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/*
* 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 "test/rtp_header_parser.h"
#include <memory>
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "rtc_base/synchronization/mutex.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:
mutable Mutex mutex_;
RtpHeaderExtensionMap rtp_header_extension_map_ RTC_GUARDED_BY(mutex_);
};
std::unique_ptr<RtpHeaderParser> RtpHeaderParser::CreateForTest() {
return std::make_unique<RtpHeaderParserImpl>();
}
RtpHeaderParserImpl::RtpHeaderParserImpl() {}
bool RtpHeaderParser::IsRtcp(const uint8_t* packet, size_t length) {
RtpUtility::RtpHeaderParser rtp_parser(packet, length);
return rtp_parser.RTCP();
}
absl::optional<uint32_t> RtpHeaderParser::GetSsrc(const uint8_t* packet,
size_t length) {
RtpUtility::RtpHeaderParser rtp_parser(packet, length);
RTPHeader header;
if (rtp_parser.Parse(&header, nullptr)) {
return header.ssrc;
}
return absl::nullopt;
}
bool RtpHeaderParserImpl::Parse(const uint8_t* packet,
size_t length,
RTPHeader* header) const {
RtpUtility::RtpHeaderParser rtp_parser(packet, length);
*header = RTPHeader();
RtpHeaderExtensionMap map;
{
MutexLock lock(&mutex_);
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) {
MutexLock lock(&mutex_);
return rtp_header_extension_map_.RegisterByUri(extension.id, extension.uri);
}
bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RTPExtensionType type,
uint8_t id) {
MutexLock lock(&mutex_);
return rtp_header_extension_map_.RegisterByType(id, type);
}
bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RtpExtension extension) {
MutexLock lock(&mutex_);
return rtp_header_extension_map_.Deregister(
rtp_header_extension_map_.GetType(extension.id));
}
bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RTPExtensionType type) {
MutexLock lock(&mutex_);
return rtp_header_extension_map_.Deregister(type) == 0;
}
} // namespace webrtc
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