07bf43c673
The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches. In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on. BUG= TEST=vie/voe_auto_test, trybots Review URL: https://webrtc-codereview.appspot.com/1001006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
271 lines
9.8 KiB
C++
271 lines
9.8 KiB
C++
/*
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* Copyright (c) 2012 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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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
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#include <map>
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#include "typedefs.h"
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#include "rtp_utility.h"
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#include "rtp_header_extension.h"
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#include "rtp_rtcp.h"
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#include "rtp_rtcp_defines.h"
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#include "rtcp_receiver_help.h"
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#include "Bitrate.h"
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namespace webrtc {
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class RtpRtcpFeedback;
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class ModuleRtpRtcpImpl;
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class Trace;
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class RTPReceiverAudio;
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class RTPReceiverVideo;
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class RTPReceiverStrategy;
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const WebRtc_Word32 kDefaultVideoFrequency = 90000;
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class RTPReceiver : public Bitrate
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{
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public:
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RTPReceiver(const WebRtc_Word32 id,
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const bool audio,
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RtpRtcpClock* clock,
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ModuleRtpRtcpImpl* owner,
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RtpAudioFeedback* incomingMessagesCallback);
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virtual ~RTPReceiver();
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RtpVideoCodecTypes VideoCodecType() const;
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WebRtc_UWord32 MaxConfiguredBitrate() const;
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WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 timeoutMS);
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void PacketTimeout();
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void ProcessDeadOrAlive(const bool RTCPalive, const WebRtc_Word64 now);
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void ProcessBitrate();
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WebRtc_Word32 RegisterIncomingDataCallback(RtpData* incomingDataCallback);
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WebRtc_Word32 RegisterIncomingRTPCallback(RtpFeedback* incomingMessagesCallback);
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WebRtc_Word32 RegisterReceivePayload(
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 frequency,
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const WebRtc_UWord8 channels,
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const WebRtc_UWord32 rate);
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WebRtc_Word32 DeRegisterReceivePayload(const WebRtc_Word8 payloadType);
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WebRtc_Word32 ReceivePayloadType(
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_UWord32 frequency,
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const WebRtc_UWord8 channels,
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const WebRtc_UWord32 rate,
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WebRtc_Word8* payloadType) const;
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WebRtc_Word32 ReceivePayload(const WebRtc_Word8 payloadType,
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char payloadName[RTP_PAYLOAD_NAME_SIZE],
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WebRtc_UWord32* frequency,
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WebRtc_UWord8* channels,
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WebRtc_UWord32* rate) const;
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WebRtc_Word32 RemotePayload(char payloadName[RTP_PAYLOAD_NAME_SIZE],
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WebRtc_Word8* payloadType,
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WebRtc_UWord32* frequency,
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WebRtc_UWord8* channels) const;
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WebRtc_Word32 IncomingRTPPacket(WebRtcRTPHeader* rtpheader,
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const WebRtc_UWord8* incomingRtpPacket,
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const WebRtc_UWord16 incomingRtpPacketLengt);
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NACKMethod NACK() const ;
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// Turn negative acknowledgement requests on/off
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WebRtc_Word32 SetNACKStatus(const NACKMethod method);
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// last received
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virtual WebRtc_UWord32 TimeStamp() const;
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int32_t LastReceivedTimeMs() const;
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virtual WebRtc_UWord16 SequenceNumber() const;
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WebRtc_Word32 EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const;
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WebRtc_UWord32 SSRC() const;
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WebRtc_Word32 CSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const;
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WebRtc_Word32 Energy( WebRtc_UWord8 arrOfEnergy[kRtpCsrcSize]) const;
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// get the currently configured SSRC filter
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WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowedSSRC) const;
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// set a SSRC to be used as a filter for incoming RTP streams
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WebRtc_Word32 SetSSRCFilter(const bool enable, const WebRtc_UWord32 allowedSSRC);
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WebRtc_Word32 Statistics(WebRtc_UWord8 *fraction_lost,
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WebRtc_UWord32 *cum_lost,
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WebRtc_UWord32 *ext_max,
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WebRtc_UWord32 *jitter, // will be moved from JB
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WebRtc_UWord32 *max_jitter,
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WebRtc_UWord32 *jitter_transmission_time_offset,
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bool reset) const;
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WebRtc_Word32 Statistics(WebRtc_UWord8 *fraction_lost,
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WebRtc_UWord32 *cum_lost,
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WebRtc_UWord32 *ext_max,
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WebRtc_UWord32 *jitter, // will be moved from JB
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WebRtc_UWord32 *max_jitter,
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WebRtc_UWord32 *jitter_transmission_time_offset,
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WebRtc_Word32 *missing,
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bool reset) const;
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WebRtc_Word32 DataCounters(WebRtc_UWord32 *bytesReceived,
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WebRtc_UWord32 *packetsReceived) const;
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WebRtc_Word32 ResetStatistics();
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WebRtc_Word32 ResetDataCounters();
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WebRtc_UWord16 PacketOHReceived() const;
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WebRtc_UWord32 PacketCountReceived() const;
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WebRtc_UWord32 ByteCountReceived() const;
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WebRtc_Word32 RegisterRtpHeaderExtension(const RTPExtensionType type,
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const WebRtc_UWord8 id);
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WebRtc_Word32 DeregisterRtpHeaderExtension(const RTPExtensionType type);
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void GetHeaderExtensionMapCopy(RtpHeaderExtensionMap* map) const;
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virtual WebRtc_UWord32 PayloadTypeToPayload(const WebRtc_UWord8 payloadType,
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ModuleRTPUtility::Payload*& payload) const;
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/*
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* RTX
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*/
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void SetRTXStatus(const bool enable, const WebRtc_UWord32 SSRC);
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void RTXStatus(bool* enable, WebRtc_UWord32* SSRC) const;
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RTPReceiverAudio* GetAudioReceiver() const { return _rtpReceiverAudio; }
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virtual WebRtc_Word32 CallbackOfReceivedPayloadData(
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadSize,
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const WebRtcRTPHeader* rtpHeader);
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virtual WebRtc_Word8 REDPayloadType() const;
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bool HaveNotReceivedPackets() const;
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protected:
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virtual bool RetransmitOfOldPacket(const WebRtc_UWord16 sequenceNumber,
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const WebRtc_UWord32 rtpTimeStamp) const;
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void UpdateStatistics(const WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord16 bytes,
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const bool oldPacket);
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private:
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// Is RED configured with payload type payloadType
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bool REDPayloadType(const WebRtc_Word8 payloadType) const;
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bool InOrderPacket(const WebRtc_UWord16 sequenceNumber) const;
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void CheckSSRCChanged(const WebRtcRTPHeader* rtpHeader);
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void CheckCSRC(const WebRtcRTPHeader* rtpHeader);
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WebRtc_Word32 CheckPayloadChanged(const WebRtcRTPHeader* rtpHeader,
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const WebRtc_Word8 firstPayloadByte,
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bool& isRED,
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ModuleRTPUtility::PayloadUnion* payload);
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void UpdateNACKBitRate(WebRtc_Word32 bytes, WebRtc_UWord32 now);
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bool ProcessNACKBitRate(WebRtc_UWord32 now);
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private:
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RTPReceiverAudio* _rtpReceiverAudio;
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RTPReceiverVideo* _rtpReceiverVideo;
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RTPReceiverStrategy* _rtpMediaReceiver;
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WebRtc_Word32 _id;
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ModuleRtpRtcpImpl& _rtpRtcp;
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CriticalSectionWrapper* _criticalSectionCbs;
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RtpFeedback* _cbRtpFeedback;
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RtpData* _cbRtpData;
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CriticalSectionWrapper* _criticalSectionRTPReceiver;
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mutable WebRtc_Word64 _lastReceiveTime;
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WebRtc_UWord16 _lastReceivedPayloadLength;
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WebRtc_Word8 _lastReceivedPayloadType;
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WebRtc_Word8 _lastReceivedMediaPayloadType;
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WebRtc_UWord32 _packetTimeOutMS;
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WebRtc_Word8 _redPayloadType;
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ModuleRTPUtility::PayloadTypeMap _payloadTypeMap;
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RtpHeaderExtensionMap _rtpHeaderExtensionMap;
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// SSRCs
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WebRtc_UWord32 _SSRC;
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WebRtc_UWord8 _numCSRCs;
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WebRtc_UWord32 _currentRemoteCSRC[kRtpCsrcSize];
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WebRtc_UWord8 _numEnergy;
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WebRtc_UWord8 _currentRemoteEnergy[kRtpCsrcSize];
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bool _useSSRCFilter;
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WebRtc_UWord32 _SSRCFilter;
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// stats on received RTP packets
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WebRtc_UWord32 _jitterQ4;
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mutable WebRtc_UWord32 _jitterMaxQ4;
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mutable WebRtc_UWord32 _cumulativeLoss;
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WebRtc_UWord32 _jitterQ4TransmissionTimeOffset;
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WebRtc_UWord32 _localTimeLastReceivedTimestamp;
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int64_t _lastReceivedFrameTimeMs;
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WebRtc_UWord32 _lastReceivedTimestamp;
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WebRtc_UWord16 _lastReceivedSequenceNumber;
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WebRtc_Word32 _lastReceivedTransmissionTimeOffset;
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WebRtc_UWord16 _receivedSeqFirst;
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WebRtc_UWord16 _receivedSeqMax;
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WebRtc_UWord16 _receivedSeqWraps;
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// current counter values
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WebRtc_UWord16 _receivedPacketOH;
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WebRtc_UWord32 _receivedByteCount;
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WebRtc_UWord32 _receivedOldPacketCount;
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WebRtc_UWord32 _receivedInorderPacketCount;
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// counter values when we sent the last report
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mutable WebRtc_UWord32 _lastReportInorderPackets;
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mutable WebRtc_UWord32 _lastReportOldPackets;
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mutable WebRtc_UWord16 _lastReportSeqMax;
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mutable WebRtc_UWord8 _lastReportFractionLost;
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mutable WebRtc_UWord32 _lastReportCumulativeLost; // 24 bits valid
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mutable WebRtc_UWord32 _lastReportExtendedHighSeqNum;
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mutable WebRtc_UWord32 _lastReportJitter;
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mutable WebRtc_UWord32 _lastReportJitterTransmissionTimeOffset;
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NACKMethod _nackMethod;
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bool _RTX;
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WebRtc_UWord32 _ssrcRTX;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
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