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      1 <?xml version="1.0" encoding="US-ASCII"?>
      2 <!DOCTYPE rfc SYSTEM "rfc2629.dtd">
      3 <?rfc toc="yes"?>
      4 <?rfc tocompact="yes"?>
      5 <?rfc tocdepth="3"?>
      6 <?rfc tocindent="yes"?>
      7 <?rfc symrefs="yes"?>
      8 <?rfc sortrefs="yes"?>
      9 <?rfc comments="yes"?>
     10 <?rfc inline="yes"?>
     11 <?rfc compact="yes"?>
     12 <?rfc subcompact="no"?>
     13 <rfc category="std" docName="draft-ietf-codec-opus-update-06"
     14      ipr="trust200902">
     15   <front>
     16     <title abbrev="Opus Update">Updates to the Opus Audio Codec</title>
     17 
     18 <author initials="JM" surname="Valin" fullname="Jean-Marc Valin">
     19 <organization>Mozilla Corporation</organization>
     20 <address>
     21 <postal>
     22 <street>331 E. Evelyn Avenue</street>
     23 <city>Mountain View</city>
     24 <region>CA</region>
     25 <code>94041</code>
     26 <country>USA</country>
     27 </postal>
     28 <phone>+1 650 903-0800</phone>
     29 <email>jmvalin (a] jmvalin.ca</email>
     30 </address>
     31 </author>
     32 
     33 <author initials="K." surname="Vos" fullname="Koen Vos">
     34 <organization>vocTone</organization>
     35 <address>
     36 <postal>
     37 <street></street>
     38 <city></city>
     39 <region></region>
     40 <code></code>
     41 <country></country>
     42 </postal>
     43 <phone></phone>
     44 <email>koenvos74 (a] gmail.com</email>
     45 </address>
     46 </author>
     47 
     48 
     49 
     50     <date day="19" month="June" year="2017" />
     51 
     52     <abstract>
     53       <t>This document addresses minor issues that were found in the specification
     54       of the Opus audio codec in <xref target="RFC6716">RFC 6716</xref>.</t>
     55     </abstract>
     56   </front>
     57 
     58   <middle>
     59     <section title="Introduction">
     60       <t>This document addresses minor issues that were discovered in the reference
     61       implementation of the Opus codec that serves as the specification in
     62       <xref target="RFC6716">RFC 6716</xref>. Only issues affecting the decoder are
     63       listed here. An up-to-date implementation of the Opus encoder can be found at
     64       https://opus-codec.org/.</t>
     65     <t>
     66       Some of the changes in this document update normative behaviour in a way that requires
     67       new test vectors. The English text of the specification is unaffected, only
     68       the C implementation is. The updated specification remains fully compatible with
     69       the original specification.
     70     </t>
     71 
     72     <t>
     73     Note: due to RFC formatting conventions, lines exceeding the column width
     74     in the patch are split using a backslash character. The backslashes
     75     at the end of a line and the white space at the beginning
     76     of the following line are not part of the patch. A properly formatted patch
     77     including all changes is available at
     78     <eref target="https://jmvalin.ca/misc_stuff/opus_update.patch"/>. (EDITOR:
     79         change to an ietf.org link when ready)
     80     </t>
     81 
     82     </section>
     83 
     84     <section title="Terminology">
     85       <t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
     86       "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
     87       document are to be interpreted as described in <xref
     88       target="RFC2119">RFC 2119</xref>.</t>
     89     </section>
     90 
     91     <section title="Stereo State Reset in SILK">
     92       <t>The reference implementation does not reinitialize the stereo state
     93       during a mode switch. The old stereo memory can produce a brief impulse
     94       (i.e. single sample) in the decoded audio. This can be fixed by changing
     95       silk/dec_API.c at line 72:
     96     </t>
     97 <figure>
     98 <artwork><![CDATA[
     99      for( n = 0; n < DECODER_NUM_CHANNELS; n++ ) {
    100          ret  = silk_init_decoder( &channel_state[ n ] );
    101      }
    102 +    silk_memset(&((silk_decoder *)decState)->sStereo, 0,
    103 +                sizeof(((silk_decoder *)decState)->sStereo));
    104 +    /* Not strictly needed, but it's cleaner that way */
    105 +    ((silk_decoder *)decState)->prev_decode_only_middle = 0;
    106  
    107      return ret;
    108  }
    109 ]]></artwork>
    110 </figure>
    111      <t>
    112      This change affects the normative part of the decoder, although the
    113      amount of change is too small to make a significant impact on testvectors.
    114       </t>
    115     </section>
    116 
    117     <section anchor="padding" title="Parsing of the Opus Packet Padding">
    118       <t>It was discovered that some invalid packets of very large size could trigger
    119       an out-of-bounds read in the Opus packet parsing code responsible for padding.
    120       This is due to an integer overflow if the signaled padding exceeds 2^31-1 bytes
    121       (the actual packet may be smaller). The code can be fixed by applying the following
    122       changes at line 596 of src/opus_decoder.c:
    123     </t>
    124 <figure>
    125 <artwork><![CDATA[
    126        /* Padding flag is bit 6 */
    127        if (ch&0x40)
    128        {
    129 -         int padding=0;
    130           int p;
    131           do {
    132              if (len<=0)
    133                 return OPUS_INVALID_PACKET;
    134              p = *data++;
    135              len--;
    136 -            padding += p==255 ? 254: p;
    137 +            len -= p==255 ? 254: p;
    138           } while (p==255);
    139 -         len -= padding;
    140        }
    141 ]]></artwork>
    142 </figure>
    143       <t>This packet parsing issue is limited to reading memory up
    144          to about 60 kB beyond the compressed buffer. This can only be triggered
    145          by a compressed packet more than about 16 MB long, so it's not a problem
    146          for RTP. In theory, it <spanx style="emph">could</spanx> crash a file
    147          decoder (e.g. Opus in Ogg) if the memory just after the incoming packet
    148          is out-of-range, but our attempts to trigger such a crash in a production
    149          application built using an affected version of the Opus decoder failed.</t>
    150     </section>
    151 
    152     <section anchor="resampler" title="Resampler buffer">
    153       <t>The SILK resampler had the following issues:
    154         <list style="numbers">
    155     <t>The calls to memcpy() were using sizeof(opus_int32), but the type of the
    156         local buffer was opus_int16.</t>
    157     <t>Because the size was wrong, this potentially allowed the source
    158         and destination regions of the memcpy() to overlap.
    159           We <spanx style="emph">believe</spanx> that nSamplesIn is at least fs_in_khZ,
    160           which is at least 8.
    161        Since RESAMPLER_ORDER_FIR_12 is only 8, that should not be a problem once
    162        the type size is fixed.</t>
    163           <t>The size of the buffer used RESAMPLER_MAX_BATCH_SIZE_IN, but the
    164         data stored in it was actually _twice_ the input batch size
    165         (nSamplesIn&lt;&lt;1).</t>
    166       </list></t>
    167       <t>
    168       The fact that the code never produced any error in testing (including when run under the
    169       Valgrind memory debugger), suggests that in practice
    170      the batch sizes are reasonable enough that none of the issues above
    171      was ever a problem. However, proving that is non-obvious.
    172     </t>
    173     <t>The code can be fixed by applying the following changes to line 78 of silk/resampler_private_IIR_FIR.c:
    174     </t>
    175 <figure>
    176 <artwork><![CDATA[
    177  )
    178  {
    179      silk_resampler_state_struct *S = \
    180 (silk_resampler_state_struct *)SS;
    181      opus_int32 nSamplesIn;
    182      opus_int32 max_index_Q16, index_increment_Q16;
    183 -    opus_int16 buf[ RESAMPLER_MAX_BATCH_SIZE_IN + \
    184 RESAMPLER_ORDER_FIR_12 ];
    185 +    opus_int16 buf[ 2*RESAMPLER_MAX_BATCH_SIZE_IN + \
    186 RESAMPLER_ORDER_FIR_12 ];
    187  
    188      /* Copy buffered samples to start of buffer */
    189 -    silk_memcpy( buf, S->sFIR, RESAMPLER_ORDER_FIR_12 \
    190 * sizeof( opus_int32 ) );
    191 +    silk_memcpy( buf, S->sFIR, RESAMPLER_ORDER_FIR_12 \
    192 * sizeof( opus_int16 ) );
    193  
    194      /* Iterate over blocks of frameSizeIn input samples */
    195      index_increment_Q16 = S->invRatio_Q16;
    196      while( 1 ) {
    197          nSamplesIn = silk_min( inLen, S->batchSize );
    198  
    199          /* Upsample 2x */
    200          silk_resampler_private_up2_HQ( S->sIIR, &buf[ \
    201 RESAMPLER_ORDER_FIR_12 ], in, nSamplesIn );
    202  
    203          max_index_Q16 = silk_LSHIFT32( nSamplesIn, 16 + 1 \
    204 );         /* + 1 because 2x upsampling */
    205          out = silk_resampler_private_IIR_FIR_INTERPOL( out, \
    206 buf, max_index_Q16, index_increment_Q16 );
    207          in += nSamplesIn;
    208          inLen -= nSamplesIn;
    209  
    210          if( inLen > 0 ) {
    211              /* More iterations to do; copy last part of \
    212 filtered signal to beginning of buffer */
    213 -            silk_memcpy( buf, &buf[ nSamplesIn << 1 ], \
    214 RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) );
    215 +            silk_memmove( buf, &buf[ nSamplesIn << 1 ], \
    216 RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
    217          } else {
    218              break;
    219          }
    220      }
    221  
    222      /* Copy last part of filtered signal to the state for \
    223 the next call */
    224 -    silk_memcpy( S->sFIR, &buf[ nSamplesIn << 1 ], \
    225 RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) );
    226 +    silk_memcpy( S->sFIR, &buf[ nSamplesIn << 1 ], \
    227 RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
    228  }
    229 ]]></artwork>
    230 </figure>
    231     </section>
    232 
    233     <section title="Integer wrap-around in inverse gain computation">
    234       <t>
    235         It was discovered through decoder fuzzing that some bitstreams could produce
    236         integer values exceeding 32-bits in LPC_inverse_pred_gain_QA(), causing
    237         a wrap-around. Although the error is harmless in practice, the C standard considers
    238         the behavior as undefined, so the following patch to line 87 of silk/LPC_inv_pred_gain.c
    239         detects values that do not fit in a 32-bit integer and considers the corresponding filters unstable:
    240       </t>
    241 <figure>
    242 <artwork><![CDATA[
    243          /* Update AR coefficient */
    244          for( n = 0; n < k; n++ ) {
    245 -            tmp_QA = Aold_QA[ n ] - MUL32_FRAC_Q( \
    246 Aold_QA[ k - n - 1 ], rc_Q31, 31 );
    247 -            Anew_QA[ n ] = MUL32_FRAC_Q( tmp_QA, rc_mult2 , mult2Q );
    248 +            opus_int64 tmp64;
    249 +            tmp_QA = silk_SUB_SAT32( Aold_QA[ n ], MUL32_FRAC_Q( \
    250 Aold_QA[ k - n - 1 ], rc_Q31, 31 ) );
    251 +            tmp64 = silk_RSHIFT_ROUND64( silk_SMULL( tmp_QA, \
    252 rc_mult2 ), mult2Q);
    253 +            if( tmp64 > silk_int32_MAX || tmp64 < silk_int32_MIN ) {
    254 +               return 0;
    255 +            }
    256 +            Anew_QA[ n ] = ( opus_int32 )tmp64;
    257          }
    258 ]]></artwork>
    259 </figure>
    260     </section>
    261 
    262     <section title="Integer wrap-around in LSF decoding">
    263       <t>
    264         It was discovered -- also from decoder fuzzing -- that an integer wrap-around could
    265         occur when decoding line spectral frequency coefficients from extreme bitstreams.
    266         The end result of the wrap-around is an illegal read access on the stack, which
    267         the authors do not believe is exploitable but should nonetheless be fixed. The following
    268         patch to line 137 of silk/NLSF_stabilize.c prevents the problem:
    269       </t>
    270 <figure>
    271 <artwork><![CDATA[
    272            /* Keep delta_min distance between the NLSFs */
    273          for( i = 1; i < L; i++ )
    274 -            NLSF_Q15[i] = silk_max_int( NLSF_Q15[i], \
    275 NLSF_Q15[i-1] + NDeltaMin_Q15[i] );
    276 +            NLSF_Q15[i] = silk_max_int( NLSF_Q15[i], \
    277 silk_ADD_SAT16( NLSF_Q15[i-1], NDeltaMin_Q15[i] ) );
    278  
    279          /* Last NLSF should be no higher than 1 - NDeltaMin[L] */
    280 ]]></artwork>
    281 </figure>
    282 
    283     </section>
    284 
    285     <section title="Cap on Band Energy">
    286       <t>On extreme bit-streams, it is possible for log-domain band energy levels
    287         to exceed the maximum single-precision floating point value once converted
    288         to a linear scale. This would later cause the decoded values to be NaN,
    289         possibly causing problems in the software using the PCM values. This can be
    290         avoided with the following patch to line 552 of celt/quant_bands.c:
    291       </t>
    292 <figure>
    293 <artwork><![CDATA[
    294        {
    295           opus_val16 lg = ADD16(oldEBands[i+c*m->nbEBands],
    296                           SHL16((opus_val16)eMeans[i],6));
    297 +         lg = MIN32(QCONST32(32.f, 16), lg);
    298           eBands[i+c*m->nbEBands] = PSHR32(celt_exp2(lg),4);
    299        }
    300        for (;i<m->nbEBands;i++)
    301 ]]></artwork>
    302 </figure>
    303     </section>
    304 
    305     <section title="Hybrid Folding" anchor="folding">
    306       <t>When encoding in hybrid mode at low bitrate, we sometimes only have
    307         enough bits to code a single CELT band (8 - 9.6 kHz). When that happens,
    308         the second band (CELT band 18, from 9.6 to 12 kHz) cannot use folding
    309         because it is wider than the amount already coded, and falls back to
    310         LCG noise. Because it can also happen on transients (e.g. stops), it
    311         can cause audible pre-echo.
    312       </t>
    313       <t>
    314         To address the issue, we change the folding behavior so that it is
    315         never forced to fall back to LCG due to the first band not containing
    316         enough coefficients to fold onto the second band. This
    317         is achieved by simply repeating part of the first band in the folding
    318         of the second band. This changes the code in celt/bands.c around line 1237:
    319       </t>
    320 <figure>
    321 <artwork><![CDATA[
    322           b = 0;
    323        }
    324  
    325 -      if (resynth && M*eBands[i]-N >= M*eBands[start] && \
    326 (update_lowband || lowband_offset==0))
    327 +      if (resynth && (M*eBands[i]-N >= M*eBands[start] || \
    328 i==start+1) && (update_lowband || lowband_offset==0))
    329              lowband_offset = i;
    330  
    331 +      if (i == start+1)
    332 +      {
    333 +         int n1, n2;
    334 +         int offset;
    335 +         n1 = M*(eBands[start+1]-eBands[start]);
    336 +         n2 = M*(eBands[start+2]-eBands[start+1]);
    337 +         offset = M*eBands[start];
    338 +         /* Duplicate enough of the first band folding data to \
    339 be able to fold the second band.
    340 +            Copies no data for CELT-only mode. */
    341 +         OPUS_COPY(&norm[offset+n1], &norm[offset+2*n1 - n2], n2-n1);
    342 +         if (C==2)
    343 +            OPUS_COPY(&norm2[offset+n1], &norm2[offset+2*n1 - n2], \
    344 n2-n1);
    345 +      }
    346 +
    347        tf_change = tf_res[i];
    348        if (i>=m->effEBands)
    349        {
    350 ]]></artwork>
    351 </figure>
    352 
    353       <t>
    354        as well as line 1260:
    355       </t>
    356 
    357 <figure>
    358 <artwork><![CDATA[
    359           fold_start = lowband_offset;
    360           while(M*eBands[--fold_start] > effective_lowband);
    361           fold_end = lowband_offset-1;
    362 -         while(M*eBands[++fold_end] < effective_lowband+N);
    363 +         while(++fold_end < i && M*eBands[fold_end] < \
    364 effective_lowband+N);
    365           x_cm = y_cm = 0;
    366           fold_i = fold_start; do {
    367             x_cm |= collapse_masks[fold_i*C+0];
    368 
    369 ]]></artwork>
    370 </figure>
    371       <t>
    372         The fix does not impact compatibility, because the improvement does
    373         not depend on the encoder doing anything special. There is also no
    374         reasonable way for an encoder to use the original behavior to
    375         improve quality over the proposed change.
    376       </t>
    377     </section>
    378 
    379     <section title="Downmix to Mono" anchor="stereo">
    380       <t>The last issue is not strictly a bug, but it is an issue that has been reported
    381       when downmixing an Opus decoded stream to mono, whether this is done inside the decoder
    382       or as a post-processing step on the stereo decoder output. Opus intensity stereo allows
    383       optionally coding the two channels 180-degrees out of phase on a per-band basis.
    384       This provides better stereo quality than forcing the two channels to be in phase,
    385       but when the output is downmixed to mono, the energy in the affected bands is cancelled
    386       sometimes resulting in audible artefacts.
    387       </t>
    388       <t>As a work-around for this issue, the decoder MAY choose not to apply the 180-degree
    389       phase shift when the output is meant to be downmixed (inside or
    390       outside of the decoder).
    391       </t>
    392     </section>
    393 
    394 
    395     <section title="New Test Vectors">
    396       <t>Changes in <xref target="folding"/> and <xref target="stereo"/> have
    397         sufficient impact on the testvectors to make them fail. For this reason,
    398         this document also updates the Opus test vectors. The new test vectors now
    399         include two decoded outputs for the same bitstream. The outputs with
    400         suffix 'm' do not apply the CELT 180-degree phase shift as allowed in
    401         <xref target="stereo"/>, while the outputs without the suffix do. An
    402         implementation is compliant as long as it passes either set of vectors.
    403       </t>
    404       <t>
    405         In addition, any Opus implementation
    406         that passes the original test vectors from <xref target="RFC6716">RFC 6716</xref>
    407         is still compliant with the Opus specification. However, newer implementations
    408         SHOULD be based on the new test vectors rather than the old ones.
    409       </t>
    410       <t>The new test vectors are located at
    411         <eref target="https://jmvalin.ca/misc_stuff/opus_newvectors.tar.gz"/>. (EDITOR:
    412         change to an ietf.org link when ready)
    413       </t>
    414     </section>
    415 
    416     <section anchor="IANA" title="IANA Considerations">
    417       <t>This document makes no request of IANA.</t>
    418 
    419       <t>Note to RFC Editor: this section may be removed on publication as an
    420       RFC.</t>
    421     </section>
    422 
    423     <section anchor="Acknowledgements" title="Acknowledgements">
    424       <t>We would like to thank Juri Aedla for reporting the issue with the parsing of
    425       the Opus padding. Also, thanks to Jonathan Lennox and Mark Harris for their
    426       feedback on this document.</t>
    427     </section>
    428   </middle>
    429 
    430   <back>
    431     <references title="References">
    432       <?rfc include="http://xml.resource.org/public/rfc/bibxml/reference.RFC.2119.xml"?>
    433       <?rfc include="http://xml.resource.org/public/rfc/bibxml/reference.RFC.6716.xml"?>
    434 
    435 
    436     </references>
    437   </back>
    438 </rfc>
    439