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Unified Diff: doc/draft-valin-codec-opus-update.xml

Issue 28553003: Updating Opus to a pre-release of 1.1 (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/deps/third_party/opus
Patch Set: Removing failing file Created 7 years, 2 months ago
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Index: doc/draft-valin-codec-opus-update.xml
diff --git a/doc/draft-valin-codec-opus-update.xml b/doc/draft-valin-codec-opus-update.xml
new file mode 100644
index 0000000000000000000000000000000000000000..b973e281ca1d0dae72632e5628b36c7f91802221
--- /dev/null
+++ b/doc/draft-valin-codec-opus-update.xml
@@ -0,0 +1,259 @@
+<?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-valin-codec-opus-update-00"
+ 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>650 Castro Street</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="T." surname="Terriberry" fullname="Timothy B. Terriberry">
+<organization>Mozilla Corporation</organization>
+<address>
+<postal>
+<street>650 Castro Street</street>
+<city>Mountain View</city>
+<region>CA</region>
+<code>94041</code>
+<country>USA</country>
+</postal>
+<phone>+1 650 903-0800</phone>
+<email>tterriberry@mozilla.com</email>
+</address>
+</author>
+
+<author initials="K." surname="Vos" fullname="Koen Vos">
+<organization>Skype Technologies S.A.</organization>
+<address>
+<postal>
+<street>Soder Malarstrand 43</street>
+<city>Stockholm</city>
+<region></region>
+<code>11825</code>
+<country>SE</country>
+</postal>
+<phone>+46 73 085 7619</phone>
+<email>koen.vos@skype.net</email>
+</address>
+</author>
+
+
+
+ <date day="12" month="July" year="2013" />
+
+ <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 address 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 that could not be achieved when attempted in a production
+ application built using an affected version of the Opus decoder.</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 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 like 70 of silk/resampler_private_IIR_FIR.c:
+<figure>
+<artwork><![CDATA[
+ opus_int16 out[], /* O Output signal */
+ const opus_int16 in[], /* I Input signal */
+ opus_int32 inLen /* I Number of input samples */
+ )
+ {
+ 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>
+ </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 Opus decoded stream to mono, whether this is done inside the decoder
+ or as a post-processing 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>A possible work-around for this issue would be to optionally allow the decoder to
+ not 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>
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