Internet DRAFT - draft-valin-codec-opus-update
draft-valin-codec-opus-update
Network Working Group JM. Valin
Internet-Draft T. Terriberry
Intended status: Standards Track Mozilla Corporation
Expires: January 13, 2014 K. Vos
Skype Technologies S.A.
July 12, 2013
Updates to the Opus Audio Codec
draft-valin-codec-opus-update-00
Abstract
This document addresses minor issues that were found in the
specification of the Opus audio codec in RFC 6716 [RFC6716].
Status of This Memo
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This Internet-Draft will expire on January 13, 2014.
Copyright Notice
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2
3. Stereo State Reset in SILK . . . . . . . . . . . . . . . . . 2
4. Parsing of the Opus Packet Padding . . . . . . . . . . . . . 3
5. Resampler buffer . . . . . . . . . . . . . . . . . . . . . . 3
6. Downmix to Mono . . . . . . . . . . . . . . . . . . . . . . . 5
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 5
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 5
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 5
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 6
1. Introduction
This document address minor issues that were discovered in the
reference implementation of the Opus codec that serves as the
specification in RFC 6716 [RFC6716]. Only issues affecting the
decoder are listed here. An up-to-date implementation of the Opus
encoder can be found at http://opus-codec.org/. The updated
specification remains fully compatible with the original
specification and only one of the changes results in any difference
in the audio output.
2. Terminology
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].
3. Stereo State Reset in SILK
The reference implementation does not reinitialize the stereo state
during a mode switch. The old stereo memory can produce a brief
impulse (i.e. single sample) in the decoded audio. This can be
fixed by changing silk/dec_API.c at line 72:
for( n = 0; n < DECODER_NUM_CHANNELS; n++ ) {
ret = silk_init_decoder( &channel_state[ n ] );
}
+ silk_memset(&((silk_decoder *)decState)->sStereo, 0,
+ sizeof(((silk_decoder *)decState)->sStereo));
+ /* Not strictly needed, but it's cleaner that way */
+ ((silk_decoder *)decState)->prev_decode_only_middle = 0;
return ret;
}
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This change affects the normative part of the decoder. Fortunately,
the modified decoder is still compliant with the original
specification because it still easily passes the testvectors. For
example, for the float decoder at 48 kHz, the opus_compare
(arbitrary) "quality score" changes from from 99.9333% to 99.925%.
4. Parsing of the Opus Packet Padding
It was discovered that some invalid packets of very large size could
trigger an out-of-bounds read in the Opus packet parsing code
responsible for padding. This is due to an integer overflow if the
signaled padding exceeds 2^31-1 bytes (the actual packet may be
smaller). The code can be fixed by applying the following changes at
line 596 of src/opus_decoder.c:
/* Padding flag is bit 6 */
if (ch&0x40)
{
- int padding=0;
int p;
do {
if (len<=0)
return OPUS_INVALID_PACKET;
p = *data++;
len--;
- padding += p==255 ? 254: p;
+ len -= p==255 ? 254: p;
} while (p==255);
- len -= padding;
}
This packet parsing issue is limited to reading memory up to about 60
kB beyond the compressed buffer. This can only be triggered by a
compressed packet more than about 16 MB long, so it's not a problem
for RTP. In theory, it _could_ crash a file decoder (e.g. Opus in
Ogg) if the memory just after the incoming packet is out-of-range,
but that could not be achieved when attempted in a production
application built using an affected version of the Opus decoder.
5. Resampler buffer
The SILK resampler had the following issues:
1. The calls to memcpy() were using sizeof(opus_int32), but the type
of the local buffer was opus_int16.
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2. Because the size was wrong, this potentially allowed the source
and destination regions of the memcpy overlap. We _believe_ that
nSamplesIn is at least fs_in_khZ, which is at least 8. Since
RESAMPLER_ORDER_FIR_12 is only 8,that should not be a problem
once the type size is fixed.
3. The size of the buffer used RESAMPLER_MAX_BATCH_SIZE_IN, but the
data stored in it was actually _twice_ the input batch size
(nSamplesIn<<1).
The fact that the code never produced any error in testing (including
when run under the Valgrind memory debugger), suggests that in
practice the batch sizes are reasonable enough that none of the
issues above was ever a problem. However, proving that is non-
obvious.
The code can be fixed by applying the following changes to like 70 of
silk/resampler_private_IIR_FIR.c:
opus_int16 out[], /* O Output signal */
const opus_int16 in[], /* I Input signal */
opus_int32 inLen /* I Number of input samples */
)
{
silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS;
opus_int32 nSamplesIn;
opus_int32 max_index_Q16, index_increment_Q16;
- opus_int16 buf[ RESAMPLER_MAX_BATCH_SIZE_IN + RESAMPLER_ORDER_FIR_12 ];
+ opus_int16 buf[ 2*RESAMPLER_MAX_BATCH_SIZE_IN + RESAMPLER_ORDER_FIR_12 ];
/* Copy buffered samples to start of buffer */
- silk_memcpy( buf, S->sFIR, RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) );
+ silk_memcpy( buf, S->sFIR, RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
/* Iterate over blocks of frameSizeIn input samples */
index_increment_Q16 = S->invRatio_Q16;
while( 1 ) {
nSamplesIn = silk_min( inLen, S->batchSize );
/* Upsample 2x */
silk_resampler_private_up2_HQ( S->sIIR, &buf[ RESAMPLER_ORDER_FIR_12 ], in, nSamplesIn );
max_index_Q16 = silk_LSHIFT32( nSamplesIn, 16 + 1 ); /* + 1 because 2x upsampling */
out = silk_resampler_private_IIR_FIR_INTERPOL( out, buf, max_index_Q16, index_increment_Q16 );
in += nSamplesIn;
inLen -= nSamplesIn;
if( inLen > 0 ) {
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/* More iterations to do; copy last part of filtered signal to beginning of buffer */
- silk_memcpy( buf, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) );
+ silk_memmove( buf, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
} else {
break;
}
}
/* Copy last part of filtered signal to the state for the next call */
- silk_memcpy( S->sFIR, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) );
+ silk_memcpy( S->sFIR, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
}
6. Downmix to Mono
The last issue is not strictly a bug, but it is an issue that has
been reported when downmixing Opus decoded stream to mono, whether
this is done inside the decoder or as a post-processing on the stereo
decoder output. Opus intensity stereo allows optionally coding the
two channels 180-degrees out of phase on a per-band basis. This
provides better stereo quality than forcing the two channels to be in
phase, but when the output is downmixed to mono, the energy in the
affected bands is cancelled sometimes resulting in audible artefacts.
A possible work-around for this issue would be to optionally allow
the decoder to not apply the 180-degree phase shift when the output
is meant to be downmixed (inside or outside of the decoder).
7. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
8. Acknowledgements
We would like to thank Juri Aedla for reporting the issue with the
parsing of the Opus padding.
9. References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, September 2012.
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Authors' Addresses
Jean-Marc Valin
Mozilla Corporation
650 Castro Street
Mountain View, CA 94041
USA
Phone: +1 650 903-0800
Email: jmvalin@jmvalin.ca
Timothy B. Terriberry
Mozilla Corporation
650 Castro Street
Mountain View, CA 94041
USA
Phone: +1 650 903-0800
Email: tterriberry@mozilla.com
Koen Vos
Skype Technologies S.A.
Soder Malarstrand 43
Stockholm 11825
SE
Phone: +46 73 085 7619
Email: koen.vos@skype.net
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