webrtc/modules/rtp_rtcp/source/rtp_sender.h

/*
 *  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 "rtp_rtcp_config.h"       // misc. defines (e.g. MAX_PACKET_LENGTH)
#include "rtp_rtcp_defines.h"
#include "common_types.h"          // Encryption
#include "ssrc_database.h"
#include "Bitrate.h"
#include "rtp_header_extension.h"
#include "video_codec_information.h"
#include "transmission_bucket.h"

#define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1

namespace webrtc {
class CriticalSectionWrapper;
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(WebRtc_UWord8* dataBuffer,
                                        const WebRtc_UWord16 payloadLength,
                                        const WebRtc_UWord16 rtpHeaderLength,
                                        int64_t capture_time_ms,
                                        const StorageType storage) = 0;
};

class RTPSender : public Bitrate, public RTPSenderInterface
{
public:
    RTPSender(const WebRtc_Word32 id, const bool audio, RtpRtcpClock* clock);
    virtual ~RTPSender();

    void ProcessBitrate();
    void ProcessSendToNetwork();

    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

    // callback
    WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport);

    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;

    WebRtc_Word32 ResetDataCounters();

    WebRtc_UWord32 StartTimestamp() const;
    WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp,
                                    const bool force = false);

    WebRtc_UWord32 GenerateNewSSRC();
    WebRtc_Word32 SetSSRC( const WebRtc_UWord32 ssrc);

    WebRtc_UWord16 SequenceNumber() const;
    WebRtc_Word32 SetSequenceNumber( WebRtc_UWord16 seq);

    WebRtc_Word32 CSRCs(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const;

    WebRtc_Word32 SetCSRCStatus(const bool include);

    WebRtc_Word32 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;

    void UpdateTransmissionTimeOffset(WebRtc_UWord8* rtp_packet,
                                      const WebRtc_UWord16 rtp_packet_length,
                                      const WebRtcRTPHeader& rtp_header,
                                      const WebRtc_Word64 time_diff_ms) const;

    void SetTransmissionSmoothingStatus(const bool enable);

    bool TransmissionSmoothingStatus() const;

    /*
    *    NACK
    */
    int SelectiveRetransmissions() const;
    int SetSelectiveRetransmissions(uint8_t settings);
    void OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
                        const WebRtc_UWord16* nackSequenceNumbers,
                        const WebRtc_UWord16 avgRTT);

    WebRtc_Word32 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(WebRtc_UWord8* dataBuffer,
                                        const WebRtc_UWord16 payloadLength,
                                        const WebRtc_UWord16 rtpHeaderLength,
                                        int64_t capture_time_ms,
                                        const StorageType storage);

    /*
    *    Audio
    */
    WebRtc_Word32 RegisterAudioCallback(RtpAudioFeedback* messagesCallback);

    // 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;

    CriticalSectionWrapper*    _sendCritsect;

    CriticalSectionWrapper*    _transportCritsect;
    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;
    TransmissionBucket        _sendBucket;
    WebRtc_Word64             _timeLastSendToNetworkUpdate;
    bool                      _transmissionSmoothing;

    // 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_