modules/rtp_rtcp/source/rtp_sender.h - src/webrtc - Git at Google

 /*
  *  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 *data_buffer, const WebRtc_Word8 payload_type,
       const bool marker_bit, const WebRtc_UWord32 capture_time_stamp,
       const bool time_stamp_provided = true,
       const bool inc_sequence_number = 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 payload_name[RTP_PAYLOAD_NAME_SIZE],
       const WebRtc_Word8 payload_type, const WebRtc_UWord32 frequency,
       const WebRtc_UWord8 channels, const WebRtc_UWord32 rate);

   WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payload_type);

   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 arr_of_csrc[kRtpCsrcSize]) const;

   void SetCSRCStatus(const bool include);

   void SetCSRCs(const WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize],
                 const WebRtc_UWord8 arr_length);

   WebRtc_Word32 SetMaxPayloadLength(const WebRtc_UWord16 length,
                                     const WebRtc_UWord16 packet_over_head);

   WebRtc_Word32 SendOutgoingData(
       const FrameType frame_type, const WebRtc_Word8 payload_type,
       const WebRtc_UWord32 time_stamp, int64_t capture_time_ms,
       const WebRtc_UWord8 *payload_data, const WebRtc_UWord32 payload_size,
       const RTPFragmentationHeader *fragmentation,
       VideoCodecInformation *codec_info = NULL,
       const RTPVideoTypeHeader * rtp_type_hdr = 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 transmission_time_offset);

   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 *data_buffer) const;

   WebRtc_UWord8 BuildTransmissionTimeOffsetExtension(
       WebRtc_UWord8 *data_buffer) 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 std::list<uint16_t>& nack_sequence_numbers,
                       const WebRtc_UWord16 avg_rtt);

   void SetStorePacketsStatus(const bool enable,
                              const WebRtc_UWord16 number_to_store);

   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 RtxMode mode, const bool set_ssrc,
                     const WebRtc_UWord32 SSRC);

   void RTXStatus(RtxMode* mode, WebRtc_UWord32 *SSRC) const;

   // Functions wrapping RTPSenderInterface.
   virtual WebRtc_Word32 BuildRTPheader(
       WebRtc_UWord8 *data_buffer, const WebRtc_Word8 payload_type,
       const bool marker_bit, const WebRtc_UWord32 capture_time_stamp,
       const bool time_stamp_provided = true,
       const bool inc_sequence_number = 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 *telephone_event) 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 packet_size_samples);

   // 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 d_bov for
   // header-extension-for-audio-level-indication.
   WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_d_bov);

   // Set payload type for Redundant Audio Data RFC 2198.
   WebRtc_Word32 SetRED(const WebRtc_Word8 payload_type);

   // Get payload type for Redundant Audio Data RFC 2198.
   WebRtc_Word32 RED(WebRtc_Word8 *payload_type) const;

   // Video.
   VideoCodecInformation *CodecInformationVideo();

   RtpVideoCodecTypes VideoCodecType() const;

   WebRtc_UWord32 MaxConfiguredBitrateVideo() const;

   WebRtc_Word32 SendRTPIntraRequest();

   // FEC.
   WebRtc_Word32 SetGenericFECStatus(const bool enable,
                                     const WebRtc_UWord8 payload_type_red,
                                     const WebRtc_UWord8 payload_type_fec);

   WebRtc_Word32 GenericFECStatus(bool *enable, WebRtc_UWord8 *payload_type_red,
                                  WebRtc_UWord8 *payload_type_fec) const;

   WebRtc_Word32 SetFecParameters(const FecProtectionParams *delta_params,
                                  const FecProtectionParams *key_params);

  protected:
   WebRtc_Word32 CheckPayloadType(const WebRtc_Word8 payload_type,
                                  RtpVideoCodecTypes *video_type);

  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);

   void BuildRtxPacket(WebRtc_UWord8* buffer, WebRtc_UWord16* length,
                       WebRtc_UWord8* buffer_rtx);

   WebRtc_Word32 id_;
   const bool audio_configured_;
   RTPSenderAudio *audio_;
   RTPSenderVideo *video_;

   PacedSender *paced_sender_;
   CriticalSectionWrapper *send_critsect_;

   Transport *transport_;
   bool sending_media_;

   WebRtc_UWord16 max_payload_length_;
   WebRtc_UWord16 target_send_bitrate_;
   WebRtc_UWord16 packet_over_head_;

   WebRtc_Word8 payload_type_;
   std::map<WebRtc_Word8, ModuleRTPUtility::Payload *> payload_type_map_;

   RtpHeaderExtensionMap rtp_header_extension_map_;
   WebRtc_Word32 transmission_time_offset_;

   // NACK
   WebRtc_UWord32 nack_byte_count_times_[NACK_BYTECOUNT_SIZE];
   WebRtc_Word32 nack_byte_count_[NACK_BYTECOUNT_SIZE];
   Bitrate nack_bitrate_;

   RTPPacketHistory *packet_history_;

   // Statistics
   WebRtc_UWord32 packets_sent_;
   WebRtc_UWord32 payload_bytes_sent_;

   // RTP variables
   bool start_time_stamp_forced_;
   WebRtc_UWord32 start_time_stamp_;
   SSRCDatabase &ssrc_db_;
   WebRtc_UWord32 remote_ssrc_;
   bool sequence_number_forced_;
   WebRtc_UWord16 sequence_number_;
   WebRtc_UWord16 sequence_number_rtx_;
   bool ssrc_forced_;
   WebRtc_UWord32 ssrc_;
   WebRtc_UWord32 time_stamp_;
   WebRtc_UWord8 csrcs_;
   WebRtc_UWord32 csrc_[kRtpCsrcSize];
   bool include_csrcs_;
   RtxMode rtx_;
   WebRtc_UWord32 ssrc_rtx_;
 };

 }  // namespace webrtc

 #endif  // 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 "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 *data_buffer, const WebRtc_Word8 payload_type,
	const bool marker_bit, const WebRtc_UWord32 capture_time_stamp,
	const bool time_stamp_provided = true,
	const bool inc_sequence_number = 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 payload_name[RTP_PAYLOAD_NAME_SIZE],
	const WebRtc_Word8 payload_type, const WebRtc_UWord32 frequency,
	const WebRtc_UWord8 channels, const WebRtc_UWord32 rate);

	WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payload_type);

	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 arr_of_csrc[kRtpCsrcSize]) const;

	void SetCSRCStatus(const bool include);

	void SetCSRCs(const WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize],
	const WebRtc_UWord8 arr_length);

	WebRtc_Word32 SetMaxPayloadLength(const WebRtc_UWord16 length,
	const WebRtc_UWord16 packet_over_head);

	WebRtc_Word32 SendOutgoingData(
	const FrameType frame_type, const WebRtc_Word8 payload_type,
	const WebRtc_UWord32 time_stamp, int64_t capture_time_ms,
	const WebRtc_UWord8 *payload_data, const WebRtc_UWord32 payload_size,
	const RTPFragmentationHeader *fragmentation,
	VideoCodecInformation *codec_info = NULL,
	const RTPVideoTypeHeader * rtp_type_hdr = 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 transmission_time_offset);

	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 *data_buffer) const;

	WebRtc_UWord8 BuildTransmissionTimeOffsetExtension(
	WebRtc_UWord8 *data_buffer) 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 std::list<uint16_t>& nack_sequence_numbers,
	const WebRtc_UWord16 avg_rtt);

	void SetStorePacketsStatus(const bool enable,
	const WebRtc_UWord16 number_to_store);

	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 RtxMode mode, const bool set_ssrc,
	const WebRtc_UWord32 SSRC);

	void RTXStatus(RtxMode* mode, WebRtc_UWord32 *SSRC) const;

	// Functions wrapping RTPSenderInterface.
	virtual WebRtc_Word32 BuildRTPheader(
	WebRtc_UWord8 *data_buffer, const WebRtc_Word8 payload_type,
	const bool marker_bit, const WebRtc_UWord32 capture_time_stamp,
	const bool time_stamp_provided = true,
	const bool inc_sequence_number = 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 *telephone_event) 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 packet_size_samples);

	// 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 d_bov for
	// header-extension-for-audio-level-indication.
	WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_d_bov);

	// Set payload type for Redundant Audio Data RFC 2198.
	WebRtc_Word32 SetRED(const WebRtc_Word8 payload_type);

	// Get payload type for Redundant Audio Data RFC 2198.
	WebRtc_Word32 RED(WebRtc_Word8 *payload_type) const;

	// Video.
	VideoCodecInformation *CodecInformationVideo();

	RtpVideoCodecTypes VideoCodecType() const;

	WebRtc_UWord32 MaxConfiguredBitrateVideo() const;

	WebRtc_Word32 SendRTPIntraRequest();

	// FEC.
	WebRtc_Word32 SetGenericFECStatus(const bool enable,
	const WebRtc_UWord8 payload_type_red,
	const WebRtc_UWord8 payload_type_fec);

	WebRtc_Word32 GenericFECStatus(bool enable, WebRtc_UWord8 payload_type_red,
	WebRtc_UWord8 *payload_type_fec) const;

	WebRtc_Word32 SetFecParameters(const FecProtectionParams *delta_params,
	const FecProtectionParams *key_params);

	protected:
	WebRtc_Word32 CheckPayloadType(const WebRtc_Word8 payload_type,
	RtpVideoCodecTypes *video_type);

	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);

	void BuildRtxPacket(WebRtc_UWord8* buffer, WebRtc_UWord16* length,
	WebRtc_UWord8* buffer_rtx);

	WebRtc_Word32 id_;
	const bool audio_configured_;
	RTPSenderAudio *audio_;
	RTPSenderVideo *video_;

	PacedSender *paced_sender_;
	CriticalSectionWrapper *send_critsect_;

	Transport *transport_;
	bool sending_media_;

	WebRtc_UWord16 max_payload_length_;
	WebRtc_UWord16 target_send_bitrate_;
	WebRtc_UWord16 packet_over_head_;

	WebRtc_Word8 payload_type_;
	std::map<WebRtc_Word8, ModuleRTPUtility::Payload *> payload_type_map_;

	RtpHeaderExtensionMap rtp_header_extension_map_;
	WebRtc_Word32 transmission_time_offset_;

	// NACK
	WebRtc_UWord32 nack_byte_count_times_[NACK_BYTECOUNT_SIZE];
	WebRtc_Word32 nack_byte_count_[NACK_BYTECOUNT_SIZE];
	Bitrate nack_bitrate_;

	RTPPacketHistory *packet_history_;

	// Statistics
	WebRtc_UWord32 packets_sent_;
	WebRtc_UWord32 payload_bytes_sent_;

	// RTP variables
	bool start_time_stamp_forced_;
	WebRtc_UWord32 start_time_stamp_;
	SSRCDatabase &ssrc_db_;
	WebRtc_UWord32 remote_ssrc_;
	bool sequence_number_forced_;
	WebRtc_UWord16 sequence_number_;
	WebRtc_UWord16 sequence_number_rtx_;
	bool ssrc_forced_;
	WebRtc_UWord32 ssrc_;
	WebRtc_UWord32 time_stamp_;
	WebRtc_UWord8 csrcs_;
	WebRtc_UWord32 csrc_[kRtpCsrcSize];
	bool include_csrcs_;
	RtxMode rtx_;
	WebRtc_UWord32 ssrc_rtx_;
	};

	} // namespace webrtc

	#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_