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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
77 lines
2.5 KiB
C++
77 lines
2.5 KiB
C++
/*
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* Copyright (c) 2017 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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#include <utility>
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#include "call/rtx_receive_stream.h"
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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RtxReceiveStream::RtxReceiveStream(
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RtpPacketSinkInterface* media_sink,
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std::map<int, int> associated_payload_types,
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uint32_t media_ssrc,
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ReceiveStatistics* rtp_receive_statistics /* = nullptr */)
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: media_sink_(media_sink),
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associated_payload_types_(std::move(associated_payload_types)),
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media_ssrc_(media_ssrc),
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rtp_receive_statistics_(rtp_receive_statistics) {
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if (associated_payload_types_.empty()) {
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LOG(LS_WARNING)
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<< "RtxReceiveStream created with empty payload type mapping.";
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}
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}
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RtxReceiveStream::~RtxReceiveStream() = default;
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void RtxReceiveStream::OnRtpPacket(const RtpPacketReceived& rtx_packet) {
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if (rtp_receive_statistics_) {
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RTPHeader header;
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rtx_packet.GetHeader(&header);
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rtp_receive_statistics_->IncomingPacket(header, rtx_packet.size(),
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false /* retransmitted */);
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}
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rtc::ArrayView<const uint8_t> payload = rtx_packet.payload();
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if (payload.size() < kRtxHeaderSize) {
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return;
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}
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auto it = associated_payload_types_.find(rtx_packet.PayloadType());
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if (it == associated_payload_types_.end()) {
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LOG(LS_VERBOSE) << "Unknown payload type "
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<< static_cast<int>(rtx_packet.PayloadType())
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<< " on rtx ssrc " << rtx_packet.Ssrc();
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return;
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}
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RtpPacketReceived media_packet;
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media_packet.CopyHeaderFrom(rtx_packet);
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media_packet.SetSsrc(media_ssrc_);
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media_packet.SetSequenceNumber((payload[0] << 8) + payload[1]);
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media_packet.SetPayloadType(it->second);
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media_packet.set_recovered(true);
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// Skip the RTX header.
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rtc::ArrayView<const uint8_t> rtx_payload =
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payload.subview(kRtxHeaderSize);
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uint8_t* media_payload = media_packet.AllocatePayload(rtx_payload.size());
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RTC_DCHECK(media_payload != nullptr);
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memcpy(media_payload, rtx_payload.data(), rtx_payload.size());
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media_sink_->OnRtpPacket(media_packet);
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}
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} // namespace webrtc
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