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

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<rfc category="std" docName="draft-ietf-codec-opus-update-01"
     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="4" month="September" year="2014" />

    <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 and only one of the changes results in any difference
      in the audio output.
      </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:
      <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>
     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%.
      </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:
      <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>
      <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:
<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>
    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="http://jmvalin.ca/misc_stuff/opus_update.patch"/>.
    </t>
    </section>

    <section title="Downmix to Mono">
      <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 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>