blob: bc05033efafd5673d21cc469d88b443715d75fda [file] [log] [blame]
* Copyright (c) 2013 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 "modules/rtp_rtcp/include/rtp_header_parser.h"
#include <string.h>
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class RtpHeaderParserImpl : public RtpHeaderParser {
~RtpHeaderParserImpl() override = default;
bool Parse(const uint8_t* packet,
size_t length,
RTPHeader* header) const override;
bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id) override;
bool DeregisterRtpHeaderExtension(RTPExtensionType type) override;
rtc::CriticalSection critical_section_;
RtpHeaderExtensionMap rtp_header_extension_map_
RtpHeaderParser* RtpHeaderParser::Create() {
return new RtpHeaderParserImpl;
RtpHeaderParserImpl::RtpHeaderParserImpl() {}
bool RtpHeaderParser::IsRtcp(const uint8_t* packet, size_t length) {
RtpUtility::RtpHeaderParser rtp_parser(packet, length);
return rtp_parser.RTCP();
bool RtpHeaderParserImpl::Parse(const uint8_t* packet,
size_t length,
RTPHeader* header) const {
RtpUtility::RtpHeaderParser rtp_parser(packet, length);
memset(header, 0, sizeof(*header));
RtpHeaderExtensionMap map;
rtc::CritScope cs(&critical_section_);
map = rtp_header_extension_map_;
const bool valid_rtpheader = rtp_parser.Parse(header, &map);
if (!valid_rtpheader) {
return false;
return true;
bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RTPExtensionType type,
uint8_t id) {
rtc::CritScope cs(&critical_section_);
return rtp_header_extension_map_.RegisterByType(id, type);
bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RTPExtensionType type) {
rtc::CritScope cs(&critical_section_);
return rtp_header_extension_map_.Deregister(type) == 0;
} // namespace webrtc