blob: e8d3cbfadbe94b46b5cc6ae7b4009b1401588e5d [file] [log] [blame]
* 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 "webrtc/call/rtp_rtcp_demuxer_helper.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
namespace webrtc {
rtc::Optional<uint32_t> ParseRtcpPacketSenderSsrc(
rtc::ArrayView<const uint8_t> packet) {
rtcp::CommonHeader header;
for (const uint8_t* next_packet = packet.begin(); next_packet < packet.end();
next_packet = header.NextPacket()) {
if (!header.Parse(next_packet, packet.end() - next_packet)) {
return rtc::Optional<uint32_t>();
switch (header.type()) {
case rtcp::Bye::kPacketType:
case rtcp::ExtendedReports::kPacketType:
case rtcp::Psfb::kPacketType:
case rtcp::ReceiverReport::kPacketType:
case rtcp::Rtpfb::kPacketType:
case rtcp::SenderReport::kPacketType: {
// Sender SSRC at the beginning of the RTCP payload.
if (header.payload_size_bytes() >= sizeof(uint32_t)) {
const uint32_t ssrc_sender =
return rtc::Optional<uint32_t>(ssrc_sender);
} else {
return rtc::Optional<uint32_t>();
return rtc::Optional<uint32_t>();
} // namespace webrtc