| /* |
| * 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. |
| */ |
| |
| #ifndef WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_H_ |
| #define WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_H_ |
| |
| #include <stdint.h> |
| |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/api/call/transport.h" |
| #include "webrtc/api/rtpparameters.h" |
| #include "webrtc/call/rtp_packet_sink_interface.h" |
| #include "webrtc/common_types.h" |
| |
| namespace webrtc { |
| |
| class FlexfecReceiveStream : public RtpPacketSinkInterface { |
| public: |
| ~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) |
| : rtcp_send_transport(rtcp_send_transport) { |
| RTC_DCHECK(rtcp_send_transport); |
| } |
| |
| 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 |
| |
| #endif // WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_H_ |