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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}
105 lines
3.5 KiB
C++
105 lines
3.5 KiB
C++
/*
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* Copyright (c) 2014 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 "test/rtcp_packet_parser.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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namespace test {
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RtcpPacketParser::RtcpPacketParser() = default;
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RtcpPacketParser::~RtcpPacketParser() = default;
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bool RtcpPacketParser::Parse(const void* data, size_t length) {
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const uint8_t* const buffer = static_cast<const uint8_t*>(data);
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const uint8_t* const buffer_end = buffer + length;
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rtcp::CommonHeader header;
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for (const uint8_t* next_packet = buffer; next_packet != buffer_end;
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next_packet = header.NextPacket()) {
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RTC_DCHECK_GT(buffer_end - next_packet, 0);
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if (!header.Parse(next_packet, buffer_end - next_packet)) {
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LOG(LS_WARNING)
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<< "Invalid rtcp header or unaligned rtcp packet at position "
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<< (next_packet - buffer);
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return false;
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}
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switch (header.type()) {
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case rtcp::App::kPacketType:
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app_.Parse(header);
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break;
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case rtcp::Bye::kPacketType:
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bye_.Parse(header, &sender_ssrc_);
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break;
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case rtcp::ExtendedReports::kPacketType:
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xr_.Parse(header, &sender_ssrc_);
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break;
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case rtcp::ExtendedJitterReport::kPacketType:
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ij_.Parse(header);
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break;
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case rtcp::Psfb::kPacketType:
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switch (header.fmt()) {
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case rtcp::Fir::kFeedbackMessageType:
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fir_.Parse(header, &sender_ssrc_);
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break;
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case rtcp::Pli::kFeedbackMessageType:
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pli_.Parse(header, &sender_ssrc_);
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break;
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case rtcp::Remb::kFeedbackMessageType:
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remb_.Parse(header, &sender_ssrc_);
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break;
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default:
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LOG(LS_WARNING) << "Unknown rtcp payload specific feedback type "
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<< header.fmt();
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break;
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}
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break;
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case rtcp::ReceiverReport::kPacketType:
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receiver_report_.Parse(header, &sender_ssrc_);
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break;
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case rtcp::Rtpfb::kPacketType:
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switch (header.fmt()) {
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case rtcp::Nack::kFeedbackMessageType:
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nack_.Parse(header, &sender_ssrc_);
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break;
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case rtcp::RapidResyncRequest::kFeedbackMessageType:
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rrr_.Parse(header, &sender_ssrc_);
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break;
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case rtcp::Tmmbn::kFeedbackMessageType:
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tmmbn_.Parse(header, &sender_ssrc_);
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break;
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case rtcp::Tmmbr::kFeedbackMessageType:
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tmmbr_.Parse(header, &sender_ssrc_);
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break;
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case rtcp::TransportFeedback::kFeedbackMessageType:
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transport_feedback_.Parse(header, &sender_ssrc_);
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break;
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default:
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LOG(LS_WARNING) << "Unknown rtcp transport feedback type "
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<< header.fmt();
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break;
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}
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break;
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case rtcp::Sdes::kPacketType:
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sdes_.Parse(header);
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break;
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case rtcp::SenderReport::kPacketType:
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sender_report_.Parse(header, &sender_ssrc_);
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break;
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default:
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LOG(LS_WARNING) << "Unknown rtcp packet type " << header.type();
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break;
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}
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}
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return true;
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}
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} // namespace test
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} // namespace webrtc
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