rtp.coffee

# RTP spec:
#   RFC 3550  http://tools.ietf.org/html/rfc3550
# RTP payload format for H.264 video:
#   RFC 6184  http://tools.ietf.org/html/rfc6184
# RTP payload format for AAC audio:
#   RFC 3640  http://tools.ietf.org/html/rfc3640
#   RFC 5691  http://tools.ietf.org/html/rfc5691
#
# TODO: Use DON (decoding order number) to carry B-frames.
#       DON is to RTP what DTS is to MPEG-TS.

Bits = require './bits'
aac = require './aac'
logger = require './logger'

# Number of seconds from 1900-01-01 to 1970-01-01
EPOCH = 2208988800

# Constant for calculating NTP fractional second
NTP_SCALE_FRAC = 4294.967295

# Minimum length of an RTP header
RTP_HEADER_LEN = 12

MAX_PAYLOAD_SIZE = 1360

MAX_SEQUENCE_NUMBER = 65535

class RTPParser
  constructor: ->
    @eventListeners = {}
    @packetBuffers = {}
    @fragmentedH264PacketBuffer = {}
    @h264NALUnitBuffer = {}
    @aacAccessUnitBuffer = {}

    # config
    @unorderedPacketBufferSize = 10

  emit: (name, data...) ->
    if @eventListeners[name]?
      for listener in @eventListeners[name]
        listener data...
    return

  on: (name, listener) ->
    if @eventListeners[name]?
      @eventListeners[name].push listener
    else
      @eventListeners[name] = [ listener ]

  feedUnorderedAACBuffer: (buf, clientId, params) ->
    packet = api.parseAACPacket buf, params
    @feedUnorderedPacket "aac:#{clientId}", packet

  feedUnorderedH264Buffer: (buf, clientId) ->
    packet = api.parseH264Packet buf
    @feedUnorderedPacket "h264:#{clientId}", packet

  clearAllUnorderedPacketBuffers: ->
    @packetBuffers = {}

  clearUnorderedPacketBuffer: (tag) ->
    delete @packetBuffers["h264:#{tag}"]
    delete @packetBuffers["aac:#{tag}"]

  feedUnorderedPacket: (tag, packet) ->
    if not @packetBuffers[tag]?
      @packetBuffers[tag] =
        nextSequenceNumber: packet.rtpHeader.sequenceNumber
        minSequenceNumberInBuffer: null
        buffer: []
    packetBuffer = @packetBuffers[tag]
    if packetBuffer.nextSequenceNumber is packet.rtpHeader.sequenceNumber
      @onOrderedPacket tag, packet
      packetBuffer.nextSequenceNumber++
      if packetBuffer.nextSequenceNumber > MAX_SEQUENCE_NUMBER
        packetBuffer.nextSequenceNumber = 0
    else
      # stash packet in buffer
      buffers = packetBuffer.buffer
      buffers.push packet
      if buffers.length >= 2
        buffers.sort (a, b) ->
          numberA = a.rtpHeader.sequenceNumber
          numberB = b.rtpHeader.sequenceNumber
          if numberA - numberB >= 60000  # large enough gap
            return -1  # a comes first
          else if numberB - numberA >= 60000  # large enough gap
            return 1   # b comes first
          else
            return numberA - numberB
        while (buffers.length) > 0 and
        (buffers[0].rtpHeader.sequenceNumber is packetBuffer.nextSequenceNumber)
          @onOrderedPacket tag, buffers.shift()
          packetBuffer.nextSequenceNumber++
          if packetBuffer.nextSequenceNumber > MAX_SEQUENCE_NUMBER
            packetBuffer.nextSequenceNumber = 0
        while buffers.length >= 2
          latestSequenceNumber = buffers[buffers.length-1].rtpHeader.sequenceNumber
          diff = latestSequenceNumber - packetBuffer.nextSequenceNumber
          if diff < 0
            diff += MAX_SEQUENCE_NUMBER + 1
          if diff < @unorderedPacketBufferSize
            break

          firstPacket = buffers.shift()
          if packetBuffer.nextSequenceNumber isnt firstPacket.rtpHeader.sequenceNumber
            discardedSequenceNumber = firstPacket.rtpHeader.sequenceNumber - 1
            if discardedSequenceNumber < 0
              discardedSequenceNumber += MAX_SEQUENCE_NUMBER
            if packetBuffer.nextSequenceNumber isnt discardedSequenceNumber
              logger.warn "rtp: #{tag}: incoming packet loss: sequence number #{packetBuffer.nextSequenceNumber}-#{discardedSequenceNumber}"
            else
              logger.warn "rtp: #{tag}: incoming packet loss: sequence number #{discardedSequenceNumber}"
          # consume the first packet
          @onOrderedPacket tag, firstPacket

          packetBuffer.nextSequenceNumber = firstPacket.rtpHeader.sequenceNumber + 1
          if packetBuffer.nextSequenceNumber > MAX_SEQUENCE_NUMBER
            packetBuffer.nextSequenceNumber = 0

  onH264NALUnit: (clientId, nalUnit, packet, timestamp) ->
    if not @h264NALUnitBuffer[clientId]?
      @h264NALUnitBuffer[clientId] = []
    @h264NALUnitBuffer[clientId].push nalUnit
    if packet.rtpHeader.marker
      @emit 'h264_nal_units', clientId, @h264NALUnitBuffer[clientId], timestamp
      @h264NALUnitBuffer[clientId] = []

  onAACAccessUnits: (clientId, accessUnits, packet, timestamp) ->
    if not @aacAccessUnitBuffer[clientId]?
      @aacAccessUnitBuffer[clientId] = []
    @aacAccessUnitBuffer[clientId] = @aacAccessUnitBuffer[clientId].concat accessUnits
    if packet.rtpHeader.marker
      @emit 'aac_access_units', clientId, @aacAccessUnitBuffer[clientId], timestamp
      @aacAccessUnitBuffer[clientId] = []

  onOrderedPacket: (tag, packet) ->
    if (match = /^h264:(.*)$/.exec tag)?
      clientId = match[1]
      if packet.h264.fu_a?  # FU-A
        # startBit and endBit won't both be set to 1 in the same FU header
        if packet.h264.fu_a.fuHeader.startBit
          @fragmentedH264PacketBuffer[tag] = [
            new Buffer [ (packet.h264.nal_ref_idc << 5) | packet.h264.fu_a.fuHeader.nal_unit_payload_type ]
            packet.h264.fu_a.nal_unit_fragment
          ]
        else if @fragmentedH264PacketBuffer[tag]?
          @fragmentedH264PacketBuffer[tag].push packet.h264.fu_a.nal_unit_fragment
        else
          logger.warn "rtp: #{tag}: discarded fragmented incoming packet: #{packet.rtpHeader.sequenceNumber}"
          return
        if packet.h264.fu_a.fuHeader.endBit
          @onH264NALUnit clientId, Buffer.concat(@fragmentedH264PacketBuffer[tag]), packet, packet.rtpHeader.timestamp
          @fragmentedH264PacketBuffer[tag] = null
      else if packet.h264.stap_a?  # STAP-A
        for nalUnit in packet.h264.stap_a.nalUnits
          @onH264NALUnit clientId, nalUnit, packet, packet.rtpHeader.timestamp
      else # single NAL unit
        @onH264NALUnit clientId, packet.h264.nal_unit, packet, packet.rtpHeader.timestamp
    else if (match = /^aac:(.*)$/.exec tag)?
      clientId = match[1]
      @onAACAccessUnits clientId, packet.aac.accessUnits, packet, packet.rtpHeader.timestamp
    else
      throw new Error "Unknown tag: #{tag}"

api =
  RTPParser: RTPParser

  # Number of bytes in RTP header
  RTP_HEADER_LEN: RTP_HEADER_LEN

  RTCP_PACKET_TYPE_SENDER_REPORT      : 200  # SR
  RTCP_PACKET_TYPE_RECEIVER_REPORT    : 201  # RR
  RTCP_PACKET_TYPE_SOURCE_DESCRIPTION : 202  # SDES
  RTCP_PACKET_TYPE_GOODBYE            : 203  # BYE
  RTCP_PACKET_TYPE_APPLICATION_DEFINED: 204  # APP

  H264_NAL_UNIT_TYPE_STAP_A: 24
  H264_NAL_UNIT_TYPE_STAP_B: 25
  H264_NAL_UNIT_TYPE_MTAP16: 26
  H264_NAL_UNIT_TYPE_MTAP24: 27
  H264_NAL_UNIT_TYPE_FU_A  : 28
  H264_NAL_UNIT_TYPE_FU_B  : 29

  # Remove padding from the end of the buffer
  removeTrailingPadding: (bits) ->
    paddingLength = bits.last_get_byte_at 0
    bits.remove_trailing_bytes paddingLength

  readRTCPSenderReport: (bits) ->
    # RFC 3550 - 6.4.1 SR: Sender Report RTCP Packet
    startBytePos = bits.current_position().byte
    info = {}
    info.version = bits.read_bits 2
    info.padding = bits.read_bit()
    if info.padding is 1
      api.removeTrailingPadding bits
    info.reportCount = bits.read_bits 5
    info.payloadType = bits.read_byte()  # == 200
    if info.payloadType isnt api.RTCP_PACKET_TYPE_SENDER_REPORT
      throw new Error "payload type must be #{api.RTCP_PACKET_TYPE_SENDER_REPORT}"
    info.wordsMinusOne = bits.read_bits 16
    info.totalBytes = (info.wordsMinusOne + 1) * 4
    info.ssrc = bits.read_bits 32
    info.ntpTimestamp = [ bits.read_bits(32), bits.read_bits(32) ]
    info.ntpTimestampInMs = api.ntpTimestampToTime info.ntpTimestamp
    info.rtpTimestamp = bits.read_bits 32
    info.senderPacketCount = bits.read_bits 32
    info.senderOctetCount = bits.read_bits 32

    info.reportBlocks = []
    for i in [0...info.reportCount]
      reportBlock = {}
      reportBlock.ssrc = bits.read_bits 32
      reportBlock.fractionLost = bits.read_byte()
      reportBlock.packetsLost = bits.read_int 24
      reportBlock.highestSequenceNumber = bits.read_bits 32
      reportBlock.jitter = bits.read_bits 32
      reportBlock.lastSR = bits.read_bits 32
      reportBlock.delaySinceLastSR = bits.read_bits 32
      info.reportBlocks.push reportBlock

    # skip padding bytes
    readBytes = bits.current_position().byte - startBytePos
    if readBytes < info.totalBytes
      bits.skip_bytes info.totalBytes - readBytes

    return info

  readRTCPReceiverReport: (bits) ->
    # RFC 3550 - 6.4.2 RR: Receiver Report RTCP Packet
    startBytePos = bits.current_position().byte
    info = {}
    info.version = bits.read_bits 2
    info.padding = bits.read_bit()
    if info.padding is 1
      api.removeTrailingPadding bits
    info.reportCount = bits.read_bits 5
    info.payloadType = bits.read_byte()  # == 201
    if info.payloadType isnt api.RTCP_PACKET_TYPE_RECEIVER_REPORT
      throw new Error "payload type must be #{api.RTCP_PACKET_TYPE_RECEIVER_REPORT}"
    info.wordsMinusOne = bits.read_bits 16
    info.totalBytes = (info.wordsMinusOne + 1) * 4
    info.ssrc = bits.read_bits 32
    info.reportBlocks = []
    for i in [0...info.reportCount]
      reportBlock = {}
      reportBlock.ssrc = bits.read_bits 32
      reportBlock.fractionLost = bits.read_byte()
      reportBlock.packetsLost = bits.read_int 24
      reportBlock.highestSequenceNumber = bits.read_bits 32
      reportBlock.jitter = bits.read_bits 32
      reportBlock.lastSR = bits.read_bits 32
      reportBlock.delaySinceLastSR = bits.read_bits 32
      info.reportBlocks.push reportBlock

    # skip padding bytes
    readBytes = bits.current_position().byte - startBytePos
    if readBytes < info.totalBytes
      bits.skip_bytes info.totalBytes - readBytes

    return info

  readRTCPSourceDescription: (bits) ->
    # RFC 3550 - 6.5 SDES: Source Description RTCP Packet
    startBytePos = bits.current_position().byte
    info = {}
    info.version = bits.read_bits 2
    info.padding = bits.read_bit()
    if info.padding is 1
      api.removeTrailingPadding bits
    info.sourceCount = bits.read_bits 5
    info.payloadType = bits.read_byte()  # == 202
    if info.payloadType isnt api.RTCP_PACKET_TYPE_SOURCE_DESCRIPTION
      throw new Error "payload type must be #{api.RTCP_PACKET_TYPE_SOURCE_DESCRIPTION}"
    info.wordsMinusOne = bits.read_bits 16
    info.totalBytes = (info.wordsMinusOne + 1) * 4
    info.chunks = []
    for i in [0...info.sourceCount]
      chunk = {}
      chunk.ssrc_csrc = bits.read_bits 32
      chunk.sdesItems = []
      chunk.sdes = {}
      loop
        sdesItem = {}
        sdesItem.type = bits.read_byte()
        if sdesItem.type is 0  # terminate the list
          # skip until the next 32-bit boundary
          bytesPastBoundary = (bits.current_position().byte - startBytePos) % 4
          if bytesPastBoundary > 0
            while bytesPastBoundary < 4
              nullOctet = bits.read_byte()
              if nullOctet isnt 0x00
                throw new Error "padding octet must be 0x00: #{nullOctet}"
              bytesPastBoundary++
          break

        sdesItem.octetCount = bits.read_byte()
        if sdesItem.octetCount > 255
          throw new Error "octet count too large: #{sdesItem.octetCount} <= 255"
        sdesItem.text = bits.read_bytes(sdesItem.octetCount).toString 'utf8'
        switch sdesItem.type
          when 1  # Canonical End-Point Identifier
            chunk.sdes.cname = sdesItem.text
          when 2  # User Name
            chunk.sdes.name = sdesItem.text
          when 3  # Electronic Mail Address
            chunk.sdes.email = sdesItem.text
          when 4  # Phone Number
            chunk.sdes.phone = sdesItem.text
          when 5  # Geographic User Location
            chunk.sdes.loc = sdesItem.text
          when 6  # Application or Tool Name
            chunk.sdes.tool = sdesItem.text
          when 7  # Notice/Status
            chunk.sdes.note = sdesItem.text
          when 8  # Private Extensions
            chunk.sdes.priv = sdesItem.text
          else
            throw new Error "unknown SDES item type in source description " +
              "RTCP packet: #{chunk.type} (maybe not implemented yet)"
        chunk.sdesItems.push sdesItem
      info.chunks.push chunk

    # skip padding bytes
    readBytes = bits.current_position().byte - startBytePos
    if readBytes < info.totalBytes
      bits.skip_bytes info.totalBytes - readBytes

    return info

  readRTCPGoodbye: (bits) ->
    # RFC 3550 - 6.6 BYE: Goodbye RTCP Packet
    startBytePos = bits.current_position().byte
    info = {}
    info.version = bits.read_bits 2
    info.padding = bits.read_bit()
    if info.padding is 1
      api.removeTrailingPadding bits
    info.sourceCount = bits.read_bits 5
    info.payloadType = bits.read_byte()  # == 203
    if info.payloadType isnt api.RTCP_PACKET_TYPE_GOODBYE
      throw new Error "payload type must be #{api.RTCP_PACKET_TYPE_GOODBYE}"
    info.wordsMinusOne = bits.read_bits 16
    info.totalBytes = (info.wordsMinusOne + 1) * 4
    info.ssrc = bits.read_bits 32

    if bits.has_more_data()
      info.reasonOctetCount = bits.read_byte()
      reason = bits.read_bytes info.reasonOctetCount

    # skip padding bytes
    readBytes = bits.current_position().byte - startBytePos
    if readBytes < info.totalBytes
      bits.skip_bytes info.totalBytes - readBytes

    return info

  readRTCPApplicationDefined: (bits) ->
    # RFC 3550 - 6.7 APP: Application-Defined RTCP Packet
    startBytePos = bits.current_position().byte
    info = {}
    info.version = bits.read_bits 2
    info.padding = bits.read_bit()
    if info.padding is 1
      api.removeTrailingPadding bits
    info.subtype = bits.read_bits 5
    info.payloadType = bits.read_byte()  # == 204
    if info.payloadType isnt api.RTCP_PACKET_TYPE_APPLICATION_DEFINED
      throw new Error "payload type must be #{api.RTCP_PACKET_TYPE_APPLICATION_DEFINED}"
    info.wordsMinusOne = bits.read_bits 16
    info.totalBytes = (info.wordsMinusOne + 1) * 4
    info.ssrc_csrc = bits.read_bits 32
    info.name = bits.read_bytes(4).toString 'ascii'

    # read the application-dependent data (remaining bytes)
    readBytes = bits.current_position().byte - startBytePos
    if readBytes < info.totalBytes
      info.applicationData = bits.read_bytes info.totalBytes - readBytes
    else
      info.applicationData = null

    return info

  readRTPFixedHeader: (bits) ->
    # RFC 3550 - 5.1 RTP Fixed Header Fields
    info = {}
    info.version = bits.read_bits 2
    info.padding = bits.read_bit()
    if info.padding is 1
      api.removeTrailingPadding bits
    info.extension = bits.read_bit()
    info.csrcCount = bits.read_bits 4
    info.marker = bits.read_bit()
    info.payloadType = bits.read_bits 7
    info.sequenceNumber = bits.read_bits 16
    info.timestamp = bits.read_bits 32
    info.ssrc = bits.read_bits 32
    info.csrc = []
    for i in [0...info.csrcCount]
      info.csrc.push bits.read_bits 32
    return info

  parseAACPacket: (buf, params) ->
    bits = new Bits buf
    packet = {}
    packet.rtpHeader = api.readRTPFixedHeader bits
    packet.aac = api.readAACPayload bits, params
    return packet

  parseH264Packet: (buf) ->
    bits = new Bits buf
    packet = {}
    packet.rtpHeader = api.readRTPFixedHeader bits
    packet.h264 = api.readH264Payload bits
    return packet

  readH264Payload: (bits) ->
    info = {}
    info.forbidden_zero_bit = bits.read_bit()  # 1 indicates error
    if info.forbidden_zero_bit isnt 0
      throw new Error "forbidden_zero_bit must be 0 (got #{info.forbidden_zero_bit})"
    info.nal_ref_idc = bits.read_bits 2  # == 00: not important, > 00: important
    info.nal_unit_type = bits.read_bits 5
    if 1 <= info.nal_unit_type <= 23  # Single NAL unit packet
      bits.push_back_byte()
      info.nal_unit = bits.remaining_buffer()
    else if 24 <= info.nal_unit_type <= 29
      switch info.nal_unit_type
        when api.H264_NAL_UNIT_TYPE_STAP_A  # STAP-A (24)
          info.stap_a = api.readH264STAP_A bits
        when api.H264_NAL_UNIT_TYPE_FU_A  # FU-A (28)
          info.fu_a = api.readH264FragmentationUnitA bits
        else
          throw new Error "Not implemented: nal_unit_type=#{info.nal_unit_type} (please report this bug)"
    else
      throw new Error "Invalid nal_unit_type=#{info.nal_unit_type}"
    return info

  # Read Single-Time Aggregation Packet type A (STAP-A)
  readH264STAP_A: (bits) ->
    info =
      nalUnits: []
    while bits.get_remaining_bytes() >= 2
      nalUnitSize = bits.read_bits 16
      info.nalUnits.push bits.read_bytes nalUnitSize
    if info.nalUnits.length < 1
      logger.error "rtp: error: STAP-A does not contain a NAL unit"
    return info

  readH264FragmentationUnitA: (bits) ->
    info = {}
    info.fuHeader = api.readH264FragmentationUnitHeader bits
    info.nal_unit_fragment = bits.remaining_buffer()
    return info

  # FU header
  readH264FragmentationUnitHeader: (bits) ->
    info = {}
    info.startBit = bits.read_bit()
    info.endBit = bits.read_bit()
    reservedBit = bits.read_bit()
    if reservedBit isnt 0
      throw new Error "reserved bit must be 0 (got #{reservedBit})"
    info.nal_unit_payload_type = bits.read_bits 5
    return info

  parsePacket: (buf) ->
    bits = new Bits buf
    packet = {}
    payloadValue = bits.get_byte_at 1  # including marker bit
    switch payloadValue
      when api.RTCP_PACKET_TYPE_SENDER_REPORT
        packet.rtcpSenderReport = api.readRTCPSenderReport bits
      when api.RTCP_PACKET_TYPE_RECEIVER_REPORT
        packet.rtcpReceiverReport = api.readRTCPReceiverReport bits
      when api.RTCP_PACKET_TYPE_SOURCE_DESCRIPTION
        packet.rtcpSourceDescription = api.readRTCPSourceDescription bits
      when api.RTCP_PACKET_TYPE_GOODBYE
        packet.rtcpGoodbye = api.readRTCPGoodbye bits
      when api.RTCP_PACKET_TYPE_APPLICATION_DEFINED
        packet.rtcpApplicationDefined = api.readRTCPApplicationDefined bits
      else  # RTP data transfer protocol - fixed header
        packet.rtpHeader = api.readRTPFixedHeader bits
    return packet

  parsePackets: (buf) ->
    bits = new Bits buf

    packets = []
    while bits.has_more_data()
      packet = {}
      payloadValue = bits.get_byte_at 1  # including marker bit
      switch payloadValue
        when api.RTCP_PACKET_TYPE_SENDER_REPORT
          packet.rtcpSenderReport = api.readRTCPSenderReport bits
        when api.RTCP_PACKET_TYPE_RECEIVER_REPORT
          packet.rtcpReceiverReport = api.readRTCPReceiverReport bits
        when api.RTCP_PACKET_TYPE_SOURCE_DESCRIPTION
          packet.rtcpSourceDescription = api.readRTCPSourceDescription bits
        when api.RTCP_PACKET_TYPE_GOODBYE
          packet.rtcpGoodbye = api.readRTCPGoodbye bits
        when api.RTCP_PACKET_TYPE_APPLICATION_DEFINED
          packet.rtcpApplicationDefined = api.readRTCPApplicationDefined bits
        else  # RTP data transfer protocol - fixed header
          packet.rtpHeader = api.readRTPFixedHeader bits
      packets.push packet

    return packets

  # Replace SSRC in-place in the given RTP header
  replaceSSRCInRTP: (buf, ssrc) ->
    buf[8]  = (ssrc >>> 24) & 0xff
    buf[9]  = (ssrc >>> 16) & 0xff
    buf[10] = (ssrc >>> 8) & 0xff
    buf[11] = ssrc & 0xff
    return

  # ntpTimestamp: [ <32-bit second part>, <32-bit fractional second part> ]
  ntpTimestampToTime: (ntpTimestamp) ->
    sec = ntpTimestamp[0] - EPOCH
    ms = ntpTimestamp[1] / NTP_SCALE_FRAC / 1000
    return sec * 1000 + ms

  # Get NTP timestamp for a time
  # time is expressed the same as Date.now()
  getNTPTimestamp: (time) ->
    sec = parseInt(time / 1000)
    ms = time - (sec * 1000)
    ntp_sec = sec + EPOCH
    ntp_usec = Math.round(ms * 1000 * NTP_SCALE_FRAC)
    return [ntp_sec, ntp_usec]

  readAACPayload: (bits, params) ->
    info = {}
    info.auHeadersLengthBits = bits.read_bits 16  # in bits
    info.numAUHeaders = info.auHeadersLengthBits / 16
    auHeaders = []
    for i in [0...info.numAUHeaders]
      params.index = i
      auHeaders.push api.readAACAUHeader bits, params
    info.auHeaders = auHeaders
    info.accessUnits = []
    for auHeader in auHeaders
      info.accessUnits.push bits.read_bytes auHeader.auSize
      accessUnit = info.accessUnits[info.accessUnits.length-1]
    return info

  readAACAUHeader: (bits, params) ->
    if not params.sizelength?
      throw new Error "sizelength is not defined in params"
    info = {}
    # size in octets of the associated Access Unit in the
    # Access Unit Data Section in the same RTP packet
    info.auSize = bits.read_bits params.sizelength
    # serial number of the associated Access Unit (fragment).
    if not params.index?
      throw new Error "index is not defined in params"
    if params.index > 0
      if not params.indexdeltalength?
        throw new Error "indexdeltalength is not defined in params"
      info.auIndexDelta = bits.read_bits params.indexdeltalength
    else
      if not params.indexlength?
        throw new Error "indexlength is not defined in params"
      info.auIndex = bits.read_bits params.indexlength
    return info

  # Used for encapsulating AAC audio data
  # opts:
  #   accessUnitLength (number): number of bytes in the access unit
  createAudioHeader: (opts) ->
    if opts.accessUnits.length > 4095
      throw new Error "too many audio access units: #{opts.accessUnits.length} (must be <= 4095)"
    numBits = opts.accessUnits.length * 16  # 2 bytes per access unit
    header = [
      ## payload
      ## See section 3.2.1 and 3.3.6 of RFC 3640 for details
      ## AU Header Section
      # AU-headers-length(16) for AAC-hbr
      # Number of bits in the AU-headers
      (numBits >> 8) & 0xff,
      numBits & 0xff,
    ]
    for accessUnit in opts.accessUnits
      header = header.concat api.createAudioAUHeader accessUnit.length
    return header

  groupAudioFrames: (adtsFrames) ->
    packetSize = RTP_HEADER_LEN
    groups = []
    currentGroup = []
    for adtsFrame, i in adtsFrames
      packetSize += adtsFrame.length + 2  # 2 bytes for AU-Header
      if packetSize > MAX_PAYLOAD_SIZE
        groups.push currentGroup
        currentGroup = []
        packetSize = RTP_HEADER_LEN + adtsFrame.length + 2
      currentGroup.push adtsFrame
    if currentGroup.length > 0
      groups.push currentGroup
    return groups

  createAudioAUHeader: (accessUnitLength) ->
    return [
      # AU Header
      # AU-size(13) by SDP
      # AU-Index(3) or AU-Index-Delta(3)
      # AU-Index is used for the first access unit, and the value must be 0.
      # AU-Index-Delta is used for the consecutive access units.
      # When interleaving is not applied, AU-Index-Delta is 0.
      accessUnitLength >> 5,
      (accessUnitLength & 0b11111) << 3,
      # There is no Auxiliary Section for AAC-hbr
    ]

  # Used for encapsulating H.264 video data
  createFragmentationUnitHeader: (opts) ->
    return [
      # Fragmentation Unit
      # See section 5.8 of RFC 6184 for details
      #
      # FU indicator
      # forbidden_zero_bit(1), nal_ref_idc(2), type(5)
      # type is 28 for FU-A
      opts.nal_ref_idc | 28,
      # FU header
      # start bit(1) == 0, end bit(1) == 1, reserved bit(1), type(5)
      (opts.isStart << 7) | (opts.isEnd << 6) | opts.nal_unit_type
    ]

  # Create RTP header
  # opts:
  #   marker (boolean): true if this is the last packet of the
  #                     access unit indicated by the RTP timestamp
  #   payloadType (number): payload type
  #   sequenceNumber (number): sequence number
  #   timestamp (number): timestamp in 90 kHz clock rate
  #   ssrc (number): SSRC (can be null)
  createRTPHeader: (opts) ->
    seqNum = opts.sequenceNumber
    ts = opts.timestamp
    ssrc = opts.ssrc ? 0
    return [
      # version(2): 2
      # padding(1): 0
      # extension(1): 0
      # CSRC count(4): 0
      0b10000000,

      # marker(1)
      # payload type(7)
      (opts.marker << 7) | opts.payloadType,

      # sequence number(16)
      seqNum >>> 8,
      seqNum & 0xff,

      # timestamp(32) in 90 kHz clock rate
      (ts >>> 24) & 0xff,
      (ts >>> 16) & 0xff,
      (ts >>> 8) & 0xff,
      ts & 0xff,

      # SSRC(32)
      (ssrc >>> 24) & 0xff,
      (ssrc >>> 16) & 0xff,
      (ssrc >>> 8) & 0xff,
      ssrc & 0xff,
    ]

  # Create RTCP BYE (Goodbye) packet
  createGoodbye: (opts) ->
    if not opts?.ssrcs?
      throw new Error "createGoodbye: ssrcs is required"
    ssrcs = opts.ssrcs
    if ssrcs.length > 0b11111
      throw new Error "createGoodbye: too many ssrcs: #{ssrcs.length} (must be <= 31)"

    # Reason for leaving
    reason = [(new Buffer 'End of stream', 'utf8')...]  # Convert Buffer to array
    reasonLen = reason.length
    # Number of bytes until the next 32-bit boundary
    padLen = 4 - (1 + reasonLen) % 4

    if reason.length > 0xff
      throw new Error "createGoodbye: reason is too long: #{reason.length} (must be <= 255)"

    # Length of this RTCP packet in 32-bit words minus one
    # including the header and any padding
    length = (4 + ssrcs.length * 4 + 1 + reasonLen + padLen) / 4 - 1

    data = [
      # See section 6.6 for details

      # version(2): 2 (RTP version 2)
      # padding(1): 0 (padding doesn't exist)
      # source count(5): number of SSRC/CSRC identifiers
      0b10000000 | ssrcs.length,

      # packet type(8): 203 (RTCP BYE)
      203,

      # length(16)
      length >> 8, length & 0xff,
    ]

    for ssrc in ssrcs
      # Append SSRC
      data.push (ssrc >>> 24) & 0xff, (ssrc >>> 16) & 0xff, (ssrc >>> 8) & 0xff, ssrc & 0xff

    data.push reason.length
    data = data.concat reason
    while padLen-- > 0
      data.push 0x00

    return data

  # Create RTCP Sender Report packet
  # opts:
  #   time: timestamp of the packet
  #   rtpTime: timestamp relative to the start point of media
  #   ssrc: SSRC
  #   packetCount: packet count
  #   octetCount: octetCount
  createSenderReport: (opts) ->
    if not opts?.ssrc?
      throw new Error "createSenderReport: ssrc is required"
    ssrc = opts.ssrc
    if not opts?.packetCount?
      throw new Error "createSenderReport: packetCount is required"
    packetCount = opts.packetCount
    if not opts?.octetCount?
      throw new Error "createSenderReport: octetCount is required"
    octetCount = opts.octetCount
    if not opts?.time?
      throw new Error "createSenderReport: time is required"
    ntp_ts = api.getNTPTimestamp opts.time
    if not opts?.rtpTime?
      throw new Error "createSenderReport: rtpTime is required"
    rtp_ts = opts.rtpTime

    length = 6  # 28 (packet bytes) / 4 (32-bit word) - 1
    return [
      # See section 6.4.1 for details

      # version(2): 2 (RTP version 2)
      # padding(1): 0 (padding doesn't exist)
      # reception report count(5): 0 (no reception report blocks)
      0b10000000,

      # packet type(8): 200 (RTCP Sender Report)
      200,

      # length(16)
      length >> 8, length & 0xff,

      # SSRC of sender(32)
      (ssrc >>> 24) & 0xff,
      (ssrc >>> 16) & 0xff,
      (ssrc >>> 8) & 0xff,
      ssrc & 0xff,

      # [sender info]
      # NTP timestamp(64)
      (ntp_ts[0] >>> 24) & 0xff,
      (ntp_ts[0] >>> 16) & 0xff,
      (ntp_ts[0] >>> 8) & 0xff,
      ntp_ts[0] & 0xff,
      (ntp_ts[1] >>> 24) & 0xff,
      (ntp_ts[1] >>> 16) & 0xff,
      (ntp_ts[1] >>> 8) & 0xff,
      ntp_ts[1] & 0xff,

      # RTP timestamp(32)
      (rtp_ts >>> 24) & 0xff,
      (rtp_ts >>> 16) & 0xff,
      (rtp_ts >>> 8) & 0xff,
      rtp_ts & 0xff,

      # sender's packet count(32)
      (packetCount >>> 24) & 0xff,
      (packetCount >>> 16) & 0xff,
      (packetCount >>> 8) & 0xff,
      packetCount & 0xff,

      # sender's octet count(32)
      (octetCount >>> 24) & 0xff,
      (octetCount >>> 16) & 0xff,
      (octetCount >>> 8) & 0xff,
      octetCount & 0xff,
    ]

  # Parse config parameter for AAC
  # see: RFC 3640, 4.1. MIME Type Registration
  parseAACConfig: (str) ->
    if str is '""'  # empty string
      return null
    buf = new Buffer str, 'hex'
    return aac.parseAudioSpecificConfig buf

module.exports = api