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Versions: (draft-valin-codec-opus-update) 00 01 02 03 04 05

Network Working Group                                          JM. Valin
Internet-Draft                                       Mozilla Corporation
Intended status: Standards Track                                  K. Vos
Expires: June 22, 2017                                           vocTone
                                                       December 19, 2016


                    Updates to the Opus Audio Codec
                    draft-ietf-codec-opus-update-05

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
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on June 22, 2017.

Copyright Notice

   Copyright (c) 2016 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
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   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  . . . . . . . . . . . . . . . . .   3
   4.  Parsing of the Opus Packet Padding  . . . . . . . . . . . . .   3
   5.  Resampler buffer  . . . . . . . . . . . . . . . . . . . . . .   4
   6.  Integer wrap-around in inverse gain computation . . . . . . .   5
   7.  Integer wrap-around in LSF decoding . . . . . . . . . . . . .   6
   8.  Cap on Band Energy  . . . . . . . . . . . . . . . . . . . . .   6
   9.  Hybrid Folding  . . . . . . . . . . . . . . . . . . . . . . .   7
   10. Downmix to Mono . . . . . . . . . . . . . . . . . . . . . . .   9
   11. New Test Vectors  . . . . . . . . . . . . . . . . . . . . . .   9
   12. IANA Considerations . . . . . . . . . . . . . . . . . . . . .   9
   13. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   9
   14. References  . . . . . . . . . . . . . . . . . . . . . . . . .  10
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  10

1.  Introduction

   This document addresses 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 https://opus-codec.org/.

   Some of the changes in this document update normative behaviour in a
   way that requires new test vectors.  The English text of the
   specification is unaffected, only the C implementation is.  The
   updated specification remains fully compatible with the original
   specification.

   Note: due to RFC formatting conventions, lines exceeding the column
   width in the patch above are split using a backslash character.  The
   backslashes at the end of a line and the white space at the beginning
   of the following line are not part of the patch.  A properly
   formatted patch including the three changes above is available at
   <https://jmvalin.ca/misc_stuff/opus_update.patch>.  (EDITOR: change
   to an ietf.org link when ready)

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].






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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;
    }

   This change affects the normative part of the decoder, although the
   amount of change is too small to make a significant impact on
   testvectors.

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;
          }





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   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 our attempts to trigger such a crash in a production application
   built using an affected version of the Opus decoder failed.

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.

   2.  Because the size was wrong, this potentially allowed the source
       and destination regions of the memcpy() to 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 line 78 of
   silk/resampler_private_IIR_FIR.c:

    )
    {
        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 ) );



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   +    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 ) {
                /* 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.  Integer wrap-around in inverse gain computation

   It was discovered through decoder fuzzing that some bitstreams could
   produce integer values exceeding 32-bits in
   LPC_inverse_pred_gain_QA(), causing a wrap-around.  Although the
   error is harmless in practice, the C standard considers the behavior
   as undefined, so the following patch to line 87 of silk/
   LPC_inv_pred_gain.c detects values that do not fit in a 32-bit
   integer and considers the corresponding filters unstable:



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           /* Update AR coefficient */
           for( n = 0; n < k; n++ ) {
  -            tmp_QA = Aold_QA[ n ] - MUL32_FRAC_Q( \
  Aold_QA[ k - n - 1 ], rc_Q31, 31 );
  -            Anew_QA[ n ] = MUL32_FRAC_Q( tmp_QA, rc_mult2 , mult2Q );
  +            opus_int64 tmp64;
  +            tmp_QA = silk_SUB_SAT32( Aold_QA[ n ], MUL32_FRAC_Q( \
  Aold_QA[ k - n - 1 ], rc_Q31, 31 ) );
  +            tmp64 = silk_RSHIFT_ROUND64( silk_SMULL( tmp_QA, \
  rc_mult2 ), mult2Q);
  +            if( tmp64 > silk_int32_MAX || tmp64 < silk_int32_MIN ) {
  +               return 0;
  +            }
  +            Anew_QA[ n ] = ( opus_int32 )tmp64;
           }

7.  Integer wrap-around in LSF decoding

   It was discovered -- also from decoder fuzzing -- that an integer
   wrap-around could occur when decoding line spectral frequency
   coefficients from extreme bitstreams.  The end result of the wrap-
   around is an illegal read access on the stack, which the authors do
   not believe is exploitable but should nonetheless be fixed.  The
   following patch to line 137 of silk/NLSF_stabilize.c prevents the
   problem:

              /* Keep delta_min distance between the NLSFs */
            for( i = 1; i < L; i++ )
   -            NLSF_Q15[i] = silk_max_int( NLSF_Q15[i], \
   NLSF_Q15[i-1] + NDeltaMin_Q15[i] );
   +            NLSF_Q15[i] = silk_max_int( NLSF_Q15[i], \
   silk_ADD_SAT16( NLSF_Q15[i-1], NDeltaMin_Q15[i] ) );

            /* Last NLSF should be no higher than 1 - NDeltaMin[L] */

8.  Cap on Band Energy

   On extreme bit-streams, it is possible for log-domain band energy
   levels to exceed the maximum single-precision floating point value
   once converted to a linear scale.  This would later cause the decoded
   values to be NaN, possibly causing problems in the software using the
   PCM values.  This can be avoided with the following patch to line 552
   of celt/quant_bands.c:








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          {
             opus_val16 lg = ADD16(oldEBands[i+c*m->nbEBands],
                             SHL16((opus_val16)eMeans[i],6));
   +         lg = MIN32(QCONST32(32.f, 16), lg);
             eBands[i+c*m->nbEBands] = PSHR32(celt_exp2(lg),4);
          }
          for (;i<m->nbEBands;i++)

9.  Hybrid Folding

   When encoding in hybrid mode at low bitrate, we sometimes only have
   enough bits to code a single CELT band (8 - 9.6 kHz).  When that
   happens, the second band (CELT band 18, from 9.6 to 12 kHz) cannot
   use folding because it is wider than the amount already coded, and
   falls back to LCG noise.  Because it can also happen on transients
   (e.g. stops), it can cause audible pre-echo.

   To address the issue, we change the folding behavior so that it is
   never forced to fall back to LCG due to the first band not containing
   enough coefficients to fold onto the second band.  This is achieved
   by simply repeating part of the first band in the folding of the
   second band.  This changes the code in celt/bands.c around line 1237:





























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            b = 0;
         }

  -      if (resynth && M*eBands[i]-N >= M*eBands[start] && \
  (update_lowband || lowband_offset==0))
  +      if (resynth && (M*eBands[i]-N >= M*eBands[start] || \
  i==start+1) && (update_lowband || lowband_offset==0))
               lowband_offset = i;

  +      if (i == start+1)
  +      {
  +         int n1, n2;
  +         int offset;
  +         n1 = M*(eBands[start+1]-eBands[start]);
  +         n2 = M*(eBands[start+2]-eBands[start+1]);
  +         offset = M*eBands[start];
  +         /* Duplicate enough of the first band folding data to \
  be able to fold the second band.
  +            Copies no data for CELT-only mode. */
  +         OPUS_COPY(&norm[offset+n1], &norm[offset+2*n1 - n2], n2-n1);
  +         if (C==2)
  +            OPUS_COPY(&norm2[offset+n1], &norm2[offset+2*n1 - n2], \
  n2-n1);
  +      }
  +
         tf_change = tf_res[i];
         if (i>=m->effEBands)
         {

   as well as line 1260:

             fold_start = lowband_offset;
             while(M*eBands[--fold_start] > effective_lowband);
             fold_end = lowband_offset-1;
   -         while(M*eBands[++fold_end] < effective_lowband+N);
   +         while(++fold_end < i && M*eBands[fold_end] < \
   effective_lowband+N);
             x_cm = y_cm = 0;
             fold_i = fold_start; do {
               x_cm |= collapse_masks[fold_i*C+0];


   The fix does not impact compatibility, because the improvement does
   not depend on the encoder doing anything special.  There is also no
   reasonable way for an encoder to use the original behavior to improve
   quality over the proposed change.





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10.  Downmix to Mono

   The last issue is not strictly a bug, but it is an issue that has
   been reported when downmixing an Opus decoded stream to mono, whether
   this is done inside the decoder or as a post-processing step 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.

   As a work-around for this issue, the decoder MAY choose not to apply
   the 180-degree phase shift when the output is meant to be downmixed
   (inside or outside of the decoder).

11.  New Test Vectors

   Changes in Section 9 and Section 10 have sufficient impact on the
   testvectors to make them fail.  For this reason, this document also
   updates the Opus test vectors.  The new test vectors now include two
   decoded outputs for the same bitstream.  The outputs with suffix 'm'
   do not apply the CELT 180-degree phase shift as allowed in
   Section 10, while the outputs without the suffix do.  An
   implementation is compliant as long as it passes either set of
   vectors.

   In addition, any Opus implementation that passes the original test
   vectors from RFC 6716 [RFC6716] is still compliant with the Opus
   specification.  However, newer implementations SHOULD be based on the
   new test vectors rather than the old ones.

   The new test vectors are located at <https://jmvalin.ca/misc_stuff/
   opus_newvectors.tar.gz>.  (EDITOR: change to an ietf.org link when
   ready)

12.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an
   RFC.

13.  Acknowledgements

   We would like to thank Juri Aedla for reporting the issue with the
   parsing of the Opus padding.  Also, thanks to Jonathan Lennox and
   Mark Harris for their feedback on this document.



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14.  References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <http://www.rfc-editor.org/info/rfc2119>.

   [RFC6716]  Valin, JM., Vos, K., and T. Terriberry, "Definition of the
              Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
              September 2012, <http://www.rfc-editor.org/info/rfc6716>.

Authors' Addresses

   Jean-Marc Valin
   Mozilla Corporation
   331 E. Evelyn Avenue
   Mountain View, CA  94041
   USA

   Phone: +1 650 903-0800
   Email: jmvalin@jmvalin.ca


   Koen Vos
   vocTone

   Email: koenvos74@gmail.com
























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