opus/doc/draft-ietf-codec-opus-update.xml

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<rfc category="std" docName="draft-ietf-codec-opus-update-04"
     ipr="trust200902">
  <front>
    <title abbrev="Opus Update">Updates to the Opus Audio Codec</title>

<author initials="JM" surname="Valin" fullname="Jean-Marc Valin">
<organization>Mozilla Corporation</organization>
<address>
<postal>
<street>331 E. Evelyn Avenue</street>
<city>Mountain View</city>
<region>CA</region>
<code>94041</code>
<country>USA</country>
</postal>
<phone>+1 650 903-0800</phone>
<email>jmvalin@jmvalin.ca</email>
</address>
</author>

<author initials="K." surname="Vos" fullname="Koen Vos">
<organization>vocTone</organization>
<address>
<postal>
<street></street>
<city></city>
<region></region>
<code></code>
<country></country>
</postal>
<phone></phone>
<email>koenvos74@gmail.com</email>
</address>
</author>



    <date day="21" month="October" year="2016" />

    <abstract>
      <t>This document addresses minor issues that were found in the specification
      of the Opus audio codec in <xref target="RFC6716">RFC 6716</xref>.</t>
    </abstract>
  </front>

  <middle>
    <section title="Introduction">
      <t>This document addresses minor issues that were discovered in the reference
      implementation of the Opus codec that serves as the specification in
      <xref target="RFC6716">RFC 6716</xref>. 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.
      </t>
    </section>

    <section title="Terminology">
      <t>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 <xref
      target="RFC2119">RFC 2119</xref>.</t>
    </section>

    <section title="Stereo State Reset in SILK">
      <t>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:
    </t>
<figure>
<artwork><![CDATA[
     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;
 }
]]></artwork>
</figure>
     <t>
     This change affects the normative part of the decoder, although the
     amount of change is too small to make a significant impact on testvectors.
      </t>
    </section>

    <section anchor="padding" title="Parsing of the Opus Packet Padding">
      <t>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:
    </t>
<figure>
<artwork><![CDATA[
       /* 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;
       }
]]></artwork>
</figure>
      <t>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 <spanx style="emph">could</spanx> 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.</t>
    </section>

    <section anchor="resampler" title="Resampler buffer">
      <t>The SILK resampler had the following issues:
        <list style="numbers">
    <t>The calls to memcpy() were using sizeof(opus_int32), but the type of the
        local buffer was opus_int16.</t>
    <t>Because the size was wrong, this potentially allowed the source
        and destination regions of the memcpy() to overlap.
          We <spanx style="emph">believe</spanx> 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.</t>
          <t>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&lt;&lt;1).</t>
      </list></t>
      <t>
      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.
    </t>
    <t>The code can be fixed by applying the following changes to line 70 of silk/resampler_private_IIR_FIR.c:
    </t>
<figure>
<artwork><![CDATA[
 )
 {
     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 ) {
             /* 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 ) );
 }
]]></artwork>
</figure>
    <t>
    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
    <eref target="https://jmvalin.ca/misc_stuff/opus_update.patch"/>. (EDITOR:
        change to an ietf.org link when ready)
    </t>
    </section>

    <section title="Integer wrap-around in inverse gain computation">
      <t>
        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:
      </t>
<figure>
<artwork><![CDATA[
         /* 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;
         }
]]></artwork>
</figure>
    </section>

    <section title="Integer wrap-around in LSF decoding">
      <t>
        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:
      </t>
<figure>
<artwork><![CDATA[
           /* 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] */
]]></artwork>
</figure>

    </section>

    <section title="Cap on Band Energy">
      <t>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:
      </t>
<figure>
<artwork><![CDATA[
       {
          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++)
]]></artwork>
</figure>
    </section>

    <section title="Hybrid Folding" anchor="folding">
      <t>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.
      </t>
      <t>
        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:
      </t>
<figure>
<artwork><![CDATA[
          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)
       {
]]></artwork>
</figure>

      <t>
       as well as line 1260:
      </t>

<figure>
<artwork><![CDATA[
          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];

]]></artwork>
</figure>
      <t>
        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.
      </t>
    </section>

    <section title="Downmix to Mono" anchor="stereo">
      <t>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.
      </t>
      <t>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).
      </t>
    </section>


    <section title="New Test Vectors">
      <t>Changes in <xref target="folding"/> and <xref target="stereo"/> 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
        <xref target="stereo"/>, while the outputs without the suffix do. An
        implementation is compliant as long as it passes either set of vectors.
      </t>
      <t>
        In addition, any Opus implementation
        that passes the original test vectors from <xref target="RFC6716">RFC 6716</xref>
        is still compliant with the Opus specification. However, newer implementations
        SHOULD be based on the new test vectors rather than the old ones.
      </t>
      <t>The new test vectors are located at
        <eref target="https://jmvalin.ca/misc_stuff/opus_newvectors.tar.gz"/>. (EDITOR:
        change to an ietf.org link when ready)
      </t>
    </section>

    <section anchor="IANA" title="IANA Considerations">
      <t>This document makes no request of IANA.</t>

      <t>Note to RFC Editor: this section may be removed on publication as an
      RFC.</t>
    </section>

    <section anchor="Acknowledgements" title="Acknowledgements">
      <t>We would like to thank Juri Aedla for reporting the issue with the parsing of
      the Opus padding.</t>
    </section>
  </middle>

  <back>
    <references title="References">
      <?rfc include="http://xml.resource.org/public/rfc/bibxml/reference.RFC.2119.xml"?>
      <?rfc include="http://xml.resource.org/public/rfc/bibxml/reference.RFC.6716.xml"?>


    </references>
  </back>
</rfc>