mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00

Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
92 lines
3 KiB
C++
92 lines
3 KiB
C++
/*
|
|
* 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/critical_section.h"
|
|
#include "rtc_base/thread_annotations.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class RtpHeaderParserImpl : public RtpHeaderParser {
|
|
public:
|
|
RtpHeaderParserImpl();
|
|
~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 RegisterRtpHeaderExtension(RtpExtension extension) override;
|
|
|
|
bool DeregisterRtpHeaderExtension(RTPExtensionType type) override;
|
|
bool DeregisterRtpHeaderExtension(RtpExtension extension) override;
|
|
|
|
private:
|
|
rtc::CriticalSection critical_section_;
|
|
RtpHeaderExtensionMap rtp_header_extension_map_
|
|
RTC_GUARDED_BY(critical_section_);
|
|
};
|
|
|
|
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(RtpExtension extension) {
|
|
rtc::CritScope cs(&critical_section_);
|
|
return rtp_header_extension_map_.RegisterByUri(extension.id, extension.uri);
|
|
}
|
|
|
|
bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RTPExtensionType type,
|
|
uint8_t id) {
|
|
rtc::CritScope cs(&critical_section_);
|
|
return rtp_header_extension_map_.RegisterByType(id, type);
|
|
}
|
|
|
|
bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RtpExtension extension) {
|
|
rtc::CritScope cs(&critical_section_);
|
|
return rtp_header_extension_map_.Deregister(
|
|
rtp_header_extension_map_.GetType(extension.id));
|
|
}
|
|
|
|
bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RTPExtensionType type) {
|
|
rtc::CritScope cs(&critical_section_);
|
|
return rtp_header_extension_map_.Deregister(type) == 0;
|
|
}
|
|
} // namespace webrtc
|