRC for FlexFEC stream to be received.
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uint32_t remote_ssrc = 0;
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// Vector containing a single element, corresponding to the SSRC of the
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// media stream being protected by this FlexFEC stream. The vector MUST have
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// size 1.
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//
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// TODO(brandtr): Update comment above when we support multistream
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// protection.
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std::vector<uint32_t> protected_media_ssrcs;
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// SSRC for RTCP reports to be sent.
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uint32_t local_ssrc = 0;
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// What RTCP mode to use in the reports.
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RtcpMode rtcp_mode = RtcpMode::kCompound;
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// Transport for outgoing RTCP packets.
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Transport* rtcp_send_transport = nullptr;
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// |transport_cc| is true whenever the send-side BWE RTCP feedback message
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// has been negotiated. This is a prerequisite for enabling send-side BWE.
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bool transport_cc = false;
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// RTP header extensions that have been negotiated for this track.
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std::vector<RtpExtension> extensions;
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};
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// Starts stream activity.
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// When a stream is active, it can receive and process packets.
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virtual void Start() = 0;
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// Stops stream activity.
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// When a stream is stopped, it can't receive nor process packets.
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virtual void Stop() = 0;
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virtual Stats GetStats() const = 0;
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protected:
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virtual ~FlexfecReceiveStream() = default;
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};
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} // namespace webrtc
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#endif // WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_H_
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