webrtc/modules/rtp_rtcp/source/rtp_sender.cc

/*
 *  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.
 */

#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"

#include <cstdlib> // srand

#include "webrtc/modules/pacing/include/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"

namespace webrtc {
RTPSender::RTPSender(const WebRtc_Word32 id,
                     const bool audio,
                     RtpRtcpClock* clock,
                     Transport* transport,
                     RtpAudioFeedback* audio_feedback,
                     PacedSender* paced_sender)
    : Bitrate(clock),
      _id(id),
      _audioConfigured(audio),
      _audio(NULL),
      _video(NULL),
      paced_sender_(paced_sender),
      _sendCritsect(CriticalSectionWrapper::CreateCriticalSection()),
      _transport(transport),
      _sendingMedia(true),  // Default to sending media

      _maxPayloadLength(IP_PACKET_SIZE-28),  // default is IP-v4/UDP
      _targetSendBitrate(0),
      _packetOverHead(28),

      _payloadType(-1),
      _payloadTypeMap(),

      _rtpHeaderExtensionMap(),
      _transmissionTimeOffset(0),

      // NACK
      _nackByteCountTimes(),
      _nackByteCount(),
      _nackBitrate(clock),
      _packetHistory(new RTPPacketHistory(clock)),

      // statistics
      _packetsSent(0),
      _payloadBytesSent(0),

      _startTimeStampForced(false),
      _startTimeStamp(0),
      _ssrcDB(*SSRCDatabase::GetSSRCDatabase()),
      _remoteSSRC(0),
      _sequenceNumberForced(false),
      _sequenceNumber(0),
      _sequenceNumberRTX(0),
      _ssrcForced(false),
      _ssrc(0),
      _timeStamp(0),
      _CSRCs(0),
      _CSRC(),
      _includeCSRCs(true),
      _RTX(false),
      _ssrcRTX(0) {
  memset(_nackByteCountTimes, 0, sizeof(_nackByteCountTimes));
  memset(_nackByteCount, 0, sizeof(_nackByteCount));
  memset(_CSRC, 0, sizeof(_CSRC));
  // We need to seed the random generator.
  srand( (WebRtc_UWord32)_clock.GetTimeInMS() );
  _ssrc = _ssrcDB.CreateSSRC();  // Can't be 0.

  if (audio) {
    _audio = new RTPSenderAudio(id, &_clock, this);
    _audio->RegisterAudioCallback(audio_feedback);
  } else {
    _video = new RTPSenderVideo(id, &_clock, this);
  }
  WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
}

RTPSender::~RTPSender() {
  if (_remoteSSRC != 0) {
    _ssrcDB.ReturnSSRC(_remoteSSRC);
  }
  _ssrcDB.ReturnSSRC(_ssrc);

  SSRCDatabase::ReturnSSRCDatabase();
  delete _sendCritsect;
  while (!_payloadTypeMap.empty()) {
    std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
        _payloadTypeMap.begin();
    delete it->second;
    _payloadTypeMap.erase(it);
  }
  delete _packetHistory;
  delete _audio;
  delete _video;

  WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
}

void RTPSender::SetTargetSendBitrate(const WebRtc_UWord32 bits) {
  _targetSendBitrate = static_cast<uint16_t>(bits / 1000);
}

WebRtc_UWord16 RTPSender::ActualSendBitrateKbit() const {
    return (WebRtc_UWord16) (Bitrate::BitrateNow() / 1000);
}

WebRtc_UWord32 RTPSender::VideoBitrateSent() const {
  if (_video) {
    return _video->VideoBitrateSent();
  }
  return 0;
}

WebRtc_UWord32 RTPSender::FecOverheadRate() const {
  if (_video) {
    return _video->FecOverheadRate();
  }
  return 0;
}

WebRtc_UWord32 RTPSender::NackOverheadRate() const {
  return _nackBitrate.BitrateLast();
}

WebRtc_Word32 RTPSender::SetTransmissionTimeOffset(
    const WebRtc_Word32 transmissionTimeOffset) {
  if (transmissionTimeOffset > (0x800000 - 1) ||
      transmissionTimeOffset < -(0x800000 - 1)) {  // Word24
    return -1;
  }
  CriticalSectionScoped cs(_sendCritsect);
  _transmissionTimeOffset = transmissionTimeOffset;
  return 0;
}

WebRtc_Word32 RTPSender::RegisterRtpHeaderExtension(const RTPExtensionType type,
                                                    const WebRtc_UWord8 id) {
  CriticalSectionScoped cs(_sendCritsect);
  return _rtpHeaderExtensionMap.Register(type, id);
}

WebRtc_Word32 RTPSender::DeregisterRtpHeaderExtension(
    const RTPExtensionType type) {
  CriticalSectionScoped cs(_sendCritsect);
  return _rtpHeaderExtensionMap.Deregister(type);
}

WebRtc_UWord16 RTPSender::RtpHeaderExtensionTotalLength() const {
  CriticalSectionScoped cs(_sendCritsect);
  return _rtpHeaderExtensionMap.GetTotalLengthInBytes();
}

WebRtc_Word32 RTPSender::RegisterPayload(
    const char payloadName[RTP_PAYLOAD_NAME_SIZE],
    const WebRtc_Word8 payloadNumber,
    const WebRtc_UWord32 frequency,
    const WebRtc_UWord8 channels,
    const WebRtc_UWord32 rate) {
  assert(payloadName);
  CriticalSectionScoped cs(_sendCritsect);

  std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
      _payloadTypeMap.find(payloadNumber);

  if (_payloadTypeMap.end() != it) {
    // we already use this payload type
    ModuleRTPUtility::Payload* payload = it->second;
    assert(payload);

    // check if it's the same as we already have
    if (ModuleRTPUtility::StringCompare(payload->name, payloadName,
                                        RTP_PAYLOAD_NAME_SIZE - 1)) {
      if (_audioConfigured && payload->audio &&
          payload->typeSpecific.Audio.frequency == frequency &&
          (payload->typeSpecific.Audio.rate == rate ||
              payload->typeSpecific.Audio.rate == 0 || rate == 0)) {
        payload->typeSpecific.Audio.rate = rate;
        // Ensure that we update the rate if new or old is zero
        return 0;
      }
      if (!_audioConfigured && !payload->audio) {
        return 0;
      }
    }
    return -1;
  }
  WebRtc_Word32 retVal = -1;
  ModuleRTPUtility::Payload* payload = NULL;
  if (_audioConfigured) {
    retVal = _audio->RegisterAudioPayload(payloadName, payloadNumber, frequency,
                                          channels, rate, payload);
  } else {
    retVal = _video->RegisterVideoPayload(payloadName, payloadNumber, rate,
                                          payload);
  }
  if (payload) {
    _payloadTypeMap[payloadNumber] = payload;
  }
  return retVal;
}

WebRtc_Word32 RTPSender::DeRegisterSendPayload(const WebRtc_Word8 payloadType) {
  CriticalSectionScoped lock(_sendCritsect);

  std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
      _payloadTypeMap.find(payloadType);

  if (_payloadTypeMap.end() == it) {
    return -1;
  }
  ModuleRTPUtility::Payload* payload = it->second;
  delete payload;
  _payloadTypeMap.erase(it);
  return 0;
}

WebRtc_Word8 RTPSender::SendPayloadType() const {
  return _payloadType;
}

int RTPSender::SendPayloadFrequency() const {
  return _audio->AudioFrequency();
}

WebRtc_Word32 RTPSender::SetMaxPayloadLength(
    const WebRtc_UWord16 maxPayloadLength,
    const WebRtc_UWord16 packetOverHead) {
  // sanity check
  if (maxPayloadLength < 100 || maxPayloadLength > IP_PACKET_SIZE) {
    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
                 "%s invalid argument", __FUNCTION__);
    return -1;
  }
  CriticalSectionScoped cs(_sendCritsect);
  _maxPayloadLength = maxPayloadLength;
  _packetOverHead = packetOverHead;

  WEBRTC_TRACE(kTraceInfo, kTraceRtpRtcp, _id,
               "SetMaxPayloadLength to %d.", maxPayloadLength);
  return 0;
}

WebRtc_UWord16 RTPSender::MaxDataPayloadLength() const {
  if (_audioConfigured) {
    return _maxPayloadLength - RTPHeaderLength();
  } else {
    return _maxPayloadLength - RTPHeaderLength() -
        _video->FECPacketOverhead() - ((_RTX) ? 2 : 0);
        // Include the FEC/ULP/RED overhead.
  }
}

WebRtc_UWord16 RTPSender::MaxPayloadLength() const {
  return _maxPayloadLength;
}

WebRtc_UWord16 RTPSender::PacketOverHead() const {
  return _packetOverHead;
}

void RTPSender::SetRTXStatus(const bool enable,
                             const bool setSSRC,
                             const WebRtc_UWord32 SSRC) {
  CriticalSectionScoped cs(_sendCritsect);
  _RTX = enable;
  if (enable) {
    if (setSSRC) {
     _ssrcRTX = SSRC;
    } else {
     _ssrcRTX = _ssrcDB.CreateSSRC();   // can't be 0
    }
  }
}

void RTPSender::RTXStatus(bool* enable, WebRtc_UWord32* SSRC) const {
  CriticalSectionScoped cs(_sendCritsect);
  *enable = _RTX;
  *SSRC = _ssrcRTX;
}

WebRtc_Word32 RTPSender::CheckPayloadType(const WebRtc_Word8 payloadType,
                                          RtpVideoCodecTypes& videoType) {
  CriticalSectionScoped cs(_sendCritsect);

  if (payloadType < 0) {
    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
                 "\tinvalid payloadType (%d)", payloadType);
    return -1;
  }
  if (_audioConfigured) {
    WebRtc_Word8 redPlType = -1;
    if (_audio->RED(redPlType) == 0) {
      // We have configured RED.
      if (redPlType == payloadType) {
        // And it's a match...
        return 0;
      }
    }
  }
  if (_payloadType == payloadType) {
    if (!_audioConfigured) {
      videoType = _video->VideoCodecType();
    }
    return 0;
  }
  std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
      _payloadTypeMap.find(payloadType);
  if (it == _payloadTypeMap.end()) {
    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
                 "\tpayloadType:%d not registered", payloadType);
    return -1;
  }
  _payloadType = payloadType;
  ModuleRTPUtility::Payload* payload = it->second;
  assert(payload);
  if (!payload->audio && !_audioConfigured) {
    _video->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType);
    videoType = payload->typeSpecific.Video.videoCodecType;
    _video->SetMaxConfiguredBitrateVideo(payload->typeSpecific.Video.maxRate);
  }
  return 0;
}

WebRtc_Word32 RTPSender::SendOutgoingData(
    const FrameType frame_type,
    const WebRtc_Word8 payload_type,
    const WebRtc_UWord32 capture_timestamp,
    int64_t capture_time_ms,
    const WebRtc_UWord8* payload_data,
    const WebRtc_UWord32 payload_size,
    const RTPFragmentationHeader* fragmentation,
    VideoCodecInformation* codec_info,
    const RTPVideoTypeHeader* rtp_type_hdr) {
  {
    // Drop this packet if we're not sending media packets.
    CriticalSectionScoped cs(_sendCritsect);
    if (!_sendingMedia) {
      return 0;
    }
  }
  RtpVideoCodecTypes video_type = kRtpNoVideo;
  if (CheckPayloadType(payload_type, video_type) != 0) {
    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
                 "%s invalid argument failed to find payloadType:%d",
                 __FUNCTION__, payload_type);
    return -1;
  }

  if (_audioConfigured) {
    assert(frame_type == kAudioFrameSpeech ||
           frame_type == kAudioFrameCN ||
           frame_type == kFrameEmpty);

    return _audio->SendAudio(frame_type, payload_type, capture_timestamp,
                             payload_data, payload_size,fragmentation);
  } else {
    assert(frame_type != kAudioFrameSpeech &&
           frame_type != kAudioFrameCN);

    if (frame_type == kFrameEmpty) {
      return SendPaddingAccordingToBitrate(payload_type, capture_timestamp,
                                           capture_time_ms);
    }
    return _video->SendVideo(video_type,
                             frame_type,
                             payload_type,
                             capture_timestamp,
                             capture_time_ms,
                             payload_data,
                             payload_size,
                             fragmentation,
                             codec_info,
                             rtp_type_hdr);
  }
}

WebRtc_Word32 RTPSender::SendPaddingAccordingToBitrate(
    WebRtc_Word8 payload_type,
    WebRtc_UWord32 capture_timestamp,
    int64_t capture_time_ms) {
  // Current bitrate since last estimate(1 second) averaged with the
  // estimate since then, to get the most up to date bitrate.
  uint32_t current_bitrate = BitrateNow();
  int bitrate_diff = _targetSendBitrate * 1000 - current_bitrate;
  if (bitrate_diff <= 0) {
    return 0;
  }
  int bytes = 0;
  if (current_bitrate == 0) {
    // Start up phase. Send one 33.3 ms batch to start with.
    bytes = (bitrate_diff / 8) / 30;
  } else {
    bytes = (bitrate_diff / 8);
    // Cap at 200 ms of target send data.
    int bytes_cap = _targetSendBitrate * 25;  // 1000 / 8 / 5
    if (bytes > bytes_cap) {
      bytes = bytes_cap;
    }
  }
  return SendPadData(payload_type, capture_timestamp, capture_time_ms, bytes);
}

WebRtc_Word32 RTPSender::SendPadData(WebRtc_Word8 payload_type,
                                     WebRtc_UWord32 capture_timestamp,
                                     int64_t capture_time_ms,
                                     WebRtc_Word32 bytes) {
  // Drop this packet if we're not sending media packets
  if (!_sendingMedia) {
    return 0;
  }
  // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
  int max_length = 224;
  WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];

  for (; bytes > 0; bytes -= max_length) {
    int padding_bytes_in_packet = max_length;
    if (bytes < max_length) {
      padding_bytes_in_packet = (bytes + 16) & 0xffe0;  // Keep our modulus 32.
    }
    if (padding_bytes_in_packet < 32) {
       // Sanity don't send empty packets.
       break;
    }
    // Correct seq num, timestamp and payload type.
    int header_length = BuildRTPheader(data_buffer,
                                       payload_type,
                                       false,  // No markerbit.
                                       capture_timestamp,
                                       true,  // Timestamp provided.
                                       true);  // Increment sequence number.
    data_buffer[0] |= 0x20;  // Set padding bit.
    WebRtc_Word32* data =
        reinterpret_cast<WebRtc_Word32*>(&(data_buffer[header_length]));

    // Fill data buffer with random data.
    for (int j = 0; j < (padding_bytes_in_packet >> 2); j++) {
      data[j] = rand();
    }
    // Set number of padding bytes in the last byte of the packet.
    data_buffer[header_length + padding_bytes_in_packet - 1] =
        padding_bytes_in_packet;
    // Send the packet
    if (0 > SendToNetwork(data_buffer,
                          padding_bytes_in_packet,
                          header_length,
                          capture_time_ms,
                          kDontRetransmit)) {
      // Error sending the packet.
      break;
    }
  }
  if (bytes > 31) {  // 31 due to our modulus 32.
    // We did not manage to send all bytes.
    return -1;
  }
  return 0;
}

void RTPSender::SetStorePacketsStatus(
    const bool enable,
    const WebRtc_UWord16 numberToStore) {
  _packetHistory->SetStorePacketsStatus(enable, numberToStore);
}

bool RTPSender::StorePackets() const {
  return _packetHistory->StorePackets();
}

WebRtc_Word32 RTPSender::ReSendPacket(WebRtc_UWord16 packet_id,
                                      WebRtc_UWord32 min_resend_time) {

  WebRtc_UWord16 length = IP_PACKET_SIZE;
  WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
  WebRtc_UWord8* buffer_to_send_ptr = data_buffer;

  int64_t stored_time_in_ms;
  StorageType type;
  bool found = _packetHistory->GetRTPPacket(packet_id,
      min_resend_time, data_buffer, &length, &stored_time_in_ms, &type);
  if (!found) {
    // Packet not found.
    return 0;
  }
  if (length == 0 || type == kDontRetransmit) {
    // No bytes copied (packet recently resent, skip resending) or
    // packet should not be retransmitted.
    return 0;
  }
  WebRtc_UWord8 data_buffer_rtx[IP_PACKET_SIZE];
  if (_RTX) {
    buffer_to_send_ptr = data_buffer_rtx;

    CriticalSectionScoped cs(_sendCritsect);
    // Add RTX header.
    ModuleRTPUtility::RTPHeaderParser rtpParser(
        reinterpret_cast<const WebRtc_UWord8*>(data_buffer),
        length);

    WebRtcRTPHeader rtp_header;
    rtpParser.Parse(rtp_header);

    // Add original RTP header.
    memcpy(data_buffer_rtx, data_buffer, rtp_header.header.headerLength);

    // Replace sequence number.
    WebRtc_UWord8* ptr = data_buffer_rtx + 2;
    ModuleRTPUtility::AssignUWord16ToBuffer(ptr, _sequenceNumberRTX++);

    // Replace SSRC.
    ptr += 6;
    ModuleRTPUtility::AssignUWord32ToBuffer(ptr, _ssrcRTX);

    // Add OSN (original sequence number).
    ptr = data_buffer_rtx + rtp_header.header.headerLength;
    ModuleRTPUtility::AssignUWord16ToBuffer(
        ptr, rtp_header.header.sequenceNumber);
    ptr += 2;

    // Add original payload data.
    memcpy(ptr,
           data_buffer + rtp_header.header.headerLength,
           length - rtp_header.header.headerLength);
    length += 2;
  }
  WebRtc_Word32 bytes_sent = ReSendToNetwork(buffer_to_send_ptr, length);
  if (bytes_sent <= 0) {
    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
                 "Transport failed to resend packet_id %u", packet_id);
    return -1;
  }
  // Store the time when the packet was last resent.
  _packetHistory->UpdateResendTime(packet_id);
  return bytes_sent;
}

WebRtc_Word32 RTPSender::ReSendToNetwork(const WebRtc_UWord8* packet,
                                         const WebRtc_UWord32 size) {
  WebRtc_Word32 bytes_sent = -1;
  if (_transport) {
    bytes_sent = _transport->SendPacket(_id, packet, size);
  }
  if (bytes_sent <= 0) {
    return -1;
  }
  // Update send statistics
  CriticalSectionScoped cs(_sendCritsect);
  Bitrate::Update(bytes_sent);
  _packetsSent++;
  // We on purpose don't add to _payloadBytesSent since this is a
  // re-transmit and not new payload data.
  return bytes_sent;
}

int RTPSender::SelectiveRetransmissions() const {
  if (!_video) return -1;
  return _video->SelectiveRetransmissions();
}

int RTPSender::SetSelectiveRetransmissions(uint8_t settings) {
  if (!_video) return -1;
  return _video->SetSelectiveRetransmissions(settings);
}

void RTPSender::OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
                               const WebRtc_UWord16* nackSequenceNumbers,
                               const WebRtc_UWord16 avgRTT) {
  const WebRtc_Word64 now = _clock.GetTimeInMS();
  WebRtc_UWord32 bytesReSent = 0;

  // Enough bandwidth to send NACK?
  if (!ProcessNACKBitRate(now)) {
    WEBRTC_TRACE(kTraceStream,
                 kTraceRtpRtcp,
                 _id,
                 "NACK bitrate reached. Skip sending NACK response. Target %d",
                 _targetSendBitrate);
    return;
  }

  for (WebRtc_UWord16 i = 0; i < nackSequenceNumbersLength; ++i) {
    const WebRtc_Word32 bytesSent = ReSendPacket(nackSequenceNumbers[i],
                                                 5+avgRTT);
    if (bytesSent > 0) {
      bytesReSent += bytesSent;
    } else if (bytesSent == 0) {
      // The packet has previously been resent.
      // Try resending next packet in the list.
      continue;
    } else if (bytesSent < 0) {
      // Failed to send one Sequence number. Give up the rest in this nack.
      WEBRTC_TRACE(kTraceWarning,
                   kTraceRtpRtcp,
                   _id,
                   "Failed resending RTP packet %d, Discard rest of packets",
                   nackSequenceNumbers[i]);
      break;
    }
    // delay bandwidth estimate (RTT * BW)
    if (_targetSendBitrate != 0 && avgRTT) {
      // kbits/s * ms = bits => bits/8 = bytes
      WebRtc_UWord32 targetBytes =
          (static_cast<WebRtc_UWord32>(_targetSendBitrate) * avgRTT) >> 3;
      if (bytesReSent > targetBytes) {
        break; // ignore the rest of the packets in the list
      }
    }
  }
  if (bytesReSent > 0) {
    // TODO(pwestin) consolidate these two methods.
    UpdateNACKBitRate(bytesReSent, now);
    _nackBitrate.Update(bytesReSent);
  }
}

bool RTPSender::ProcessNACKBitRate(const WebRtc_UWord32 now) {
  WebRtc_UWord32 num = 0;
  WebRtc_Word32 byteCount = 0;
  const WebRtc_UWord32 avgInterval=1000;

  CriticalSectionScoped cs(_sendCritsect);

  if (_targetSendBitrate == 0) {
    return true;
  }
  for (num = 0; num < NACK_BYTECOUNT_SIZE; num++) {
    if ((now - _nackByteCountTimes[num]) > avgInterval) {
      // don't use data older than 1sec
      break;
    } else {
      byteCount += _nackByteCount[num];
    }
  }
  WebRtc_Word32 timeInterval = avgInterval;
  if (num == NACK_BYTECOUNT_SIZE) {
    // More than NACK_BYTECOUNT_SIZE nack messages has been received
    // during the last msgInterval
    timeInterval = now - _nackByteCountTimes[num-1];
    if (timeInterval < 0) {
      timeInterval = avgInterval;
    }
  }
  return (byteCount*8) < (_targetSendBitrate * timeInterval);
}

void RTPSender::UpdateNACKBitRate(const WebRtc_UWord32 bytes,
                                  const WebRtc_UWord32 now) {
  CriticalSectionScoped cs(_sendCritsect);

  // save bitrate statistics
  if (bytes > 0) {
    if (now == 0) {
      // add padding length
      _nackByteCount[0] += bytes;
    } else {
      if (_nackByteCountTimes[0] == 0) {
        // first no shift
      } else {
        // shift
        for (int i = (NACK_BYTECOUNT_SIZE-2); i >= 0 ; i--) {
          _nackByteCount[i+1] = _nackByteCount[i];
          _nackByteCountTimes[i+1] = _nackByteCountTimes[i];
        }
      }
      _nackByteCount[0] = bytes;
      _nackByteCountTimes[0] = now;
    }
  }
}

void RTPSender::TimeToSendPacket(uint16_t sequence_number,
                                 int64_t capture_time_ms) {
  StorageType type;
  uint16_t length = IP_PACKET_SIZE;
  uint8_t data_buffer[IP_PACKET_SIZE];
  int64_t stored_time_ms;  // TODO(pwestin) can we depricate this?

  if (_packetHistory == NULL) {
    return;
  }
  if (!_packetHistory->GetRTPPacket(sequence_number, 0, data_buffer,
                                    &length, &stored_time_ms, &type)) {
    assert(false);
    return;
  }
  assert(length > 0);

  ModuleRTPUtility::RTPHeaderParser rtpParser(data_buffer, length);
  WebRtcRTPHeader rtp_header;
  rtpParser.Parse(rtp_header);

  int64_t diff_ms = _clock.GetTimeInMS() - capture_time_ms;
  if (UpdateTransmissionTimeOffset(data_buffer, length, rtp_header, diff_ms)) {
    // Update stored packet in case of receiving a re-transmission request.
    _packetHistory->ReplaceRTPHeader(data_buffer,
                                     rtp_header.header.sequenceNumber,
                                     rtp_header.header.headerLength);
  }
  int bytes_sent = -1;
  if (_transport) {
    bytes_sent = _transport->SendPacket(_id, data_buffer, length);
  }
  if (bytes_sent <= 0) {
    return;
  }
  // Update send statistics
  CriticalSectionScoped cs(_sendCritsect);
  Bitrate::Update(bytes_sent);
  _packetsSent++;
  if (bytes_sent > rtp_header.header.headerLength) {
    _payloadBytesSent += bytes_sent - rtp_header.header.headerLength;
  }
}

// TODO(pwestin): send in the RTPHeaderParser to avoid parsing it again
WebRtc_Word32 RTPSender::SendToNetwork(uint8_t* buffer,
                                       int payload_length,
                                       int rtp_header_length,
                                       int64_t capture_time_ms,
                                       StorageType storage) {
  ModuleRTPUtility::RTPHeaderParser rtpParser(buffer,
      payload_length + rtp_header_length);
  WebRtcRTPHeader rtp_header;
  rtpParser.Parse(rtp_header);

  // |capture_time_ms| <= 0 is considered invalid.
  // TODO(holmer): This should be changed all over Video Engine so that negative
  // time is consider invalid, while 0 is considered a valid time.
  if (capture_time_ms > 0) {
    int64_t time_now = _clock.GetTimeInMS();
    UpdateTransmissionTimeOffset(buffer, payload_length + rtp_header_length,
                                 rtp_header, time_now - capture_time_ms);
  }
  // Used for NACK and to spread out the transmission of packets.
  if (_packetHistory->PutRTPPacket(buffer, rtp_header_length + payload_length,
      _maxPayloadLength, capture_time_ms, storage) != 0) {
    return -1;
  }
  if (paced_sender_) {
    if (!paced_sender_ ->SendPacket(PacedSender::kNormalPriority,
                                    rtp_header.header.ssrc,
                                    rtp_header.header.sequenceNumber,
                                    capture_time_ms,
                                    payload_length + rtp_header_length)) {
      // We can't send the packet right now.
      // We will be called when it is time.
      return payload_length + rtp_header_length;
    }
  }
  // Send packet
  WebRtc_Word32 bytes_sent = -1;
  if (_transport) {
    bytes_sent = _transport->SendPacket(_id,
                                        buffer,
                                        payload_length + rtp_header_length);
  }
  if (bytes_sent <= 0) {
    return -1;
  }
  // Update send statistics
  CriticalSectionScoped cs(_sendCritsect);
  Bitrate::Update(bytes_sent);
  _packetsSent++;
  if (bytes_sent > rtp_header_length) {
    _payloadBytesSent += bytes_sent - rtp_header_length;
  }
  return 0;
}

void RTPSender::ProcessBitrate() {
  CriticalSectionScoped cs(_sendCritsect);
  Bitrate::Process();
  _nackBitrate.Process();
  if (_audioConfigured) {
    return;
  }
  _video->ProcessBitrate();
}

WebRtc_UWord16 RTPSender::RTPHeaderLength() const {
  WebRtc_UWord16 rtpHeaderLength = 12;
  if (_includeCSRCs) {
    rtpHeaderLength += sizeof(WebRtc_UWord32)*_CSRCs;
  }
  rtpHeaderLength += RtpHeaderExtensionTotalLength();
  return rtpHeaderLength;
}

WebRtc_UWord16 RTPSender::IncrementSequenceNumber() {
  CriticalSectionScoped cs(_sendCritsect);
  return _sequenceNumber++;
}

void RTPSender::ResetDataCounters() {
  _packetsSent = 0;
  _payloadBytesSent = 0;
}

WebRtc_UWord32 RTPSender::Packets() const {
  // Don't use critsect to avoid potental deadlock
  return _packetsSent;
}

// number of sent RTP bytes
// dont use critsect to avoid potental deadlock
WebRtc_UWord32 RTPSender::Bytes() const {
  return _payloadBytesSent;
}

WebRtc_Word32 RTPSender::BuildRTPheader(WebRtc_UWord8* dataBuffer,
                                        const WebRtc_Word8 payloadType,
                                        const bool markerBit,
                                        const WebRtc_UWord32 captureTimeStamp,
                                        const bool timeStampProvided,
                                        const bool incSequenceNumber) {
  assert(payloadType>=0);
  CriticalSectionScoped cs(_sendCritsect);

  dataBuffer[0] = static_cast<WebRtc_UWord8>(0x80);  // version 2
  dataBuffer[1] = static_cast<WebRtc_UWord8>(payloadType);
  if (markerBit) {
    dataBuffer[1] |= kRtpMarkerBitMask;  // MarkerBit is set
  }
  if (timeStampProvided) {
    _timeStamp = _startTimeStamp + captureTimeStamp;
  } else {
    // make a unique time stamp
    // we can't inc by the actual time, since then we increase the risk of back
    // timing.
    _timeStamp++;
  }
  ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, _sequenceNumber);
  ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+4, _timeStamp);
  ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, _ssrc);
  WebRtc_Word32 rtpHeaderLength = 12;

  // Add the CSRCs if any
  if (_includeCSRCs && _CSRCs > 0) {
    if (_CSRCs > kRtpCsrcSize) {
      // error
      assert(false);
      return -1;
    }
    WebRtc_UWord8* ptr = &dataBuffer[rtpHeaderLength];
    for (WebRtc_UWord32 i = 0; i < _CSRCs; ++i) {
      ModuleRTPUtility::AssignUWord32ToBuffer(ptr, _CSRC[i]);
      ptr +=4;
    }
    dataBuffer[0] = (dataBuffer[0]&0xf0) | _CSRCs;

    // Update length of header
    rtpHeaderLength += sizeof(WebRtc_UWord32)*_CSRCs;
  }
  _sequenceNumber++;  // prepare for next packet

  WebRtc_UWord16 len = BuildRTPHeaderExtension(dataBuffer + rtpHeaderLength);
  if (len) {
    dataBuffer[0] |= 0x10;  // set eXtension bit
    rtpHeaderLength += len;
  }
  return rtpHeaderLength;
}

WebRtc_UWord16 RTPSender::BuildRTPHeaderExtension(
    WebRtc_UWord8* dataBuffer) const {
  if (_rtpHeaderExtensionMap.Size() <= 0) {
    return 0;
  }
  /* RTP header extension, RFC 3550.
     0                   1                   2                   3
     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |      defined by profile       |           length              |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |                        header extension                       |
    |                             ....                              |
  */
  const WebRtc_UWord32 kPosLength = 2;
  const WebRtc_UWord32 kHeaderLength = RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES;

  // Add extension ID (0xBEDE).
  ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer,
                                          RTP_ONE_BYTE_HEADER_EXTENSION);

  // Add extensions.
  WebRtc_UWord16 total_block_length = 0;

  RTPExtensionType type = _rtpHeaderExtensionMap.First();
  while (type != kRtpExtensionNone) {
    WebRtc_UWord8 block_length = 0;
    if (type == kRtpExtensionTransmissionTimeOffset) {
      block_length = BuildTransmissionTimeOffsetExtension(
          dataBuffer + kHeaderLength + total_block_length);
    }
    total_block_length += block_length;
    type = _rtpHeaderExtensionMap.Next(type);
  }
  if (total_block_length == 0) {
    // No extension added.
    return 0;
  }
  // Set header length (in number of Word32, header excluded).
  assert(total_block_length % 4 == 0);
  ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer + kPosLength,
                                          total_block_length / 4);
  // Total added length.
  return kHeaderLength + total_block_length;
}

WebRtc_UWord8 RTPSender::BuildTransmissionTimeOffsetExtension(
    WebRtc_UWord8* dataBuffer) const {
  // From RFC 5450: Transmission Time Offsets in RTP Streams.
  //
  // The transmission time is signaled to the receiver in-band using the
  // general mechanism for RTP header extensions [RFC5285]. The payload
  // of this extension (the transmitted value) is a 24-bit signed integer.
  // When added to the RTP timestamp of the packet, it represents the
  // "effective" RTP transmission time of the packet, on the RTP
  // timescale.
  //
  // The form of the transmission offset extension block:
  //
  //    0                   1                   2                   3
  //    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
  //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  //   |  ID   | len=2 |              transmission offset              |
  //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

  // Get id defined by user.
  WebRtc_UWord8 id;
  if (_rtpHeaderExtensionMap.GetId(kRtpExtensionTransmissionTimeOffset, &id)
      != 0) {
    // Not registered.
    return 0;
  }
  int pos = 0;
  const WebRtc_UWord8 len = 2;
  dataBuffer[pos++] = (id << 4) + len;
  ModuleRTPUtility::AssignUWord24ToBuffer(dataBuffer + pos,
                                          _transmissionTimeOffset);
  pos += 3;
  assert(pos == TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES);
  return TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES;
}

bool RTPSender::UpdateTransmissionTimeOffset(
    WebRtc_UWord8* rtp_packet,
    const WebRtc_UWord16 rtp_packet_length,
    const WebRtcRTPHeader& rtp_header,
    const WebRtc_Word64 time_diff_ms) const {
  CriticalSectionScoped cs(_sendCritsect);

  // Get length until start of transmission block.
  int transmission_block_pos =
      _rtpHeaderExtensionMap.GetLengthUntilBlockStartInBytes(
          kRtpExtensionTransmissionTimeOffset);
  if (transmission_block_pos < 0) {
    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
                 "Failed to update transmission time offset, not registered.");
    return false;
  }
  int block_pos = 12 + rtp_header.header.numCSRCs + transmission_block_pos;
  if (rtp_packet_length < block_pos + 4 ||
      rtp_header.header.headerLength < block_pos + 4) {
    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
                 "Failed to update transmission time offset, invalid length.");
    return false;
  }
  // Verify that header contains extension.
  if (!((rtp_packet[12 + rtp_header.header.numCSRCs] == 0xBE) &&
      (rtp_packet[12 + rtp_header.header.numCSRCs + 1] == 0xDE))) {
    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
        "Failed to update transmission time offset, hdr extension not found.");
    return false;
  }
  // Get id.
  WebRtc_UWord8 id = 0;
  if (_rtpHeaderExtensionMap.GetId(kRtpExtensionTransmissionTimeOffset,
                                   &id) != 0) {
    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
                 "Failed to update transmission time offset, no id.");
    return false;
  }
  // Verify first byte in block.
  const WebRtc_UWord8 first_block_byte = (id << 4) + 2;
  if (rtp_packet[block_pos] != first_block_byte) {
    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
                 "Failed to update transmission time offset.");
    return false;
  }
  // Update transmission offset field.
  ModuleRTPUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1,
                                          time_diff_ms * 90);  // RTP timestamp.
  return true;
}

void RTPSender::SetSendingStatus(const bool enabled) {
  if (enabled) {
    WebRtc_UWord32 freq;
    if (_audioConfigured) {
      WebRtc_UWord32 frequency = _audio->AudioFrequency();

      // sanity
      switch(frequency) {
        case 8000:
        case 12000:
        case 16000:
        case 24000:
        case 32000:
          break;
        default:
          assert(false);
          return;
      }
      freq = frequency;
    } else {
      freq = 90000; // 90 KHz for all video
    }
    WebRtc_UWord32 RTPtime = ModuleRTPUtility::GetCurrentRTP(&_clock, freq);

    // will be ignored if it's already configured via API
    SetStartTimestamp(RTPtime, false);
  } else {
    if (!_ssrcForced) {
      // generate a new SSRC
      _ssrcDB.ReturnSSRC(_ssrc);
      _ssrc = _ssrcDB.CreateSSRC();   // can't be 0

    }
    // Don't initialize seq number if SSRC passed externally.
    if (!_sequenceNumberForced && !_ssrcForced) {
      // generate a new sequence number
      _sequenceNumber = rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);
    }
  }
}

void RTPSender::SetSendingMediaStatus(const bool enabled) {
  CriticalSectionScoped cs(_sendCritsect);
  _sendingMedia = enabled;
}

bool RTPSender::SendingMedia() const {
  CriticalSectionScoped cs(_sendCritsect);
  return _sendingMedia;
}

WebRtc_UWord32 RTPSender::Timestamp() const {
  CriticalSectionScoped cs(_sendCritsect);
  return _timeStamp;
}

void RTPSender::SetStartTimestamp(WebRtc_UWord32 timestamp, bool force) {
  CriticalSectionScoped cs(_sendCritsect);
  if (force) {
    _startTimeStampForced = force;
    _startTimeStamp = timestamp;
  } else {
    if (!_startTimeStampForced) {
      _startTimeStamp = timestamp;
    }
  }
}

WebRtc_UWord32 RTPSender::StartTimestamp() const {
  CriticalSectionScoped cs(_sendCritsect);
  return _startTimeStamp;
}

WebRtc_UWord32 RTPSender::GenerateNewSSRC() {
  // if configured via API, return 0
  CriticalSectionScoped cs(_sendCritsect);

  if (_ssrcForced) {
    return 0;
  }
  _ssrc = _ssrcDB.CreateSSRC();   // can't be 0
  return _ssrc;
}

void RTPSender::SetSSRC(WebRtc_UWord32 ssrc) {
  // this is configured via the API
  CriticalSectionScoped cs(_sendCritsect);

  if (_ssrc == ssrc && _ssrcForced) {
    return; // since it's same ssrc, don't reset anything
  }
  _ssrcForced = true;
  _ssrcDB.ReturnSSRC(_ssrc);
  _ssrcDB.RegisterSSRC(ssrc);
  _ssrc = ssrc;
  if (!_sequenceNumberForced) {
    _sequenceNumber = rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);
  }
}

WebRtc_UWord32 RTPSender::SSRC() const {
  CriticalSectionScoped cs(_sendCritsect);
  return _ssrc;
}

void RTPSender::SetCSRCStatus(const bool include) {
  _includeCSRCs = include;
}

void RTPSender::SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
                         const WebRtc_UWord8 arrLength) {
  assert(arrLength <= kRtpCsrcSize);
  CriticalSectionScoped cs(_sendCritsect);

  for (int i = 0; i < arrLength;i++) {
    _CSRC[i] = arrOfCSRC[i];
  }
  _CSRCs = arrLength;
}

WebRtc_Word32 RTPSender::CSRCs(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const {
  assert(arrOfCSRC);
  CriticalSectionScoped cs(_sendCritsect);
  for (int i = 0; i < _CSRCs && i < kRtpCsrcSize;i++) {
    arrOfCSRC[i] = _CSRC[i];
  }
  return _CSRCs;
}

void RTPSender::SetSequenceNumber(WebRtc_UWord16 seq) {
  CriticalSectionScoped cs(_sendCritsect);
  _sequenceNumberForced = true;
  _sequenceNumber = seq;
}

WebRtc_UWord16 RTPSender::SequenceNumber() const {
  CriticalSectionScoped cs(_sendCritsect);
  return _sequenceNumber;
}

/*
 *    Audio
 */
WebRtc_Word32 RTPSender::SendTelephoneEvent(const WebRtc_UWord8 key,
                                            const WebRtc_UWord16 time_ms,
                                            const WebRtc_UWord8 level) {
  if (!_audioConfigured) {
    return -1;
  }
  return _audio->SendTelephoneEvent(key, time_ms, level);
}

bool RTPSender::SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const {
  if (!_audioConfigured) {
    return false;
  }
  return _audio->SendTelephoneEventActive(telephoneEvent);
}

WebRtc_Word32 RTPSender::SetAudioPacketSize(
    const WebRtc_UWord16 packetSizeSamples) {
  if (!_audioConfigured) {
    return -1;
  }
  return _audio->SetAudioPacketSize(packetSizeSamples);
}

WebRtc_Word32
RTPSender::SetAudioLevelIndicationStatus(const bool enable,
                                         const WebRtc_UWord8 ID) {
  if (!_audioConfigured) {
    return -1;
  }
  return _audio->SetAudioLevelIndicationStatus(enable, ID);
}

WebRtc_Word32 RTPSender::AudioLevelIndicationStatus(bool& enable,
                                                    WebRtc_UWord8& ID) const {
  return _audio->AudioLevelIndicationStatus(enable, ID);
}

WebRtc_Word32 RTPSender::SetAudioLevel(const WebRtc_UWord8 level_dBov) {
  return _audio->SetAudioLevel(level_dBov);
}

WebRtc_Word32 RTPSender::SetRED(const WebRtc_Word8 payloadType) {
  if (!_audioConfigured) {
    return -1;
  }
  return _audio->SetRED(payloadType);
}

WebRtc_Word32 RTPSender::RED(WebRtc_Word8& payloadType) const {
  if (!_audioConfigured) {
    return -1;
  }
  return _audio->RED(payloadType);
}

/*
 *    Video
 */
VideoCodecInformation* RTPSender::CodecInformationVideo() {
  if (_audioConfigured) {
    return NULL;
  }
  return _video->CodecInformationVideo();
}

RtpVideoCodecTypes RTPSender::VideoCodecType() const {
  if (_audioConfigured) {
    return kRtpNoVideo;
  }
  return _video->VideoCodecType();
}

WebRtc_UWord32 RTPSender::MaxConfiguredBitrateVideo() const {
  if (_audioConfigured) {
    return 0;
  }
  return _video->MaxConfiguredBitrateVideo();
}

WebRtc_Word32 RTPSender::SendRTPIntraRequest() {
  if (_audioConfigured) {
    return -1;
  }
  return _video->SendRTPIntraRequest();
}

WebRtc_Word32 RTPSender::SetGenericFECStatus(
    const bool enable,
    const WebRtc_UWord8 payloadTypeRED,
    const WebRtc_UWord8 payloadTypeFEC) {
  if (_audioConfigured) {
    return -1;
  }
  return _video->SetGenericFECStatus(enable, payloadTypeRED, payloadTypeFEC);
}

WebRtc_Word32 RTPSender::GenericFECStatus(bool& enable,
                                          WebRtc_UWord8& payloadTypeRED,
                                          WebRtc_UWord8& payloadTypeFEC) const {
  if (_audioConfigured) {
    return -1;
  }
  return _video->GenericFECStatus(enable, payloadTypeRED, payloadTypeFEC);
}

WebRtc_Word32 RTPSender::SetFecParameters(
    const FecProtectionParams* delta_params,
    const FecProtectionParams* key_params) {
  if (_audioConfigured) {
    return -1;
  }
  return _video->SetFecParameters(delta_params, key_params);
}
}  // namespace webrtc