From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/call/rtx_receive_stream.h | 59 +++++++++++++++++++++++++ 1 file changed, 59 insertions(+) create mode 100644 third_party/libwebrtc/call/rtx_receive_stream.h (limited to 'third_party/libwebrtc/call/rtx_receive_stream.h') diff --git a/third_party/libwebrtc/call/rtx_receive_stream.h b/third_party/libwebrtc/call/rtx_receive_stream.h new file mode 100644 index 0000000000..79b03d306b --- /dev/null +++ b/third_party/libwebrtc/call/rtx_receive_stream.h @@ -0,0 +1,59 @@ +/* + * Copyright (c) 2017 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 CALL_RTX_RECEIVE_STREAM_H_ +#define CALL_RTX_RECEIVE_STREAM_H_ + +#include +#include + +#include "api/sequence_checker.h" +#include "call/rtp_packet_sink_interface.h" +#include "rtc_base/system/no_unique_address.h" + +namespace webrtc { + +class ReceiveStatistics; + +// This class is responsible for RTX decapsulation. The resulting media packets +// are passed on to a sink representing the associated media stream. +class RtxReceiveStream : public RtpPacketSinkInterface { + public: + RtxReceiveStream(RtpPacketSinkInterface* media_sink, + std::map associated_payload_types, + uint32_t media_ssrc, + // TODO(nisse): Delete this argument, and + // corresponding member variable, by moving the + // responsibility for rtcp feedback to + // RtpStreamReceiverController. + ReceiveStatistics* rtp_receive_statistics = nullptr); + ~RtxReceiveStream() override; + + // Update payload types post construction. Must be called from the same + // calling context as `OnRtpPacket` is called on. + void SetAssociatedPayloadTypes(std::map associated_payload_types); + + // RtpPacketSinkInterface. + void OnRtpPacket(const RtpPacketReceived& packet) override; + + private: + RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_checker_; + RtpPacketSinkInterface* const media_sink_; + // Map from rtx payload type -> media payload type. + std::map associated_payload_types_ RTC_GUARDED_BY(&packet_checker_); + // TODO(nisse): Ultimately, the media receive stream shouldn't care about the + // ssrc, and we should delete this. + const uint32_t media_ssrc_; + ReceiveStatistics* const rtp_receive_statistics_; +}; + +} // namespace webrtc + +#endif // CALL_RTX_RECEIVE_STREAM_H_ -- cgit v1.2.3