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988cc0870b
This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
85 lines
2.4 KiB
C++
85 lines
2.4 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 CALL_FLEXFEC_RECEIVE_STREAM_H_
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#define 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 "api/call/transport.h"
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#include "api/rtp_headers.h"
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#include "api/rtpparameters.h"
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#include "call/rtp_packet_sink_interface.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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Config(const Config&);
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~Config();
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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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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 // CALL_FLEXFEC_RECEIVE_STREAM_H_
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