Network Working Group JM. Valin
Internet-Draft Octasic Inc.
Intended status: Standards Track K. Vos
Expires: January 9, 2011 Skype Technologies S.A.
July 8, 2010
Prototype for IETF Interactive Audio Codec
draft-valin-codec-prototype-01
Abstract
This document provides a quick overview of a prototype codec
combining a linear prediction layer (SILK) with an MDCT-based layer
(CELT). These codecs are used because of the authors' familiarity
with the source code, but it does not prevent inclusion of code from
other codecs as well. This is a work in progress.
Status of this Memo
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Valin & Vos Expires January 9, 2011 [Page 1]
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described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Hybrid Codec . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1. Source Code . . . . . . . . . . . . . . . . . . . . . . . 4
3. Codec Modes . . . . . . . . . . . . . . . . . . . . . . . . . 5
3.1. Examples . . . . . . . . . . . . . . . . . . . . . . . . . 6
4. Security Considerations . . . . . . . . . . . . . . . . . . . 8
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
6. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 10
7. Informative References . . . . . . . . . . . . . . . . . . . . 11
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 12
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1. Introduction
We propose a hybrid codec based on a linear prediction layer (LP) and
an MDCT-based enhancement layer. The main idea behind the proposal
is that the speech low frequencies are usually more efficiently coded
using linear prediction codecs (such as CELP variants), while the
higher frequencies are more efficiently coded in the transform domain
(e.g. MDCT). For low sampling rates, the MDCT layer is not useful
and only the LP-based layer is used. On the other hand, non-speech
signals are not always adequately coded using linear prediction, so
for music only the MDCT-based layer is used.
In this proposed prototype, the LP layer is based on the SILK [1]
codec [SILK] and the MDCT layer is based on the CELT [2] codec
[CELT]. These codecs are used because of the authors' familiarity
with the source code, but it does not prevent inclusion of code from
other codecs in the future.
This is a work in progress.
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2. Hybrid Codec
In hybrid mode, each frame is coded first by the LP layer and then by
the MDCT layer. In the current prototype, the cutoff frequency is 8
kHz. In the MDCT layer, all bands below 8 kHz are discarded, such
that there is no coding redundancy between the two layers. Also both
layers use the same instance of the range coder to encode the signal,
which ensures that no "padding bits" are wasted. The hybrid approach
makes it easy to support both constant bit-rate (CBR) and varaible
bit-rate (VBR) coding. Although the SILK layer used is VBR, it is
easy to make the bit allocation of the CELT layer produce a final
stream that is CBR by using all the bits left unused by the SILK
layer.
The implementation of SILK-based LP layer is similar to the
description in the SILK Internet-Draft [SILK] with the main exception
that SILK was modified to use the same range coder as CELT. The
implementation of the CELT-based MDCT layer is available from the
CELT website and is a more recent version (0.8.1) of the CELT
Internet-Draft [CELT]. The main changes include better support for
20 ms frames as well as the ability to encode only the higher bands
using a range coder partially filled by the SILK layer.
In addition to their frame size, the SILK and CELT codecs require a
look-ahead of 5.2 ms and 2.5 ms, respectively. SILK's look-ahead is
due to noise shaping estimation (5 ms) and the internal resampling
(0.2 ms), while CELT's look-ahead is due to the overlapping MDCT
windows. To compensate for the difference, the CELT encoder input is
delayed by 2.7 ms. This ensures that low frequencies and high
frequencies arrive at the same time.
2.1. Source Code
The source code is currently available in a Git repository [3] which
references two other repositories (for SILK and CELT). Some
snapshots are provided for convenience at
along with sample files.
Although the build system is very primitive, some instructions are
provided in the toplevel README file. This is very early development
so both the quality and feature set should greatly improve over time.
In the current version, only 48 kHz audio is supported, but support
for all configurations listed in Section 3 is planned.
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3. Codec Modes
There are three possible operating modes for the proposed prototype:
1. A linear prediction (LP) mode for use in low bit-rate connections
with up to 8 kHz audio bandwidth (16 kHz sampling rate)
2. A hybrid (LP+MDCT) mode for full-bandwidth speech at medium
bitrates
3. An MDCT-only mode for very low delay speech transmission as well
as music transmission.
Each of these modes supports a number of difference frame sizes and
sampling rates. In order to distinguish between the various modes
and configurations, we need to define a simple header that can used
in the transport layer (e.g RTP) to signal this information. The
following describes the proposed header.
The LP mode supports the following configurations (numbered from
00000...01011 in binary):
o 16 kHz: 10, 20, 40, 60 ms (00000...00011)
o 12 kHz: 10, 20, 40, 60 ms (00100...00111)
o 8 kHz: 10, 20, 40, 60 ms (01000...01011)
for a total of 12 configurations.
The hybrid mode supports the following configurations (numbered from
01100...01111):
o 48 kHz: 10, 20 ms (01100...01101)
o 32 kHz: 10, 20 ms (01110...01111)
for a total of 4 configurations.
The MDCT-only mode supports the following configurations (numbered
from 10000...11101):
o 48 kHz: 2.5, 5, 10, 20 ms (10000...10011)
o 32 kHz: 2.5, 5, 10, 20 ms (10100...10111)
o 16 kHz: 2.5, 5, 10, 20 ms (11000...11011)
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o 8 kHz: 2.5, 5, 10, 20 ms (11100...11111)
for a total of 16 configurations.
There is thus a total of 32 configurations, so 5 bits are necessary
to indicate the mode, frame size and sampling rate (MFS). This
leaves 3 bits for the number of frames per packets (codes 0 to 7):
o 0-2: 1-3 frames in the packet, each with equal compressed size
o 3: arbitrary number of frames in the packet, each with equal
compressed size (one size needs to be encoded)
o 4-5: 2-3 frames in the packet, with different compressed sizes,
which need to be encoded (except the last one)
o 6: arbitrary number of frames in the packet, with different
compressed sizes, each of which needs to be encoded
o 7: The first frame has this MFS, but others have different MFS.
Each compressed size needs to be encoded.
When code 7 is used and the last frames of a packet have the same
MFS, it is allowed to switch to another code for them.
The compressed size of the frames (if needed) is indicated -- usually
-- with one byte, with the following meaning:
o 0: No frame (DTX or lost packet)
o 1-251: Size of the frame in bytes
o 252-255: A second byte is needed. The total size is (size[1]*4)+
(size[0]%4)+252
The maximum size representable is 255*4+3+252=1275 bytes. For 20 ms
frames, that represents a bit-rate of 510 kb/s, which is really the
highest rate anyone would want to use in stereo mode (beyond that
point, lossless codecs would be more appropriate).
3.1. Examples
Simplest case: one packet
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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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| MFS |0|0|0| compressed data... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Four frames of the same compressed size:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| MFS |0|1|1| compressed data... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Two frames of different compressed size:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| MFS |1|0|1| frame size | compressed data... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Three frames of different _durations_:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 1st MFS |1|1|1| frame size | 2nd MFS |1|1|1| frame size |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 3rd MFS |1|1|1| frame size | compressed data... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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4. Security Considerations
The codec needs to take appropriate security considerations into
account, as outlined in [DOS] and [SECGUIDE]. It is extremely
important for the decoder to be robust against malicious payloads.
Malicious payloads must not cause the decoder to overrun its
allocated memory or to take much more resources to decode. Although
problems in encoders are typically rarer, the same applies to the
encoder. Malicious audio stream must not cause the encoder to
misbehave because this would allow an attacker to attack transcoding
gateways.
In its current version, this prototype codec likely does NOT meet
these security considerations, so it should be used with caution.
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5. IANA Considerations
This document has no actions for IANA.
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6. Acknowledgments
Thanks to all other developers, including Soeren Skak Jensen, Gregory
Maxwell, Christopher Montgomery, Karsten Vandborg Soerensen, and
Timothy Terriberry.
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7. Informative References
[SILK] Vos, K., Jensen, S., and K. Soerensen, "SILK Speech
Codec", draft-vos-silk-01 (work in progress), March 2010.
[CELT] Valin, J-M., Terriberry, T., Maxwell, G., and C.
Montgomery, "Constrained-Energy Lapped Transform (CELT)
Codec", draft-valin-celt-codec-02 (work in progress),
July 2010.
[DOS] Handley, M., Rescorla, E., and IAB, "Internet Denial-of-
Service Considerations", RFC 4732, December 2006.
[SECGUIDE]
Rescorla, E. and B. Korver, "Guidelines for Writing RFC
Text on Security Considerations", BCP 72, RFC 3552,
July 2003.
[1]
[2]
[3]
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Authors' Addresses
Jean-Marc Valin
Octasic Inc.
4101, Molson Street
Montreal, Quebec
Canada
Phone: +1 514 282-8858
Email: jean-marc.valin@octasic.com
Koen Vos
Skype Technologies S.A.
Stadsgaarden 6
Stockholm, 11645
SE
Phone: +46 855 921 989
Email: koen.vos@skype.net
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