[Opus] Update to v1.2.1

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

Index: third_party/opus/src/doc/draft-ietf-codec-opus-update.xml
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+<?xml version="1.0" encoding="US-ASCII"?>
+<!DOCTYPE rfc SYSTEM "rfc2629.dtd">
+<?rfc toc="yes"?>
+<?rfc tocompact="yes"?>
+<?rfc tocdepth="3"?>
+<?rfc tocindent="yes"?>
+<?rfc symrefs="yes"?>
+<?rfc sortrefs="yes"?>
+<?rfc comments="yes"?>
+<?rfc inline="yes"?>
+<?rfc compact="yes"?>
+<?rfc subcompact="no"?>
+<rfc category="std" docName="draft-ietf-codec-opus-update-06"
+     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="19" month="June" year="2017" />
+
+    <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
+      https://opus-codec.org/.</t>
+    <t>
+      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.
+    </t>
+
+    <t>
+    Note: due to RFC formatting conventions, lines exceeding the column width
+    in the patch 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 all changes 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="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 78 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>
+    </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. Also, thanks to Jonathan Lennox and Mark Harris for their
+      feedback on this document.</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>