20ed36dada
The goal with this new clock interface is to have something which is used all over WebRTC to make it easier to switch clock implementation depending on where the components are used. This is a first step in that direction. Next steps will be to, step by step, move all modules, video engine and voice engine over to the new interface, effectively deprecating the old clock interfaces. Long-term my vision is that we should be able to deprecate the clock of WebRTC and rely on the user providing the implementation. TEST=vie_auto_test, rtp_rtcp_unittests, trybots Review URL: https://webrtc-codereview.appspot.com/1041004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
530 lines
18 KiB
C++
530 lines
18 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_RTCP_IMPL_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
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#include <list>
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#include <vector>
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#ifdef MATLAB
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class MatlabPlot;
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#endif
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namespace webrtc {
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class ModuleRtpRtcpImpl : public RtpRtcp {
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public:
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explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
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virtual ~ModuleRtpRtcpImpl();
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// Returns the number of milliseconds until the module want a worker thread to
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// call Process.
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virtual WebRtc_Word32 TimeUntilNextProcess();
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// Process any pending tasks such as timeouts.
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virtual WebRtc_Word32 Process();
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// Receiver part.
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// Configure a timeout value.
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virtual WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 rtp_timeout_ms,
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const WebRtc_UWord32 rtcp_timeout_ms);
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// Set periodic dead or alive notification.
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virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus(
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const bool enable,
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const WebRtc_UWord8 sample_time_seconds);
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// Get periodic dead or alive notification status.
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virtual WebRtc_Word32 PeriodicDeadOrAliveStatus(
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bool& enable,
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WebRtc_UWord8& sample_time_seconds);
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virtual WebRtc_Word32 RegisterReceivePayload(const CodecInst& voice_codec);
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virtual WebRtc_Word32 RegisterReceivePayload(const VideoCodec& video_codec);
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virtual WebRtc_Word32 ReceivePayloadType(const CodecInst& voice_codec,
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WebRtc_Word8* pl_type);
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virtual WebRtc_Word32 ReceivePayloadType(const VideoCodec& video_codec,
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WebRtc_Word8* pl_type);
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virtual WebRtc_Word32 DeRegisterReceivePayload(
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const WebRtc_Word8 payload_type);
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// Register RTP header extension.
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virtual WebRtc_Word32 RegisterReceiveRtpHeaderExtension(
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const RTPExtensionType type,
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const WebRtc_UWord8 id);
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virtual WebRtc_Word32 DeregisterReceiveRtpHeaderExtension(
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const RTPExtensionType type);
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// Get the currently configured SSRC filter.
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virtual WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowed_ssrc) const;
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// Set a SSRC to be used as a filter for incoming RTP streams.
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virtual WebRtc_Word32 SetSSRCFilter(const bool enable,
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const WebRtc_UWord32 allowed_ssrc);
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// Get last received remote timestamp.
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virtual WebRtc_UWord32 RemoteTimestamp() const;
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// Get the local time of the last received remote timestamp.
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virtual int64_t LocalTimeOfRemoteTimeStamp() const;
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// Get the current estimated remote timestamp.
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virtual WebRtc_Word32 EstimatedRemoteTimeStamp(
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WebRtc_UWord32& timestamp) const;
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virtual WebRtc_UWord32 RemoteSSRC() const;
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virtual WebRtc_Word32 RemoteCSRCs(
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WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const;
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virtual WebRtc_Word32 SetRTXReceiveStatus(const bool enable,
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const WebRtc_UWord32 ssrc);
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virtual WebRtc_Word32 RTXReceiveStatus(bool* enable,
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WebRtc_UWord32* ssrc) const;
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// Called by the network module when we receive a packet.
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virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incoming_packet,
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const WebRtc_UWord16 packet_length);
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// Sender part.
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virtual WebRtc_Word32 RegisterSendPayload(const CodecInst& voice_codec);
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virtual WebRtc_Word32 RegisterSendPayload(const VideoCodec& video_codec);
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virtual WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payload_type);
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virtual WebRtc_Word8 SendPayloadType() const;
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// Register RTP header extension.
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virtual WebRtc_Word32 RegisterSendRtpHeaderExtension(
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const RTPExtensionType type,
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const WebRtc_UWord8 id);
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virtual WebRtc_Word32 DeregisterSendRtpHeaderExtension(
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const RTPExtensionType type);
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// Get start timestamp.
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virtual WebRtc_UWord32 StartTimestamp() const;
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// Configure start timestamp, default is a random number.
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virtual WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp);
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virtual WebRtc_UWord16 SequenceNumber() const;
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// Set SequenceNumber, default is a random number.
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virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq);
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virtual WebRtc_UWord32 SSRC() const;
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// Configure SSRC, default is a random number.
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virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc);
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virtual WebRtc_Word32 CSRCs(WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const;
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virtual WebRtc_Word32 SetCSRCs(const WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize],
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const WebRtc_UWord8 arr_length);
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virtual WebRtc_Word32 SetCSRCStatus(const bool include);
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virtual WebRtc_UWord32 PacketCountSent() const;
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virtual int CurrentSendFrequencyHz() const;
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virtual WebRtc_UWord32 ByteCountSent() const;
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virtual WebRtc_Word32 SetRTXSendStatus(const bool enable,
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const bool set_ssrc,
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const WebRtc_UWord32 ssrc);
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virtual WebRtc_Word32 RTXSendStatus(bool* enable,
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WebRtc_UWord32* ssrc) const;
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// Sends kRtcpByeCode when going from true to false.
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virtual WebRtc_Word32 SetSendingStatus(const bool sending);
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virtual bool Sending() const;
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// Drops or relays media packets.
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virtual WebRtc_Word32 SetSendingMediaStatus(const bool sending);
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virtual bool SendingMedia() const;
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// Used by the codec module to deliver a video or audio frame for
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// packetization.
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virtual WebRtc_Word32 SendOutgoingData(
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const FrameType frame_type,
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const WebRtc_Word8 payload_type,
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const WebRtc_UWord32 time_stamp,
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int64_t capture_time_ms,
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const WebRtc_UWord8* payload_data,
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const WebRtc_UWord32 payload_size,
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const RTPFragmentationHeader* fragmentation = NULL,
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const RTPVideoHeader* rtp_video_hdr = NULL);
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virtual void TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number,
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int64_t capture_time_ms);
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// RTCP part.
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// Get RTCP status.
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virtual RTCPMethod RTCP() const;
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// Configure RTCP status i.e on/off.
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virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
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// Set RTCP CName.
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virtual WebRtc_Word32 SetCNAME(const char c_name[RTCP_CNAME_SIZE]);
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// Get RTCP CName.
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virtual WebRtc_Word32 CNAME(char c_name[RTCP_CNAME_SIZE]);
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// Get remote CName.
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virtual WebRtc_Word32 RemoteCNAME(const WebRtc_UWord32 remote_ssrc,
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char c_name[RTCP_CNAME_SIZE]) const;
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// Get remote NTP.
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virtual WebRtc_Word32 RemoteNTP(WebRtc_UWord32* received_ntp_secs,
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WebRtc_UWord32* received_ntp_frac,
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WebRtc_UWord32* rtcp_arrival_time_secs,
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WebRtc_UWord32* rtcp_arrival_time_frac,
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WebRtc_UWord32* rtcp_timestamp) const;
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virtual WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 ssrc,
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const char c_name[RTCP_CNAME_SIZE]);
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virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 ssrc);
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// Get RoundTripTime.
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virtual WebRtc_Word32 RTT(const WebRtc_UWord32 remote_ssrc,
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WebRtc_UWord16* rtt,
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WebRtc_UWord16* avg_rtt,
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WebRtc_UWord16* min_rtt,
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WebRtc_UWord16* max_rtt) const;
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// Reset RoundTripTime statistics.
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virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remote_ssrc);
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virtual void SetRtt(uint32_t rtt);
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// Force a send of an RTCP packet.
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// Normal SR and RR are triggered via the process function.
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virtual WebRtc_Word32 SendRTCP(WebRtc_UWord32 rtcp_packet_type = kRtcpReport);
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// Statistics of our locally created statistics of the received RTP stream.
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virtual WebRtc_Word32 StatisticsRTP(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,
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WebRtc_UWord32* max_jitter = NULL) const;
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// Reset RTP statistics.
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virtual WebRtc_Word32 ResetStatisticsRTP();
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virtual WebRtc_Word32 ResetReceiveDataCountersRTP();
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virtual WebRtc_Word32 ResetSendDataCountersRTP();
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// Statistics of the amount of data sent and received.
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virtual WebRtc_Word32 DataCountersRTP(WebRtc_UWord32* bytes_sent,
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WebRtc_UWord32* packets_sent,
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WebRtc_UWord32* bytes_received,
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WebRtc_UWord32* packets_received) const;
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virtual WebRtc_Word32 ReportBlockStatistics(
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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,
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WebRtc_UWord32* jitter_transmission_time_offset);
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// Get received RTCP report, sender info.
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virtual WebRtc_Word32 RemoteRTCPStat(RTCPSenderInfo* sender_info);
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// Get received RTCP report, report block.
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virtual WebRtc_Word32 RemoteRTCPStat(
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std::vector<RTCPReportBlock>* receive_blocks) const;
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// Set received RTCP report block.
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virtual WebRtc_Word32 AddRTCPReportBlock(
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const WebRtc_UWord32 ssrc, const RTCPReportBlock* receive_block);
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virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 ssrc);
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// (REMB) Receiver Estimated Max Bitrate.
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virtual bool REMB() const;
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virtual WebRtc_Word32 SetREMBStatus(const bool enable);
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virtual WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate,
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const WebRtc_UWord8 number_of_ssrc,
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const WebRtc_UWord32* ssrc);
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// (IJ) Extended jitter report.
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virtual bool IJ() const;
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virtual WebRtc_Word32 SetIJStatus(const bool enable);
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// (TMMBR) Temporary Max Media Bit Rate.
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virtual bool TMMBR() const;
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virtual WebRtc_Word32 SetTMMBRStatus(const bool enable);
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WebRtc_Word32 SetTMMBN(const TMMBRSet* bounding_set);
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virtual WebRtc_UWord16 MaxPayloadLength() const;
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virtual WebRtc_UWord16 MaxDataPayloadLength() const;
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virtual WebRtc_Word32 SetMaxTransferUnit(const WebRtc_UWord16 size);
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virtual WebRtc_Word32 SetTransportOverhead(
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const bool tcp,
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const bool ipv6,
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const WebRtc_UWord8 authentication_overhead = 0);
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// (NACK) Negative acknowledgment part.
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// Is Negative acknowledgment requests on/off?
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virtual NACKMethod NACK() const;
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// Turn negative acknowledgment requests on/off.
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virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method);
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virtual int SelectiveRetransmissions() const;
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virtual int SetSelectiveRetransmissions(uint8_t settings);
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// Send a Negative acknowledgment packet.
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virtual WebRtc_Word32 SendNACK(const WebRtc_UWord16* nack_list,
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const WebRtc_UWord16 size);
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// Store the sent packets, needed to answer to a negative acknowledgment
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// requests.
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virtual WebRtc_Word32 SetStorePacketsStatus(
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const bool enable, const WebRtc_UWord16 number_to_store = 200);
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// (APP) Application specific data.
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virtual WebRtc_Word32 SetRTCPApplicationSpecificData(
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const WebRtc_UWord8 sub_type,
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const WebRtc_UWord32 name,
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const WebRtc_UWord8* data,
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const WebRtc_UWord16 length);
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// (XR) VOIP metric.
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virtual WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
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// Audio part.
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// Set audio packet size, used to determine when it's time to send a DTMF
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// packet in silence (CNG).
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virtual WebRtc_Word32 SetAudioPacketSize(
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const WebRtc_UWord16 packet_size_samples);
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// Outband DTMF detection.
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virtual WebRtc_Word32 SetTelephoneEventStatus(
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const bool enable,
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const bool forward_to_decoder,
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const bool detect_end_of_tone = false);
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// Is outband DTMF turned on/off?
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virtual bool TelephoneEvent() const;
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// Is forwarding of outband telephone events turned on/off?
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virtual bool TelephoneEventForwardToDecoder() const;
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virtual bool SendTelephoneEventActive(WebRtc_Word8& telephone_event) const;
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// Send a TelephoneEvent tone using RFC 2833 (4733).
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virtual WebRtc_Word32 SendTelephoneEventOutband(const WebRtc_UWord8 key,
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const WebRtc_UWord16 time_ms,
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const WebRtc_UWord8 level);
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// Set payload type for Redundant Audio Data RFC 2198.
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virtual WebRtc_Word32 SetSendREDPayloadType(const WebRtc_Word8 payload_type);
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// Get payload type for Redundant Audio Data RFC 2198.
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virtual WebRtc_Word32 SendREDPayloadType(WebRtc_Word8& payload_type) const;
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// Set status and id for header-extension-for-audio-level-indication.
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virtual WebRtc_Word32 SetRTPAudioLevelIndicationStatus(
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const bool enable, const WebRtc_UWord8 id);
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// Get status and id for header-extension-for-audio-level-indication.
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virtual WebRtc_Word32 GetRTPAudioLevelIndicationStatus(
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bool& enable, WebRtc_UWord8& id) const;
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// Store the audio level in d_bov for header-extension-for-audio-level-
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// indication.
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virtual WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_d_bov);
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// Video part.
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virtual RtpVideoCodecTypes ReceivedVideoCodec() const;
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virtual RtpVideoCodecTypes SendVideoCodec() const;
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virtual WebRtc_Word32 SendRTCPSliceLossIndication(
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const WebRtc_UWord8 picture_id);
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// Set method for requestion a new key frame.
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virtual WebRtc_Word32 SetKeyFrameRequestMethod(
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const KeyFrameRequestMethod method);
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// Send a request for a keyframe.
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virtual WebRtc_Word32 RequestKeyFrame();
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virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delay_ms);
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virtual void SetTargetSendBitrate(const WebRtc_UWord32 bitrate);
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virtual WebRtc_Word32 SetGenericFECStatus(
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const bool enable,
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const WebRtc_UWord8 payload_type_red,
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const WebRtc_UWord8 payload_type_fec);
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virtual WebRtc_Word32 GenericFECStatus(
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bool& enable,
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WebRtc_UWord8& payload_type_red,
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WebRtc_UWord8& payload_type_fec);
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virtual WebRtc_Word32 SetFecParameters(
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const FecProtectionParams* delta_params,
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const FecProtectionParams* key_params);
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virtual WebRtc_Word32 LastReceivedNTP(WebRtc_UWord32& NTPsecs,
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WebRtc_UWord32& NTPfrac,
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WebRtc_UWord32& remote_sr);
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virtual WebRtc_Word32 BoundingSet(bool& tmmbr_owner,
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TMMBRSet*& bounding_set_rec);
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virtual void BitrateSent(WebRtc_UWord32* total_rate,
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WebRtc_UWord32* video_rate,
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WebRtc_UWord32* fec_rate,
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WebRtc_UWord32* nackRate) const;
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virtual int EstimatedReceiveBandwidth(
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WebRtc_UWord32* available_bandwidth) const;
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virtual void SetRemoteSSRC(const WebRtc_UWord32 ssrc);
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virtual WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 send_report);
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// Good state of RTP receiver inform sender.
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virtual WebRtc_Word32 SendRTCPReferencePictureSelection(
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const WebRtc_UWord64 picture_id);
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void OnReceivedTMMBR();
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// Bad state of RTP receiver request a keyframe.
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void OnRequestIntraFrame();
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// Received a request for a new SLI.
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void OnReceivedSliceLossIndication(const WebRtc_UWord8 picture_id);
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// Received a new reference frame.
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void OnReceivedReferencePictureSelectionIndication(
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const WebRtc_UWord64 picture_id);
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void OnReceivedNACK(const WebRtc_UWord16 nack_sequence_numbers_length,
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const WebRtc_UWord16* nack_sequence_numbers);
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void OnRequestSendReport();
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// Following function is only called when constructing the object so no
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// need to worry about data race.
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void OwnsClock() {
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owns_clock_ = true;
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}
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protected:
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void RegisterChildModule(RtpRtcp* module);
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void DeRegisterChildModule(RtpRtcp* module);
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bool UpdateRTCPReceiveInformationTimers();
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void ProcessDeadOrAliveTimer();
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WebRtc_UWord32 BitrateReceivedNow() const;
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// Get remote SequenceNumber.
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WebRtc_UWord16 RemoteSequenceNumber() const;
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// Only for internal testing.
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WebRtc_UWord32 LastSendReport(WebRtc_UWord32& last_rtcptime);
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RTPSender rtp_sender_;
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RTPReceiver rtp_receiver_;
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RTCPSender rtcp_sender_;
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RTCPReceiver rtcp_receiver_;
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bool owns_clock_;
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Clock& clock_;
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private:
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int64_t RtcpReportInterval();
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WebRtc_Word32 id_;
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const bool audio_;
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bool collision_detected_;
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WebRtc_Word64 last_process_time_;
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WebRtc_Word64 last_bitrate_process_time_;
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WebRtc_Word64 last_packet_timeout_process_time_;
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WebRtc_UWord16 packet_overhead_;
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scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_;
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scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_feedback_;
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ModuleRtpRtcpImpl* default_module_;
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std::list<ModuleRtpRtcpImpl*> child_modules_;
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// Dead or alive.
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bool dead_or_alive_active_;
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WebRtc_UWord32 dead_or_alive_timeout_ms_;
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WebRtc_Word64 dead_or_alive_last_timer_;
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// Send side
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NACKMethod nack_method_;
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WebRtc_UWord32 nack_last_time_sent_;
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WebRtc_UWord16 nack_last_seq_number_sent_;
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bool simulcast_;
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VideoCodec send_video_codec_;
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KeyFrameRequestMethod key_frame_req_method_;
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RemoteBitrateEstimator* remote_bitrate_;
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RtcpRttObserver* rtt_observer_;
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#ifdef MATLAB
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MatlabPlot* plot1_;
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#endif
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
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