blob: 2f7438f9a48c49ba16b3d9d3b4d7119dc91c0136 [file] [log] [blame]
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <stdint.h>
#include <string>
#include <vector>
#include "api/call/transport.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "call/rtp_packet_sink_interface.h"
namespace webrtc {
class FlexfecReceiveStream : public RtpPacketSinkInterface {
~FlexfecReceiveStream() override = default;
struct Stats {
std::string ToString(int64_t time_ms) const;
// TODO(brandtr): Add appropriate stats here.
int flexfec_bitrate_bps;
struct Config {
explicit Config(Transport* rtcp_send_transport);
Config(const Config&);
std::string ToString() const;
// Returns true if all RTP information is available in order to
// enable receiving FlexFEC.
bool IsCompleteAndEnabled() const;
// Payload type for FlexFEC.
int payload_type = -1;
// SSRC for FlexFEC stream to be received.
uint32_t remote_ssrc = 0;
// Vector containing a single element, corresponding to the SSRC of the
// media stream being protected by this FlexFEC stream. The vector MUST have
// size 1.
// TODO(brandtr): Update comment above when we support multistream
// protection.
std::vector<uint32_t> protected_media_ssrcs;
// SSRC for RTCP reports to be sent.
uint32_t local_ssrc = 0;
// What RTCP mode to use in the reports.
RtcpMode rtcp_mode = RtcpMode::kCompound;
// Transport for outgoing RTCP packets.
Transport* rtcp_send_transport = nullptr;
// |transport_cc| is true whenever the send-side BWE RTCP feedback message
// has been negotiated. This is a prerequisite for enabling send-side BWE.
bool transport_cc = false;
// RTP header extensions that have been negotiated for this track.
std::vector<RtpExtension> rtp_header_extensions;
virtual Stats GetStats() const = 0;
virtual const Config& GetConfig() const = 0;
} // namespace webrtc