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
347 lines
12 KiB
C++
347 lines
12 KiB
C++
/*
|
|
* Copyright (c) 2012 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.
|
|
*/
|
|
|
|
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
|
|
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
|
|
|
|
#include <cassert>
|
|
#include <cmath>
|
|
#include <map>
|
|
|
|
#include "webrtc/common_types.h"
|
|
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
|
|
|
|
#define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1
|
|
|
|
namespace webrtc {
|
|
class CriticalSectionWrapper;
|
|
class PacedSender;
|
|
class RTPPacketHistory;
|
|
class RTPSenderAudio;
|
|
class RTPSenderVideo;
|
|
|
|
class RTPSenderInterface {
|
|
public:
|
|
RTPSenderInterface() {}
|
|
virtual ~RTPSenderInterface() {}
|
|
|
|
virtual WebRtc_UWord32 SSRC() const = 0;
|
|
virtual WebRtc_UWord32 Timestamp() const = 0;
|
|
|
|
virtual WebRtc_Word32 BuildRTPheader(WebRtc_UWord8* dataBuffer,
|
|
const WebRtc_Word8 payloadType,
|
|
const bool markerBit,
|
|
const WebRtc_UWord32 captureTimeStamp,
|
|
const bool timeStampProvided = true,
|
|
const bool incSequenceNumber = true) = 0;
|
|
|
|
virtual WebRtc_UWord16 RTPHeaderLength() const = 0;
|
|
virtual WebRtc_UWord16 IncrementSequenceNumber() = 0;
|
|
virtual WebRtc_UWord16 SequenceNumber() const = 0;
|
|
virtual WebRtc_UWord16 MaxPayloadLength() const = 0;
|
|
virtual WebRtc_UWord16 MaxDataPayloadLength() const = 0;
|
|
virtual WebRtc_UWord16 PacketOverHead() const = 0;
|
|
virtual WebRtc_UWord16 ActualSendBitrateKbit() const = 0;
|
|
|
|
virtual WebRtc_Word32 SendToNetwork(uint8_t* data_buffer,
|
|
int payload_length,
|
|
int rtp_header_length,
|
|
int64_t capture_time_ms,
|
|
StorageType storage) = 0;
|
|
};
|
|
|
|
class RTPSender : public Bitrate, public RTPSenderInterface {
|
|
public:
|
|
RTPSender(const WebRtc_Word32 id,
|
|
const bool audio,
|
|
Clock* clock,
|
|
Transport* transport,
|
|
RtpAudioFeedback* audio_feedback,
|
|
PacedSender* paced_sender);
|
|
virtual ~RTPSender();
|
|
|
|
void ProcessBitrate();
|
|
|
|
WebRtc_UWord16 ActualSendBitrateKbit() const;
|
|
|
|
WebRtc_UWord32 VideoBitrateSent() const;
|
|
WebRtc_UWord32 FecOverheadRate() const;
|
|
WebRtc_UWord32 NackOverheadRate() const;
|
|
|
|
void SetTargetSendBitrate(const WebRtc_UWord32 bits);
|
|
|
|
WebRtc_UWord16 MaxDataPayloadLength() const; // with RTP and FEC headers
|
|
|
|
WebRtc_Word32 RegisterPayload(
|
|
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
|
|
const WebRtc_Word8 payloadType,
|
|
const WebRtc_UWord32 frequency,
|
|
const WebRtc_UWord8 channels,
|
|
const WebRtc_UWord32 rate);
|
|
|
|
WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType);
|
|
|
|
WebRtc_Word8 SendPayloadType() const;
|
|
|
|
int SendPayloadFrequency() const;
|
|
|
|
void SetSendingStatus(const bool enabled);
|
|
|
|
void SetSendingMediaStatus(const bool enabled);
|
|
bool SendingMedia() const;
|
|
|
|
// number of sent RTP packets
|
|
WebRtc_UWord32 Packets() const;
|
|
|
|
// number of sent RTP bytes
|
|
WebRtc_UWord32 Bytes() const;
|
|
|
|
void ResetDataCounters();
|
|
|
|
WebRtc_UWord32 StartTimestamp() const;
|
|
void SetStartTimestamp(WebRtc_UWord32 timestamp, bool force);
|
|
|
|
WebRtc_UWord32 GenerateNewSSRC();
|
|
void SetSSRC(const WebRtc_UWord32 ssrc);
|
|
|
|
WebRtc_UWord16 SequenceNumber() const;
|
|
void SetSequenceNumber(WebRtc_UWord16 seq);
|
|
|
|
WebRtc_Word32 CSRCs(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const;
|
|
|
|
void SetCSRCStatus(const bool include);
|
|
|
|
void SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
|
|
const WebRtc_UWord8 arrLength);
|
|
|
|
WebRtc_Word32 SetMaxPayloadLength(const WebRtc_UWord16 length,
|
|
const WebRtc_UWord16 packetOverHead);
|
|
|
|
WebRtc_Word32 SendOutgoingData(const FrameType frameType,
|
|
const WebRtc_Word8 payloadType,
|
|
const WebRtc_UWord32 timeStamp,
|
|
int64_t capture_time_ms,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord32 payloadSize,
|
|
const RTPFragmentationHeader* fragmentation,
|
|
VideoCodecInformation* codecInfo = NULL,
|
|
const RTPVideoTypeHeader* rtpTypeHdr = NULL);
|
|
|
|
WebRtc_Word32 SendPadData(WebRtc_Word8 payload_type,
|
|
WebRtc_UWord32 capture_timestamp,
|
|
int64_t capture_time_ms,
|
|
WebRtc_Word32 bytes);
|
|
/*
|
|
* RTP header extension
|
|
*/
|
|
WebRtc_Word32 SetTransmissionTimeOffset(
|
|
const WebRtc_Word32 transmissionTimeOffset);
|
|
|
|
WebRtc_Word32 RegisterRtpHeaderExtension(const RTPExtensionType type,
|
|
const WebRtc_UWord8 id);
|
|
|
|
WebRtc_Word32 DeregisterRtpHeaderExtension(const RTPExtensionType type);
|
|
|
|
WebRtc_UWord16 RtpHeaderExtensionTotalLength() const;
|
|
|
|
WebRtc_UWord16 BuildRTPHeaderExtension(WebRtc_UWord8* dataBuffer) const;
|
|
|
|
WebRtc_UWord8 BuildTransmissionTimeOffsetExtension(
|
|
WebRtc_UWord8* dataBuffer) const;
|
|
|
|
bool UpdateTransmissionTimeOffset(WebRtc_UWord8* rtp_packet,
|
|
const WebRtc_UWord16 rtp_packet_length,
|
|
const WebRtcRTPHeader& rtp_header,
|
|
const WebRtc_Word64 time_diff_ms) const;
|
|
|
|
void TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms);
|
|
|
|
/*
|
|
* NACK
|
|
*/
|
|
int SelectiveRetransmissions() const;
|
|
int SetSelectiveRetransmissions(uint8_t settings);
|
|
void OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
|
|
const WebRtc_UWord16* nackSequenceNumbers,
|
|
const WebRtc_UWord16 avgRTT);
|
|
|
|
void SetStorePacketsStatus(const bool enable,
|
|
const WebRtc_UWord16 numberToStore);
|
|
|
|
bool StorePackets() const;
|
|
|
|
WebRtc_Word32 ReSendPacket(WebRtc_UWord16 packet_id,
|
|
WebRtc_UWord32 min_resend_time = 0);
|
|
|
|
WebRtc_Word32 ReSendToNetwork(const WebRtc_UWord8* packet,
|
|
const WebRtc_UWord32 size);
|
|
|
|
bool ProcessNACKBitRate(const WebRtc_UWord32 now);
|
|
|
|
/*
|
|
* RTX
|
|
*/
|
|
void SetRTXStatus(const bool enable,
|
|
const bool setSSRC,
|
|
const WebRtc_UWord32 SSRC);
|
|
|
|
void RTXStatus(bool* enable, WebRtc_UWord32* SSRC) const;
|
|
|
|
/*
|
|
* Functions wrapping RTPSenderInterface
|
|
*/
|
|
virtual WebRtc_Word32 BuildRTPheader(WebRtc_UWord8* dataBuffer,
|
|
const WebRtc_Word8 payloadType,
|
|
const bool markerBit,
|
|
const WebRtc_UWord32 captureTimeStamp,
|
|
const bool timeStampProvided = true,
|
|
const bool incSequenceNumber = true);
|
|
|
|
virtual WebRtc_UWord16 RTPHeaderLength() const ;
|
|
virtual WebRtc_UWord16 IncrementSequenceNumber();
|
|
virtual WebRtc_UWord16 MaxPayloadLength() const;
|
|
virtual WebRtc_UWord16 PacketOverHead() const;
|
|
|
|
// current timestamp
|
|
virtual WebRtc_UWord32 Timestamp() const;
|
|
virtual WebRtc_UWord32 SSRC() const;
|
|
|
|
virtual WebRtc_Word32 SendToNetwork(uint8_t* data_buffer,
|
|
int payload_length,
|
|
int rtp_header_length,
|
|
int64_t capture_time_ms,
|
|
StorageType storage);
|
|
/*
|
|
* Audio
|
|
*/
|
|
// Send a DTMF tone using RFC 2833 (4733)
|
|
WebRtc_Word32 SendTelephoneEvent(const WebRtc_UWord8 key,
|
|
const WebRtc_UWord16 time_ms,
|
|
const WebRtc_UWord8 level);
|
|
|
|
bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const;
|
|
|
|
// Set audio packet size, used to determine when it's time to send a DTMF
|
|
// packet in silence (CNG)
|
|
WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples);
|
|
|
|
// Set status and ID for header-extension-for-audio-level-indication.
|
|
WebRtc_Word32 SetAudioLevelIndicationStatus(const bool enable,
|
|
const WebRtc_UWord8 ID);
|
|
|
|
// Get status and ID for header-extension-for-audio-level-indication.
|
|
WebRtc_Word32 AudioLevelIndicationStatus(bool& enable,
|
|
WebRtc_UWord8& ID) const;
|
|
|
|
// Store the audio level in dBov for
|
|
// header-extension-for-audio-level-indication.
|
|
WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov);
|
|
|
|
// Set payload type for Redundant Audio Data RFC 2198
|
|
WebRtc_Word32 SetRED(const WebRtc_Word8 payloadType);
|
|
|
|
// Get payload type for Redundant Audio Data RFC 2198
|
|
WebRtc_Word32 RED(WebRtc_Word8& payloadType) const;
|
|
|
|
/*
|
|
* Video
|
|
*/
|
|
VideoCodecInformation* CodecInformationVideo();
|
|
|
|
RtpVideoCodecTypes VideoCodecType() const;
|
|
|
|
WebRtc_UWord32 MaxConfiguredBitrateVideo() const;
|
|
|
|
WebRtc_Word32 SendRTPIntraRequest();
|
|
|
|
// FEC
|
|
WebRtc_Word32 SetGenericFECStatus(const bool enable,
|
|
const WebRtc_UWord8 payloadTypeRED,
|
|
const WebRtc_UWord8 payloadTypeFEC);
|
|
|
|
WebRtc_Word32 GenericFECStatus(bool& enable,
|
|
WebRtc_UWord8& payloadTypeRED,
|
|
WebRtc_UWord8& payloadTypeFEC) const;
|
|
|
|
WebRtc_Word32 SetFecParameters(
|
|
const FecProtectionParams* delta_params,
|
|
const FecProtectionParams* key_params);
|
|
|
|
protected:
|
|
WebRtc_Word32 CheckPayloadType(const WebRtc_Word8 payloadType,
|
|
RtpVideoCodecTypes& videoType);
|
|
|
|
private:
|
|
void UpdateNACKBitRate(const WebRtc_UWord32 bytes,
|
|
const WebRtc_UWord32 now);
|
|
|
|
WebRtc_Word32 SendPaddingAccordingToBitrate(
|
|
WebRtc_Word8 payload_type,
|
|
WebRtc_UWord32 capture_timestamp,
|
|
int64_t capture_time_ms);
|
|
|
|
WebRtc_Word32 _id;
|
|
const bool _audioConfigured;
|
|
RTPSenderAudio* _audio;
|
|
RTPSenderVideo* _video;
|
|
|
|
PacedSender* paced_sender_;
|
|
CriticalSectionWrapper* _sendCritsect;
|
|
|
|
Transport* _transport;
|
|
bool _sendingMedia;
|
|
|
|
WebRtc_UWord16 _maxPayloadLength;
|
|
WebRtc_UWord16 _targetSendBitrate;
|
|
WebRtc_UWord16 _packetOverHead;
|
|
|
|
WebRtc_Word8 _payloadType;
|
|
std::map<WebRtc_Word8, ModuleRTPUtility::Payload*> _payloadTypeMap;
|
|
|
|
RtpHeaderExtensionMap _rtpHeaderExtensionMap;
|
|
WebRtc_Word32 _transmissionTimeOffset;
|
|
|
|
// NACK
|
|
WebRtc_UWord32 _nackByteCountTimes[NACK_BYTECOUNT_SIZE];
|
|
WebRtc_Word32 _nackByteCount[NACK_BYTECOUNT_SIZE];
|
|
Bitrate _nackBitrate;
|
|
|
|
RTPPacketHistory* _packetHistory;
|
|
|
|
// Statistics
|
|
WebRtc_UWord32 _packetsSent;
|
|
WebRtc_UWord32 _payloadBytesSent;
|
|
|
|
// RTP variables
|
|
bool _startTimeStampForced;
|
|
WebRtc_UWord32 _startTimeStamp;
|
|
SSRCDatabase& _ssrcDB;
|
|
WebRtc_UWord32 _remoteSSRC;
|
|
bool _sequenceNumberForced;
|
|
WebRtc_UWord16 _sequenceNumber;
|
|
WebRtc_UWord16 _sequenceNumberRTX;
|
|
bool _ssrcForced;
|
|
WebRtc_UWord32 _ssrc;
|
|
WebRtc_UWord32 _timeStamp;
|
|
WebRtc_UWord8 _CSRCs;
|
|
WebRtc_UWord32 _CSRC[kRtpCsrcSize];
|
|
bool _includeCSRCs;
|
|
bool _RTX;
|
|
WebRtc_UWord32 _ssrcRTX;
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
|