c0d481a4a6
In preparation of making RTP packet demuxing many-to-one (one SSRC goes to one sink, but one sink may have multiple SSRCs), we need to remove FlexFEC's dependence on being able to register itself with the demuxer. Instead, we register FlexFEC streams with the streams they protect; when those streams get packets, they'll forward them to their associated FlexFEC streams, too. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2974453002 Cr-Commit-Position: refs/heads/master@{#19219}
94 lines
2.7 KiB
C++
94 lines
2.7 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 <stdint.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/call/rtp_packet_sink_interface.h"
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#include "webrtc/config.h"
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namespace webrtc {
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class FlexfecReceiveStream : public RtpPacketSinkInterface {
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public:
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~FlexfecReceiveStream() override = default;
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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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explicit Config(Transport* rtcp_send_transport)
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: rtcp_send_transport(rtcp_send_transport) {
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RTC_DCHECK(rtcp_send_transport);
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}
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std::string ToString() const;
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// Returns true if all RTP information is available in order to
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// enable receiving FlexFEC.
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bool IsCompleteAndEnabled() 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> rtp_header_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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virtual const Config& GetConfig() const = 0;
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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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