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Versions: 00 draft-ietf-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

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
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   This Internet-Draft will expire on January 13, 2014.

Copyright Notice

   Copyright (c) 2013 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
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   described in the Simplified BSD License.




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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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