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3e113438b1
One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
88 lines
2.5 KiB
C++
88 lines
2.5 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/rtp_headers.h"
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#include "api/call/transport.h"
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#include "api/rtpparameters.h"
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#include "call/rtp_packet_sink_interface.h"
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#include "common_types.h" // NOLINT(build/include)
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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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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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