a5cb98cbbd
This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video. Moving bandwidth estimation before the RTP module is also required for RTX. TEST=vie_auto_test, voe_auto_test, trybots. BUG=1811 R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1545004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
80 lines
2.7 KiB
C++
80 lines
2.7 KiB
C++
/*
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* Copyright (c) 2013 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 "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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namespace webrtc {
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class RtpHeaderParserImpl : public RtpHeaderParser {
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public:
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RtpHeaderParserImpl();
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virtual ~RtpHeaderParserImpl() {}
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virtual bool Parse(const uint8_t* packet, int length,
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RTPHeader* header) const;
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virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
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virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type);
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private:
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scoped_ptr<CriticalSectionWrapper> critical_section_;
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RtpHeaderExtensionMap rtp_header_extension_map_;
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};
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RtpHeaderParser* RtpHeaderParser::Create() {
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return new RtpHeaderParserImpl;
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}
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RtpHeaderParserImpl::RtpHeaderParserImpl()
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: critical_section_(CriticalSectionWrapper::CreateCriticalSection()) {}
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bool RtpHeaderParser::IsRtcp(const uint8_t* packet, int length) {
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ModuleRTPUtility::RTPHeaderParser rtp_parser(packet, length);
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return rtp_parser.RTCP();
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}
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bool RtpHeaderParserImpl::Parse(const uint8_t* packet, int length,
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RTPHeader* header) const {
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ModuleRTPUtility::RTPHeaderParser rtp_parser(packet, length);
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memset(header, 0, sizeof(*header));
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RtpHeaderExtensionMap map;
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{
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CriticalSectionScoped cs(critical_section_.get());
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rtp_header_extension_map_.GetCopy(&map);
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}
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const bool valid_rtpheader = rtp_parser.Parse(*header, &map);
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if (!valid_rtpheader) {
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WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, -1,
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"IncomingPacket invalid RTP header");
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return false;
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}
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return true;
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}
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bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RTPExtensionType type,
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uint8_t id) {
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CriticalSectionScoped cs(critical_section_.get());
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return rtp_header_extension_map_.Register(type, id) == 0;
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}
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bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RTPExtensionType type) {
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CriticalSectionScoped cs(critical_section_.get());
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return rtp_header_extension_map_.Deregister(type) == 0;
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}
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} // namespace webrtc
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