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}
56 lines
2.0 KiB
C++
56 lines
2.0 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 "call/rtp_rtcp_demuxer_helper.h"
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#include "modules/rtp_rtcp/source/byte_io.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
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namespace webrtc {
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rtc::Optional<uint32_t> ParseRtcpPacketSenderSsrc(
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rtc::ArrayView<const uint8_t> packet) {
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rtcp::CommonHeader header;
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for (const uint8_t* next_packet = packet.begin(); next_packet < packet.end();
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next_packet = header.NextPacket()) {
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if (!header.Parse(next_packet, packet.end() - next_packet)) {
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return rtc::Optional<uint32_t>();
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}
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switch (header.type()) {
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case rtcp::Bye::kPacketType:
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case rtcp::ExtendedReports::kPacketType:
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case rtcp::Psfb::kPacketType:
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case rtcp::ReceiverReport::kPacketType:
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case rtcp::Rtpfb::kPacketType:
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case rtcp::SenderReport::kPacketType: {
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// Sender SSRC at the beginning of the RTCP payload.
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if (header.payload_size_bytes() >= sizeof(uint32_t)) {
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const uint32_t ssrc_sender =
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ByteReader<uint32_t>::ReadBigEndian(header.payload());
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return rtc::Optional<uint32_t>(ssrc_sender);
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} else {
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return rtc::Optional<uint32_t>();
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}
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}
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}
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}
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return rtc::Optional<uint32_t>();
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}
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} // namespace webrtc
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