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/*
* 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.
*/
#include "call/rtx_receive_stream.h"
#include <string.h>
#include <utility>
#include "api/array_view.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
RtxReceiveStream::RtxReceiveStream(
RtpPacketSinkInterface* media_sink,
std::map<int, int> associated_payload_types,
uint32_t media_ssrc,
ReceiveStatistics* rtp_receive_statistics /* = nullptr */)
: media_sink_(media_sink),
associated_payload_types_(std::move(associated_payload_types)),
media_ssrc_(media_ssrc),
rtp_receive_statistics_(rtp_receive_statistics) {
packet_checker_.Detach();
if (associated_payload_types_.empty()) {
RTC_LOG(LS_WARNING)
<< "RtxReceiveStream created with empty payload type mapping.";
}
}
RtxReceiveStream::~RtxReceiveStream() = default;
void RtxReceiveStream::SetAssociatedPayloadTypes(
std::map<int, int> associated_payload_types) {
RTC_DCHECK_RUN_ON(&packet_checker_);
associated_payload_types_ = std::move(associated_payload_types);
}
void RtxReceiveStream::OnRtpPacket(const RtpPacketReceived& rtx_packet) {
RTC_DCHECK_RUN_ON(&packet_checker_);
if (rtp_receive_statistics_) {
rtp_receive_statistics_->OnRtpPacket(rtx_packet);
}
rtc::ArrayView<const uint8_t> payload = rtx_packet.payload();
if (payload.size() < kRtxHeaderSize) {
return;
}
auto it = associated_payload_types_.find(rtx_packet.PayloadType());
if (it == associated_payload_types_.end()) {
RTC_DLOG(LS_VERBOSE) << "Unknown payload type "
<< static_cast<int>(rtx_packet.PayloadType())
<< " on rtx ssrc " << rtx_packet.Ssrc();
return;
}
RtpPacketReceived media_packet;
media_packet.CopyHeaderFrom(rtx_packet);
media_packet.SetSsrc(media_ssrc_);
media_packet.SetSequenceNumber((payload[0] << 8) + payload[1]);
media_packet.SetPayloadType(it->second);
media_packet.set_recovered(true);
media_packet.set_arrival_time(rtx_packet.arrival_time());
// Skip the RTX header.
rtc::ArrayView<const uint8_t> rtx_payload = payload.subview(kRtxHeaderSize);
uint8_t* media_payload = media_packet.AllocatePayload(rtx_payload.size());
RTC_DCHECK(media_payload != nullptr);
memcpy(media_payload, rtx_payload.data(), rtx_payload.size());
media_sink_->OnRtpPacket(media_packet);
}
} // namespace webrtc
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