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AirTunes 2 Protocol
===================

.. contents:: :depth: 4

Introduction
------------

Airtunes 2 is the protocol used by Apple's Airport Express, iTunes and Rogue
Amoeba's Airfoil. The `previously known RAOP protocol
<http://xmms2.org/wiki/
Technical_note_to_describe_the_Remote_Audio_Access_Protocol_(RAOP)
_as_used_in_Apple_iTunes_to_stream_music_to_the_Airport_Express_(ApEx).>`_ does
not support more advanced timing features, and as such, some software, such as
`PulseAudio <http://www.pulseaudio.org/>`_ do not support audio/video syncing
and sometime suffer from unreliable playback. The Airtunes 2 project seeks to
remedy this.

Credits
-------

* `The Airtunes 2 Team <http://git.zx2c4.com/Airtunes2/about/>`_
* `Apple Inc. <http://www.apple.com/>`_
* `Rogue Amoeba Software, LLC <http://www.rogueamoeba.com/>`_

Conventions
-----------

In the examples below, values to be replaced are put into curly
braces ("{}"). The braces should not be included after replacing
the values.

Streaming audio to an AirTunes 2 compatible server
--------------------------------------------------

If encryption is necessary, a random key and IV (initialization vector) for AES
encryption, 16 bytes each, should be generated.

Every stream has an associated RTP timestamp (uint32; initially set to a
random number transmitted to the server in the RECORD RTSP request)
and a sequence number (int16; initially set to random value transmitted to 
the server in the RECORD RTSP request).
Both are updated when sending audio packets. Audio data is encapsulated in
RTP packets which are sent sent as UDP packets to the audio port.

There are ``TIMESTAMPS_PER_SECOND`` RTP timestamp ticks per second (equivalent
to the number of frames per second).

Up to ``PACKET_BACKLOG`` audio packets should be kept around after encoding and
encryption to resend if necessary. After sending an audio packet, the sender
should check if a sync packet should also be sent (basically every
``TIMESYNC_INTERVAL`` frames and just after connecting).


Connect
~~~~~~~

#. Establish TCP connection to RTSP port

   - IP address(es) from Zeroconf TXT record

#. Send RTSP ``OPTIONS`` request
#. Send RTSP ``ANNOUNCE`` request

   - Use password authentication based on authentication type from Zeroconf TXT
     record or after receiving HTTP status code 401 (``401 Unauthorized``)

#. Send RTSP ``SETUP`` request
#. Set sequence number of connection to a random 16 bit unsigned int and
   initial RTP timestamp to a random 32 bit unsigned int. Normal play
   time (npt) is initially 0.
#. Send RTSP ``RECORD`` request
#. Send RTSP ``SET_PARAMETER`` request to set initial volume
   (see `Setting volume`_)
#. Prepare RTP connection for audio packets


Transmitting Audio
~~~~~~~~~~~~~~~~~~

The audio data is encapsulated in an RTP packet (see RFC3550 Section 5.1)

#. Bytes 0-1 of the RTP Header are 0x80, 0x60 (0x80, 0xe0 for the 1st
   packet in a stream or after a FLUSH).
#. Bytes 2-3 store the current sequence number (whose initial value was set
   in the RECORD RTSP request)
#. Bytes 4-7 store the current RTP timestamp (initial value was transmitted
   to the Airtunes device in the RECORD RTSP request)
#. Bytes 8-11 store the SSRC (random number which will be the same in all
   audio packets, see RFC3550 Section 5.1 for more details) (TODO: is the
   SSRC the same as the RTSP client session ID ?)
#. Starting at byte 12 comes the audio data (see `Packetizing audio`_ for
   its format)
#. Send this RTP packet as UDP on the audio data port
#. Increase sequence number by one for next audio packet
#. Increase timestamp by number of frames in this audio packet


Suspending Audio Transmission
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

#. Send a ``FLUSH`` RTSP request
#. Stop sending audio packets
#. If the audio transmission is suspsending for too long (iTunes defines
   "too long" as being equal to 2 seconds), send an RTSP ``TEARDOWN`` request.


Resuming Audio Transmission
~~~~~~~~~~~~~~~~~~~~~~~~~~~

#. If a ``TEARDOWN`` request has been sent, the RTSP connection has to be
   recreated, so the steps described in ``Connect`` have to be replayed
   (the ``OPTIONS`` step can be omitted)
#. Start sending audio as if it was a new audio stream even if the
   ``Connect`` steps weren't replayed. This means the transmission will
   start with a SyncPacket marked as a "first" packet, and it will be
   followed by an Audio packet with the "first" marker too.


Disconnect
~~~~~~~~~~

#. Stop sending audio data
#. Close RTSP connection


Preferred TCP/UDP ports
-----------------------

=========== ====
Connection  Port
=========== ====
RTSP        5000
Audio data  6000
RTP control 6001
Timing      6002
=========== ====


Payload types
-------------

=============== ====
Timing request  0x52
Timing response 0x53
Sync            0x54
Range resend    0x55
Audio data      0x60
=============== ====

Data types
----------

When transferred over the network, multi-byte values need to converted
to network byte order. No aligning must be used within the packet
structures.

RtpHeader
~~~~~~~~~

::

  /* RTP header bits */
  RTP_HEADER_A_EXTENSION = 0x10;
  RTP_HEADER_A_SOURCE = 0x0f;

  RTP_HEADER_B_PAYLOAD_TYPE = 0x7f;
  RTP_HEADER_B_MARKER = 0x80;

  /* sizeof(RtpHeader) == 4 */
  RtpHeader {
   uint8_t a;
   uint8_t b;
   uint16_t seqnum;
  
   /* extension = bool(a & RTP_HEADER_A_EXTENSION) */
   /* source = a & RTP_HEADER_A_SOURCE */
  
   /* payload_type = b & RTP_HEADER_B_PAYLOAD_TYPE */
   /* marker = bool(b & RTP_HEADER_B_MARKER) */
  }


NTP Timestamp
~~~~~~~~~~~~~

This is an NTP Timestamp as described in RFC 3450 Section 4. and RFC 1305.
The timestamps used by iTunes and the device seems to come from a monotonic
clock which starts at 0 when they just started/booted. This monotonic clock's
origin of time is the unix epoch, which corresponds to 0x83aa7e80 seconds
in NTP time.

::

  /* sizeof(NtpTime) == 8 */
  struct NtpTime {
    /* Seconds since 1900-01-01 00:00:00 (TODO: Timezone?) */
    uint32_t integer;
    
    /* Fraction of second (0..2^32) */
    uint32_t fraction;
  }


RTP Timestamp
~~~~~~~~~~~~~

This is a 32 bit network order value increasing by 1 for each frame of data
transmitted, which means it increases by FRAMES_PER_PACKET for every RTP
packet sent.


TimingPacket
~~~~~~~~~~~~

::

  /* sizeof(TimingPacket) == 32 */
  struct TimingPacket {
    RtpHeader header;
    uint32_t zero_padding;
    NtpTime reference_time;
    NtpTime received_time;
    NtpTime send_time;
  }


SyncPacket
~~~~~~~~~~

::

  /* sizeof(SyncPacket) == 20 */
  struct SyncPacket {
    RtpHeader header;
    RtpTimestamp now_minus_latency;
    NtpTime time_last_sync;
    RtpTimestamp now;
  }


ResendPacket
~~~~~~~~~~~~

::

  /* sizeof(RtpResendPacket) == 8 */
  struct RtpResendHeader {
    RtpHeader header;
    uint16_t missed_seqnum;
    uint16_t count;
  }


Constants
---------

===================== ========================= ==========================
Name                  Value                     Description
===================== ========================= ==========================
FRAMES_PER_PACKET     352                       Audio frames per packet
SHORTS_PER_PACKET     2 * FRAMES_PER_PACKET     Shorts per packet
TIMESTAMPS_PER_SECOND 44100                     Timestamps per second
TIMESYNC_INTERVAL     44100                     Once per second
TIME_PER_PACKET       FRAMES_PER_PACKET / 44100 Milliseconds
PACKET_BACKLOG        1000                      Packet resend buffer size
===================== ========================= ==========================


RTSP
----

Common request headers
~~~~~~~~~~~~~~~~~~~~~~

.. _rtp-info:

================ =================================================
Client-Instance  | 64 random bytes in hex. Must be unique per
                   connection.
CSeq             | Request sequence number. Can either be counted
                   locally or response sequence number can be
                   increased by one.
RTP-Info         ``rtptime={RTP timestamp}``
Session          Server session ID (after SETUP)
User-Agent       | ``iTunes/{Version} (Windows; N;)``
                   (e.g. Version=``iTunes/7.6.2 (Windows; N;)``)
                   (for a mac, this looks like ``User-Agent: iTunes/9.2.1
                   (Macintosh; Intel Mac OS X 10.5.8) AppleWebKit/533.17.8``)
================ =================================================


Request URI
~~~~~~~~~~~

Unless specified otherwise, ``rtsp://{Local IP address}/{Client session ID}``
must be used as the request URI. The client session ID is a random number
between 0 and 2^32 generated once per connection.

Note that the Local IP address cannot be a Link Local address (i.e. it cannot
begin with 169.254) - it must be the primary IP address of the Airport, which
is the first address record listed in the Airport's RAOP service announcement.


ANNOUNCE
~~~~~~~~

======= ===========================================================
Headers | ``Content-Type: application/sdp``
Body    | ``v=0\r\n``
        | ``o=iTunes {Client session ID} O IN IP4 {Local IP address}\r\n``
        | ``s=iTunes\r\n``
        | ``c=IN IP4 {Server IP address}\r\n``
        | ``t=0 0\r\n``
        | ``m=audio 0 RTP/AVP 96\r\n``
        | ``a=rtpmap:96 AppleLossless\r\n``
        | ``a=fmtp:96 {Frames per packet} 0 16 40 10 14 2 255 0 0 44100\r\n``
        | ``a=rsaaeskey:{AES key in base64 w/o padding}\r\n``
        | ``a=aesiv:{AES IV in base64 w/o padding}\r\n``
        | ``\r\n``
======= ===========================================================

FLUSH
~~~~~

======= ===========================================================
Headers ``RTP-Info: seq={Last RTP seqnum};rtptime={Last RTP time}``
======= ===========================================================

OPTIONS
~~~~~~~

======= ============================================================
URI     ``*``
Headers ``Apple-Challenge: {16 random bytes in base64 w/o padding}``
======= ============================================================

RECORD
~~~~~~

======= =========================================
Headers | ``Range: ntp=0-``
        | ``RTP-Info: seq={Note 1};rtptime={Note 2}``
======= =========================================

Note 1: Initial value for the RTP Sequence Number, random 16 bit value

Note 2: Initial value for the RTP Timestamps, random 32 bit value

SET_PARAMETER
~~~~~~~~~~~~~

Setting volume
``````````````

======= =================================
Headers ``Content-Type: text/parameters``
Body    ``volume: %f``
======= =================================

Volume is either -144.0 (muted) or (-30.0)..(0.0).

Set progress
````````````

======= =================================
Headers ``Content-Type: text/parameters``
Body    ``progress: %f/%f/%f``
======= =================================

Values are RTP timestamp as unsigned integers (TODO).

Set DAAP metadata
`````````````````

======= =================================
Headers | ``Content-Type: application/x-dmap-tagged``
        | RTP-Info_
Body    DAAP metadata
======= =================================

SETUP
~~~~~

======= ====================================================
Headers ``Transport: RTP/AVP/UDP;unicast;interleaved=0-1;mode=record;control_port={Control port};timing_port={Timing port}``
======= ====================================================

Get ``server_port``, ``control_port`` and ``timing_port`` from ``Transport``
response header. Get ``Session`` response header and use it as server session ID.
The ``Audio-Jack-Status`` header is part of this response too.

TEARDOWN
~~~~~~~~

Nothing special.

Rogue Amoeba extensions
~~~~~~~~~~~~~~~~~~~~~~~

X_RA_SET_ALBUM_ART
``````````````````

Use this only if server wants PList metadata. Use the ``SET_PARAMETER``
method if DAAP metadata is requested.

======= ========================================
Headers | ``Content-Type: {Image content type}``
        | RTP-Info_
Body    Image data
======= ========================================


X_RA_SET_PLIST_METADATA
```````````````````````

======= ===================================
Headers | ``Content-Type: application/xml``
        | RTP-Info_
Body    Metadata in PList format
======= ===================================


Authentication
~~~~~~~~~~~~~~

AirTunes 2 uses the HTTP Digest authentication method as described
in RFC2617.


Detect speaker type
~~~~~~~~~~~~~~~~~~~

If ``Audio-Jack-Status`` is in response:

::

  speaker_type() {
    if ("disconnected" in Audio-Jack-Status) {
      return unplugged;
  
    } else if ("connected" in Audio-Jack-Status) {
      if ("digital" in Audio-Jack-Status) {
        return digital;
      }
  
      return analog;
    }
    
    return unknown;
  }


Detect metadata and audio latency
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

If ``Apple-Response``, ``Server`` or ``Audio-Latency`` in response:

::

  if (Apple-Response in response) {
    lowercase_password = False;
    audio_format = EncryptedALAC;
    wants_album_art = False;
    wants_metadata = False;
    wants_progress = False;
    has_bad_latency_header = False;
  }

  if (Server in response) {
    lowercase_password = True;
    has_bad_latency_header = True;

    if (not Apple-Response in response) {
      audio_format = UnencryptedALAC;
      wants_album_art = DAAP;
      wants_metadata = DAAP;
      wants_progress = True;
    }
  }
  
  if (Audio-Latency in response) {
    if (not has_bad_latency_header) {
      audio_latency = Audio-Latency;
    } else {
      if (Audio-Latency == 322 or
          Audio-Latency == 15049) {
        audio_latency = 11025;
      }
  
      /* Why always 11025? */
      audio_latency = 11025;
    }
  }


Timing
------

The server will send Timing request UDP packets to the timing port. The
sender will answer to this Timing request with a Timing response UDP packet
sent to the server timing port.

Replying to timing packet
~~~~~~~~~~~~~~~~~~~~~~~~~ 

::

  on_timing_packet(TimingPacket req) {
   assert req.header.payload_type == PAYLOAD_TIMING_REQUEST;
  
   TimingPacket res;
   res.header = req.header;
   res.header.payload_type = PAYLOAD_TIMING_RESPONSE;
   /* these 3 times are NTP times (ie 64 bit values) */
   res.reference_time = req.send_time;
   res.received_time = time_now();
   res.send_time = time_now();
   /* TODO: is req.send_time the time on the server? the (virtual) NTP time
    * of the audio packet that is being played by the server? Is one of the
    * other timestamp the time on the sender, and the other one, some kind of
    * "common time" to be used by the server and the sender?
    */
  
   send(res);
  }


Sync
----

Sync packets are sent once per second or when adding a speaker. They are sent
to the device control port as UDP packets. The timestamp argument corresponds
to the RTP timestamp of the next audio packet that will be sent. The latency
argument is gathered from the Audio-Latency RTSP header. First is set on the
first packet and after a FLUSH.


Sending sync packet
~~~~~~~~~~~~~~~~~~~

::
  
  send_sync(uint32_t timestamp, uint32_t latency, bool first) {
   SyncPacket packet;
   packet.header.payload_type = PAYLOAD_SYNC;
   packet.header.marker = True;
   packet.header.seqnum = 7; /* Why fixed? */
  
   if (first) {
     packet.header.extension = True;
   }

   packet.now_minus_latency = timestamp - latency;
   packet.now = timestamp;
   packet.time_last_sync = time_of_last_sync_packet;
  }


Audio
-----

Audio packet
~~~~~~~~~~~~

Header::

  /* The first 4 bytes are an RtpHeader */
  { 0x80, 0x60, 0x00, 0x00,
    0x00, 0x00, 0x00, 0x00,
    {A}, {B}, {C}, {D} }

{A}, {B}, {C}, and {D} are four random bytes that are generated per-session.


Audio codec
~~~~~~~~~~~

=============== =====================
Codec           Apple Lossless (ALAC)
Sample size     16 Bit
Channels        2
Sample rate     44100
=============== =====================


Packetizing audio
~~~~~~~~~~~~~~~~~

#. Collect ``FRAMES_PER_PACKET`` frames from input data (each frame is
   2 bytes)
#. Encode input frames using ALAC codec
#. Encode packet data

   - Raw L16
   
     #. Convert raw input data to big endian (it's an array of uint16)
     #. Copy audio header and converted audio data into one buffer
     #. Set 2nd byte of buffer to 0xa

   - Unencrypted ALAC
   
     #. Copy audio header to buffer
     #. Append ALAC encoded audio data to buffer
     
   - Encrypted ALAC
   
     #. Encrypt ALAC encoded audio data (only complete 16 byte blocks,
        the rest stays unencrypted)
     #. Copy audio header to buffer
     #. Append encrypted audio data to buffer


Metadata
--------

DAAP metadata
~~~~~~~~~~~~~

=============== =============================
Content-type    ``application/x-dmap-tagged``
Item name field ``dmap.itemname``
Artist field    ``daap.songartist``
Album field     ``daap.songalbum``
=============== =============================

PList metadata
~~~~~~~~~~~~~~

=============== =============================
Content-type    ``application/xml``
Title field     ``title``
Artist field    ``artist``
Album field     ``album``
=============== =============================


Zeroconf TXT record
-------------------

======= =======================================================
Field   Description
======= =======================================================
txtvers TXT record version (always ``1``)
pw      ``true`` if password required, ``false`` otherwise
sr      Audio sample rate
ss      Audio bit rate
ch      Number of audio channels
tp      Protocol (``UDP`` [TODO: or ``TCP``?])
======= =======================================================


Rogue Amoeba extensions
~~~~~~~~~~~~~~~~~~~~~~~

============== =======================================
Field          Description
============== =======================================
rast           ``afs`` if Airfoil speaker
ramach         ``{Platform name}.{OS major version}``
raver          Library version
raAudioFormats ``ALAC`` or ``L16``
============== =======================================