RFC4184
Network Working Group B. Link Request for Comments: 4184 T. Hager Category: Standards Track Dolby Laboratories
J. Flaks
Microsoft Corporation
October 2005
RTP Payload Format for AC-3 Audio
Status of This Memo
This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards" (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2005).
Abstract
This document describes an RTP payload format for transporting audio data using the AC-3 audio compression standard. AC-3 is a high quality, multichannel audio coding system that is used for United States HDTV, DVD, cable television, satellite television and other media. The RTP payload format presented in this document includes support for data fragmentation.
Contents
Introduction
AC-3 [ATSC] is a high-quality audio codec (audio coding format) designed to encode multiple channels of audio into a low bit-rate format. AC-3 achieves its large compression ratios via encoding a multiplicity of channels as a single entity. Dolby Digital, which is a branded version of AC-3, encodes up to 5.1 channels of audio.
AC-3 has been adopted as an audio compression scheme for many consumer and professional applications. It is a mandatory audio codec for DVD-video, Advanced Television Standards Committee (ATSC) digital terrestrial television and Digital Living Network Alliance (DLNA) home networking, as well as an optional multichannel audio format for DVD-audio.
There is a need to stream AC-3 data over IP networks. The Internet Real Time Protocol (RTP) provides a mechanism for stream
synchronization and hence serves as the best transport solution for AC-3, which is primarily used in audio-for-video applications. Applications for streaming AC-3 include streaming movies from a home media server to a display, video on demand, and multichannel Internet radio.
Section 2 gives a brief overview of the AC-3 algorithm. Section 3 specifies values for fields in the RTP header, while Section 4 specifies the AC-3 payload format. Section 5 discusses media types and SDP usage. Security considerations are covered in Section 6, congestion control in Section 7, and IANA considerations in Section 8. References are given in Sections 9 and 10.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 RFC2119.
Overview of AC-3
AC-3 can deliver up to 5.1 channels of audio at data rates approximately equal to half of one PCM channel [ATSC], [1994AC3], [1996AC3]. The ".1" refers to a band-limited, optional, low- frequency effects (LFE) channel. AC-3 was designed for signals sampled at rates of 32, 44.1, or 48 kHz. Data rates can vary between 32 kbps and 640 kbps, depending on the number of channels and the desired quality.
AC-3 exploits psycho-acoustic phenomena that cause a significant fraction of the information contained in a typical audio signal to be inaudible. Substantial data reduction occurs via the removal of inaudible information contained in an audio stream. Source coding techniques are further used to reduce the data rate.
Like most perceptual coders, AC-3 operates in the frequency domain. A 512-point TDAC transform is taken with 50% overlap, providing 256 new frequency samples. Frequency samples are then converted to exponents and mantissas. Exponents are differentially encoded. Mantissas are allocated a varying number of bits depending on the audibility of the associated spectral components. Audibility is determined via a masking curve. Bits for mantissas are allocated from a global bit pool.
AC-3 Bit Stream
AC-3 bit streams are organized into synchronization frames. Each AC-3 frame contains a Synchronization Information (SI) field, a Bit Stream Information (BSI) field, and 6 audio blocks (ABs) that each represent 256 PCM samples for all channels. The frame ends with an
optional auxiliary data field (AUX) and an error correction field (CRC). The entire frame represents the time duration of 1536 PCM samples across all coded channels [ATSC]. AC-3 encodes audio sampled at 32 kHz, 44.1 kHz, and 48 kHz. From Annex A, Part 2, of [ATSC], the time duration of an AC-3 frame varies with the sampling rate as follows:
Sampling rate Frame duration
_____________________________________
48 kHz 32 ms
44.1 kHz approx. 34.83 ms
32 kHz 48 ms
Figure 1 shows the AC-3 frame format.
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |SI |BSI| AB0 | AB1 | AB2 | AB3 | AB4 | AB5 |AUX|CRC| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1. AC-3 Frame Format
The Synchronization Information field contains information needed to acquire and maintain codec synchronization. The Bit Stream Information field contains parameters that describe the coded audio service [ATSC]. Each audio block contains fields that indicate the use of various coding tools: block switching, dither, coupling, and exponent strategy. They also contain metadata, optionally used to enhance the playback, such as dynamic range control. Finally, the exponents and bit allocation data needed to decode the mantissas into audio data, and the mantissas themselves, are included. The placement of these fields in an AC-3 audio block is shown in Figure 2. The fields shown in this figure are described in detail in [ATSC]. Note that field sizes vary depending on the coded data.
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Block |Dither |Dynamic |Coupling |Coupling |Exponent | | Switch |Flags |Range Ctrl |Strategy |Coordinates |Strategy | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Exponents | Bit Allocation | Mantissas | | | Parameters | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2. AC-3 Audio Block Format
RTP AC-3 Header Fields
o Payload Type (PT): The assignment of an RTP payload type for this
packet format is outside the scope of this document. It is specified by the RTP profile under which this payload format is used, or signaled dynamically out-of-band (e.g., using SDP).
o Marker (M) bit: The M bit is set to one to indicate that the RTP
packet payload contains at least one complete AC-3 frame or contains the final fragment of an AC-3 frame.
o Extension (X) bit: Defined by the RTP profile used.
o Timestamp: A 32-bit word that corresponds to the sampling instant
for the first AC-3 frame in the RTP packet. Packets containing fragments of the same frame MUST have the same time stamp. The timestamp of the first RTP packet sent SHOULD be selected at random. Thereafter, the timestamp increases linearly with the number of samples included in each frame (i.e., by 1536 for each frame).
RTP AC-3 Payload Format
This payload format is defined for AC-3, as defined in the main part and Annex D of [ATSC]. It is not defined for E-AC-3, as defined in Annex E of [ATSC], and it MUST NOT be used to carry E-AC-3.
According to RFC2736, RTP payload formats should contain an integral number of application data units (ADUs). The AC-3 frame corresponds to an ADU, in the context of this payload format. Each RTP payload MUST start with the two-byte payload header followed by an integral number of complete AC-3 frames or by a single fragment of an AC-3 frame.
If an AC-3 frame exceeds the MTU for a network, it SHOULD be fragmented for transmission within an RTP packet. Section 4.2 provides guidelines for creating frame fragments.
Payload-Specific Header
There is a two-octet Payload Header at the beginning of each payload.
Payload Header
Each AC-3 RTP payload MUST begin with the payload header shown in Figure 3.
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| MBZ | FT| NF |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 3. AC-3 RTP Payload Header
o MBZ (Must Be Zero): Bits marked MBZ SHALL be set to the value zero
and SHALL be ignored by receivers. The bits are reserved for future extensions.
o FT (Frame Type): This two-bit field indicates the type of frame(s)
present in the payload. It takes the following values:
0 - One or more complete frames.
1 - Initial fragment of frame which includes the first 5/8ths of
the frame. (See Section 4.2.)
2 - Initial fragment of frame, which does not include the first
5/8ths of the frame.
3 - Fragment of frame other than initial fragment. (Note that M
bit in RTP header is set for final fragment).
o NF (Number of frames/fragments): An 8-bit field whose meaning
depends on the Frame Type (FT) in this payload. For complete frames (FT of 0), it is used to indicate the number of AC-3 frames in the RTP payload. For frame fragments (FT of 1, 2, or 3), it is used to indicate the number fragments (and therefore packets) that make up the current frame. NF MUST be identical for packets containing fragments of the same frame.
Figure 4 shows the full AC-3 RTP payload format.
+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. +-+-+-+-+
| Payload | Frame | Frame | | Frame |
| Header | (1) | (2) | | (n) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. +-+-+-+-+
Figure 4. Full AC-3 RTP payload
When receiving AC-3 payloads with FT = 0 and more than a single frame (NF > 1), a receiver needs to use the "frmsizecod" field in the Synchronization Information (syncinfo) block in each AC-3 frame to determine the frame's length. That way a receiver can determine the boundary of the next frame. Note that the frame length may vary from frame to frame.
Fragmentation of AC-3 Frames
The size of an AC-3 frame depends on the sample rate of the audio and the data rate used by the encoder (which are indicated in the "Synchronization Information" header in the AC-3 frame). The size of a frame, for a given sample rate and data rate, is specified in Table 5.18 ("Frame Size Code Table") of [ATSC]. This table shows that AC-3 frames range in size from a minimum of 128 bytes to a maximum of 3840 bytes. If the size of an AC-3 frame exceeds the MTU size, the frame SHOULD be fragmented at the RTP level. The fragmentation MAY be performed at any byte boundary in the frame. RTP packets containing fragments of the same AC-3 frame SHALL be sent in consecutive order, from first to last fragment. This enables a receiver to assemble the fragments in correct order.
When an AC-3 frame is fragmented, it MAY be fragmented such that at least the first 5/8ths of the frame data is in the first fragment. This provides greater resilience to packet loss. This initial portion of any frame is guaranteed to contain the data necessary to decode the first two blocks of the frame. Any frame fragments other than those containing the first 5/8ths of frame data are only decodable once the complete frame is received. The 5/8ths point of the frame is defined in Table 7.34 ("5/8_frame Size Table") of [ATSC].
Types and Names
Media Type Registration
This registration uses the template defined in [DRAFT-FREED] and follows RFC 3555 RFC3555.
o Type name: audio
o Subtype name: ac3
o Required parameters:
rate: The RTP timestamp clock rate that is equal to the audio
sampling rate. Permitted rates are 32000, 44100, and 48000.
o Optional parameters:
channels: From a sender, the maximum number of channels present in
the AC3 stream. From a receiver, the maximum number of output
channels the receiver will deliver. This MUST be a number
between 1 and 6. The LFE (".1") channel MUST be counted as one
channel. Note that the channel order used in AC-3 differs from
the channel order scheme in RFC3551. The AC-3 channel order
scheme can be found in Table 5.8 of [ATSC].
ptime: See RFC 2327 RFC2327.
maxptime: See RFC 3267 RFC3267.
o Encoding considerations:
This media type is framed (see section 4.8 in [DRAFT-FREED])
and contains binary data.
o Security considerations:
See Section 6 of this document.
o Interoperability considerations:
None
o Published specification:
This payload format specification and see [ATSC].
o Applications that use this media type:
Multichannel audio compression of audio and audio for video.
o Additional Information:
Magic number(s):
The first two octets of an AC-3 frame are always the
synchronization word, which has the hex value 0x0B77.
o Person & email address to contact for further information:
Brian Link <[email protected]> IETF AVT working group.
o Intended Usage:
COMMON
o Restrictions on usage:
This media type depends on RTP framing, and hence is only
defined for transfer via RTP RFC3550. Transport within other
framing protocols is not defined at this time.
Author/Change controller:
IETF Audio/Video Transport Working Group delegated from the
IESG.
SDP Usage
The information carried in the MIME media type specification has a specific mapping to fields in the Session Description Protocol (SDP) RFC2327, which is commonly used to describe RTP sessions. When SDP is used to specify sessions employing AC-3, the mapping is as follows:
o The Media type ("audio") goes in SDP "m=" as the media name.
o The Media subtype ("ac3") goes in SDP "a=rtpmap" as the encoding
name.
o The required parameter "rate" also goes in "a=rtpmap" as the clock
rate, optionally followed by the parameter "channel".
o The optional parameters "ptime" and "maxptime" go in the SDP
"a=ptime" and "a=maxptime" attributes, respectively.
An example of the SDP data for AC-3:
m=audio 49111 RTP/AVP 100
a=rtpmap:100 ac3/48000/6
Certain considerations are needed when SDP is used to perform offer/answer exchanges RFC3264.
o The "rate" is a symmetric parameter, and the answer MUST use
the same value or remove the payload type.
o The "channels" parameter is declarative and indicates, for
recvonly or sendrecv, the desired channel configuration to
receive, and for sendonly, the intended channel configuration
to transmit. All receivers are capable of receiving any of the
defined channel configurations, and the parameter exchange
might be used to help optimize the transmission to the number
of channels the receiver requests. If the "channels" parameter
is omitted, a default maximum value of 6 is implied.
o The "ptime" and "maxptime" parameters are negotiated as defined
for "ptime" in RFC 3264 RFC3264.
Security Considerations
The payload format described in this document is subject to the security considerations defined in the RTP specification RFC3550 and in any applicable RTP profile (e.g., RFC3551). To protect the user's privacy and any copyrighted material, confidentiality protection would have to be applied. To also protect against modification by intermediate entities and ensure the authenticity of the stream, integrity protection and authentication would be required. Confidentiality, integrity protection, and authentication have to be provided by a mechanism external to this payload format, e.g., SRTP RFC3711.
The AC-3 format is designed so that the validity of data frames can determined by decoders. A decoder that encounters a malformed frame discards the malformed data and conceals the errors in the audio output until a valid frame is detected and decoded. This is expected to prevent crashes and other abnormal decoder behavior in response to errors or attacks.
Congestion Control
The general congestion control considerations for transporting RTP data apply to AC-3 audio over RTP as well. See the RTP specification RFC3550 and any applicable RTP profile (e.g., RFC3551).
AC-3 encoders may use a range of bit rates to encode audio data, so it is possible to adapt network bandwidth by adjusting the encoder bit rate in real time or by having multiple copies of content encoded at different bit rates. Additionally, packing more frames in each RTP payload can reduce the number of packets sent and hence the overhead from IP/UDP/RTP headers, at the expense of increased delay and reduced robustness against packet losses.
IANA Considerations
A new media subtype has been assigned for AC-3; see Section 5.1.
Normative References
RFC2119 Bradner, S., "Key Words for use in RFCs to Indicate
Requirement Levels", RFC 2119, March 1997.
[ATSC] U.S. Advanced Television Systems Committee (ATSC),
"ATSC Standard: Digital Audio Compression (AC-3),
Revision B," Doc A/52B, June 2005.
RFC2327 Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
RFC3550 Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
RFC3264 Rosenberg, J. and H. Schulzrinne, "An Offer/Answer
Model with Session Description Protocol (SDP)", RFC
3264, June 2002.
RFC3267 Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie,
"Real-Time Transport Protocol (RTP) Payload Format and
File Storage Format for the Adaptive Multi-Rate (AMR)
and Adaptive Multi-Rate Wideband (AMR-WB) Audio
Codecs", RFC 3267, June 2002.
RFC3555 Casner, S. and P. Hoschka, "MIME Type Registration of
RTP Payload Formats", RFC 3555, July 2003.
10. Informative References
RFC2736 Handley, M. and C. Perkins, "Guidelines for Writers of
RTP Payload Format Specifications", BCP 36, RFC 2736, December 1999.
RFC3551 Schulzrinne, H. and S. Casner, "RTP Profile for Audio
and Video Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
[1994AC3] Todd, C., et al., "AC-3: Flexible Perceptual Coding for
Audio Transmission and Storage," Preprint 3796,
Presented at the 96th Convention of the Audio
Engineering Society, May 1994.
[1996AC3] Fielder, L., et al., "AC-2 and AC-3: Low-Complexity
Transform-Based Audio Coding," Collected Papers on
Digital Audio Bit-Rate Reduction, pp. 54-72, Audio
Engineering Society, September 1996.
RFC3711 Baugher, M., et al., "The Secure Real-time Transport
Protocol (SRTP)", RFC 3711, March 2004.
[DRAFT-FREED] Freed, N. and Klensin, J., "Media Type Specifications
and Registration Procedures", Work in Progress, April
2005.
Authors' Addresses
Brian Link Dolby Laboratories 100 Potrero Ave San Francisco, CA 94103
Phone: +1 415 558 0200 EMail: [email protected]
Todd Hager Dolby Laboratories 100 Potrero Ave San Francisco, CA 94103
Phone: +1 415 558 0136 EMail: [email protected]
Jason Flaks Microsoft Corporation 1 Microsoft Way Redmond, WA 98052
Phone: +1 425 722 2543 EMail: [email protected]
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