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<rfc
  xmlns:xi="http://www.w3.org/2001/XInclude"
  category="std"
  docName="draft-ietf-mlcodec-opus-speech-coding-enhancement-02"
  ipr="trust200902"
  obsoletes=""
  updates="6716"
  submissionType="IETF"
  xml:lang="en"
  version="3">
<!-- [REPLACE] 
       * docName with name of your draft
     [CHECK] 
       * category should be one of std, bcp, info, exp, historic
       * ipr should be one of trust200902, noModificationTrust200902, noDerivativesTrust200902, pre5378Trust200902
       * updates can be an RFC number as NNNN
       * obsoletes can be an RFC number as NNNN 
-->

  <front>
    <title abbrev="Opus Speech Coding Enhancement">Integration of Speech Codec Enhancement Methods into the Opus Codec</title>
    <!--  [REPLACE/DELETE] abbrev. The abbreviated title is required if the full title is longer than 39 characters -->

    <seriesInfo name="Internet-Draft" value="draft-ietf-mlcodec-opus-speech-coding-enhancement-02"/>

    <author fullname="Jan" initials="J." role="editor" surname="Buethe">
      <organization>Meta</organization>
      <address>
        <postal>
          <country>US</country>
          <!-- Uses two letter country code -->
        </postal>
        <email>jan.buethe@googlemail.com</email> 
      </address>
    </author>

    <author fullname="Jean-Marc" initials="J.-M." surname="Valin">
      <organization>Google</organization>
      <address>
        <postal>
          <country>CA</country>
          <!-- Uses two letter country code -->
        </postal>
        <email>jmvalin@jmvalin.ca</email>
      </address>
    </author>

    <date year="2025"/>
    <!-- On draft subbmission:
         * If only the current year is specified, the current day and month will be used.
         * If the month and year are both specified and are the current ones, the current day will
           be used
         * If the year is not the current one, it is necessary to specify at least a month and day="1" will be used.
    -->

    <area>Applications and Real-Time</area>
    <workgroup>Internet Engineering Task Force</workgroup>
    <!-- "Internet Engineering Task Force" is fine for individual submissions.  If this element is 
          not present, the default is "Network Working Group", which is used by the RFC Editor as 
          a nod to the history of the RFC Series. -->

    <keyword>Opus, RFC6716</keyword>
    <!-- [REPLACE/DELETE]. Multiple allowed.  Keywords are incorporated into HTML output files for 
         use by search engines. -->

    <abstract>
      <t>This document proposes a set of requirements for integrating a speech codec enhancement method into the Opus codec <xref target="RFC6716"/></t>
    </abstract>
  </front>

  <middle>
    <section>
      <name>Introduction</name>
      <t>
        Since the specification of the original Opus codec <xref target="RFC6716"/>, new data-driven speech codec enhancement methods emerged which outperform classical
        enhancement methods by a large margin. Using such enhancement methods to improve the quality of the Opus speech codec SILK requires an update of <xref target="RFC6716"/>
        since SILK is an embedded coding mode and changing the output of the SILK decoder will lead to a violation of the Opus conformance criteria. The purpose of this document
        is hence to update <xref target="RFC6716"/> to enable the use of a speech codec enhancement algorihm. Specifically, this document defines the notion of a  SILK enhancement
        algorithm and sets forth a list of requirements, some mandatory, some optional, that aim to ensure
      </t>
      <ol type="(%d)">
        <li>consistent performance of the enhancement method itself, </li>
        <li>preservation of decoder performance (e.g. seamless mode switching), and</li>
        <li>preservation of basic interoperability when tuning the Opus encoder for use with an enhanced decoder.</li>
      </ol>
      <t>
        While the first two objectives target the Opus decoder alone, the third objective introduces new restrictions on the Opus encoder. However, these are not expected
        to interfer with any existing implementation of an Opus encoder since they target potential interoperability issues arising from new incentives connected to
        the possibility to enhance the Opus decoder.
      </t>
      <t>
        The approach of specifying requirements instead of specifying the enhancement algorithm itself has the advantage of allowing the Opus decoder to benefit from
        future improvements in a field that currently sees rapid development. Still, a description of the linear-adaptive coding enhancer (LACE) and its integration
        into the Opus decoder is included as an illustrative example for a SILK enhancement method.
      </t>

      <section>
        <name>Requirements Language</name>
        <t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL",
          "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT
          RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be
          interpreted as described in BCP 14 <xref target="RFC2119"/>
          <xref target="RFC8174"/> when, and only when, they appear in
          all capitals, as shown here.</t>
      </section>
      <!-- [CHECK] The 'Requirements Language' section is optional -->

    </section>

    <section anchor="LACE">
      <name>An Illustrative Example</name>
      <t>
        We use the linear-adaptive coding enhancer (LACE) <xref target="lace-paper"/> as an illustrative example to highlight the specific challenges of integrating a speech codec enhancement method into the Opus decoder.
         LACE is trained to enhance the output signal of the SILK decoder, the speech coding mode of Opus, and <xref target="opus-with-lace"/> depicts a high-level overview of the Opus decoder with LACE added as
         an enhancement algorithm.
      </t>
      <t>
        The first requirement for a speech coding enhancement method concerns the performance of the method itself. In this example it relates to the question how the SILK decoder output compares to the LACE output.
        In <xref target="lace-paper"/> this has been evaluated on clean speech samples using a P.808 listening test <xref target="p.808"/> as well as the objective method PESQ, which showed consistent improvement for all tested bitrates.
        For a general enhancement method it will be necessary to specify testing material and performance criteria to prevent unintended quality degradation of the Opus codec.
      </t>
      <t>
        The second requirement concerns performance of the Opus decoder as a whole. Depending on the bitstream the decoder may have to perform mode switching, e.g. between SILK and CELT, or it may combine the SILK and CELT outputs
        when the codec operates in hybrid mode. Changes to the SILK output signal by an enhancement method, such as added delay, phase shifts, or level alterations can therefore negatively impact the performance
        of the Opus decoder even if the first requirement is met. LACE solves this problem by adding no delay and by being approximately phase and level preserving. However, since many enhancement methods are non causal and
        non phase preserving, these requirements may be too strict for a general enhancement method.
      </t>

      <t>
        The third requirement concerns interoperability. The Opus specification provides significant freedom for tuning the encoder and the presence of an enhancement method in the decoder may change the optimal encoding choices
        significantly. In the present example encoding e.g. wideband content at 6 kb/s still leads to fair-to-good quality when using then LACE-enhanced decoder while the quality of a legacy decoder is significantly worse.
        To make full use of these new enhancement methods, such encoder tunings should be allowed but basic interoperability with legacy decoders or other enhanced decoders needs to be ensured.
      </t>

      <figure anchor="opus-with-lace">
        <name>A simplified Opus decoder diagram including LACE as enhancement module</name>
        <artwork type="ascii-art" name="lace.txt">
      <![CDATA[
                 ┌──────────────────────────────┐
                 │           Bitstream          │
                 └─────┬──────────────────┬─────┘
                       │                  │
                       ▼                  ▼
                 ┌───────────┐      ┌───────────┐
                 │   CELT    │      │   SILK    │
                 │  decoder  │      │  decoder  │
                 └─────┬─────┘      └─────┬─────┘
                       │                  │
                       │                  ▼
                       │            ┌───────────┐
                       │            │   LACE    │
                       │            └─────┬─────┘
                       │                  │
                       │                  ▼
                       │            ┌───────────┐
                       │            │ Resampler │
                       │            └─────┬─────┘
                       │                  │
                       ▼                  ▼
                 ┌──────────────────────────────┐
                 │        Mode Handling         │
                 └──────────────┬───────────────┘
                                │
                                ▼
                         decoded  signal
      ]]>
        </artwork>
      </figure>

      <t>
        LACE has meanwhile been superceded by the Non-Linear adaptive coding enhancer (NoLACE) <xref target="nolace-paper"/> which shares all basic properties of
        LACE outlined above but provides higher quality. This stresses the advantage of specifying requirements for an enhancement method over specifying the method itself.
      </t>
    </section>

    <section anchor="enhancement-definition">
    <name>Definition of a SILK enhancement algorithm</name>
      <t>
      A SILK enhancement algorithm denotes any algorithm that modifies or replaces the output of the SILK decoder an example of which is
      depicted in <xref target="opus-with-lace"/>. If the decoder sampling rate allows for a higher bandwidth than the encoded bandwidth,
      a SILK enhancement algorithm may also increase the bandwidth of the output signal, replacing the resampler in <xref target="opus-with-lace"/>,
      or it may modify the combination of a SILK-decoded wideband signal with a CELT decoded highband signal in hybrid mode. However, it may not modify
      the output of pure CELT frames. A SILK enhancement algorithm that extends the bandwidth of the input signal will be referred to as extending, whereas
      a SILK enhancement algorithm preserving the bandwidth of the input signal will be referred to as non-extending. Furthermore, an Opus decoder including
      a SILK enhancement algorithm will be referred to as enhanced decoder. Note however, that simply resampling the signal to a higher sampling rate is neither
      considered enhancement nor extending.
      </t>
    </section>

    <section>
    <name>Qualification of a SILK enhancement algorithm</name>

      <section>
      <name>General requirements</name>
        <section anchor="enhancement-subjective">
          <name>Subjective Evaluation</name>
          <t>
            Objective metrics for quality evaluation have often proved unreliable especially for evaluating completely new algorithms for processing speech
            or audio signals. Therefore, any SILK enhancement algorithm SHOULD undergo subjective evaluation before integration into the Opus decoder. For genuinely
            new algorithms, it is RECOMMENDED to perform either an absolute category rating (ACR) or degradation category rating (DCR)
            listening test according to <xref target="p.800"/> or <xref target="p.808"/>, where the test conditions SHOULD cover a relevant
            range of bitrates. For modifications of previously tested algorithms, e.g. changing the size of a LACE model or adding small tunings for quality improvement or
            complexity reduction, at least an informal subjective evaluation SHOULD be carried out. Any enhancement method SHOULD significantly improve quality for at least
            one encoder operating point while showing no significant degradation for other operating points.
          </t>
        </section>

        <section>
        <name>Delay and Phase considerations</name>
          <t>
          SILK is approximately phase preserving and to avoid additional delay and maintain usability for applications relying on phase information, any SILK enhancement algorithm
          SHOULD also be approximately phase preserving.
          </t>
        </section>

        <section anchor="encoder-interop">
          <name>Encoder Requirements</name>
          <t>
            The Opus specification <xref target="RFC6716"/> provides much freedom for encoding an audio signal and the presence of a powerful enhancement
            method can provide an incentive to use that freedom to produce bitstreams that, when decoded with a legacy Opus decoder, do not result in a reproduction
            of the input signal anymore. To prevent this, the following requirement is added for an Opus encoder that is designed to be used with an enhanced Opus
            decoder: if an Opus encoder produces a bitstream that can be decoded into a human-recognizable reproduction of the encoded signal with an enhanced Opus decoder, then
            that bitstream MUST also result in a human-recognizable reproduction of the encoded signal when decoded with a legacy Opus decoder.
          </t>
        </section>
      </section>

      <section>
      <name>Requirements specific to non-extending SILK enhancement algorithms for wideband speech</name>

        <section anchor="nonext-enhancement-objective">
          <name>Objective Evaluation</name>
          <t>
            Every non-extending SILK enhancement algorithm for SILK decoded wideband speech signals MUST pass all objective tests put forth in this section.
            This collection of tests is designed to uncover major failure points of the tested algorithm that could be due to improper design or training data,
            or due to improper integration into the opus decoder. It is not designed to (and cannot) assess the quality of a particular enhancement method.
          </t>

          <t>
            The tests are based on comparing a degradation score for audio samples decoded from a list of bitstreams contained in
            <eref target="https://media.xiph.org/opus/ietf/osce_testvectors_v0.zip"/> (FIXME: find final location) to a reference degradation score computed from audio
            decoded with a reference decoder. The exact reference decoder is TBD. Each test corresponds to an encoder operating point and the
            test names follow the scheme
          </t>
          <t>osce_test_BITRATE_BITRATEMODE_FRAMESIZEms_BANDWIDTH_cCOMPLEXITY_MODE</t>
          <t>where</t>
          <ol type="(%d)">
            <li>BITRATE is either a number specifying the encoder bitrate in
            bits per second or the string "SWITCHING" indicating the bitrate has been switched during encoding,</li>
            <li>BITRATEMODE is either vbr or cbr indicating variable bitrate or constant bitrate encoding,</li>
            <li>FRAMESIZE is either 10 or 20,</li>
            <li>BANDWIDTH specifies the maximal bandwidth and is always WB for this test (note however that the actual bandwidth can be lower),</li>
            <li>COMPLEXITY is a number from 0 to 10 and specifies the encoder complexity,</li>
            <li>MODE refers to the coding mode and is either "native" or "celtswitching". In "native" mode, the encoder decision whether to use SILK or CELT is
            based on signal classification whereas in "celtswitching" mode the encoder has been forced to switch between SILK and CELT at a fix rate.</li>
          </ol>
          <t>
            The testvectors are further divided into groups, where each group contains either speech samples from the same language or dialect, or music
            content. Each group GROUP is tested separately and the test is passed if it is passed for every group. The bitstreams in TESTNAME/GROUP follow
            the naming pattern CLIPNAME_TESTNAME which associates each bitstream uniquely with a reference signal reference_clips/CLIPNAME.s16. For every
            CLIPNAME in GROUP let REFMOC(CLIPNAME) denote the reference degradation score stored in the YAML <xref target="RFC9512"/> file TESTNAME/reference_scores_TESTNAME.yml
            under GROUP as primary key and CLIPNAME as secondary key. Furthermore, let CLIPNAME_test.s16 denote the signal decoded with the enhanced decoder under test
            at a sampling frequency of 16 kHz after delay compensation. The degradation for the test signal CLIPNAME_test.s16 is calculated using the moc.py tool
            <eref target="https://gitlab.xiph.org/xiph/opus/-/blob/osce-testing/dnn/torch/osce/stndrd/qualification/moc.py"/> (FIXME: moc should be implemented in C and PLC will require
            masking) with reference signal path as first argument and test signal path as second argument. The resulting degradation score will be referred to as TESTMOC(CLIPNAME).
          </t>
          <t>From the reference degradation score REFMOC(CLIPNAME) and the test degradation score TESTMOC(CLIPNAME) a difference score is calculated according to</t>

            <artwork type="ascii-art" name="moc_diff.txt">
          <![CDATA[
                    REFMOC(CLIPNAME) - TESTMOC(CLIPNAME)
      D(CLIPNAME) = ------------------------------------
                                               0.5
                        0.1 + REFMOC(CLIPNAME)
      ]]>
            </artwork>

          <t>To pass the test for group GROUP, the following two criteria MUST be met:</t>
            <ol type="(%d)">
            <li>D(CLIPNAME) is larger than A for every CLIPNAME in GROUP,</li>
            <li>The average of D(CLIPNAME) over GROUP is larger than B.</li>
          </ol>
          <t>
          The exact thresholds A and B are TBD. A test is passed if it is passed for all groups in that test.
          </t>
        </section>
        <section>
        <name>Requirements specific to extending SILK enhancement algorithms for wideband speech</name>
        <t>
          Requirements for SILK enhancement algorithms extending the bandwidth of wideband speech are TBD.
        </t>
        </section>
      </section>
    </section>

    <section anchor="IANA">
    <!-- All drafts are required to have an IANA considerations section. See RFC 8126 for a guide.-->
      <name>IANA Considerations</name>
      <t>The decoder should be able to signal the presence of an enhancement method to the encoder over SDP. The exact mechanism is TBD and the following options are
      open for discussion.
      </t>
      <ol type="(%d)">
        <li> update audio/opus media type registration <xref target="RFC7587"/> to include a parameter speech_enhancement with possible values 0 and 1</li>
        <li> assign an extension ID, e.g. 33, from the registry defined in <xref target="opus-extension"/> to implement speech coding enhancement. This has the
        advantage of a double use, meaning the extension ID can both be used to signal the decoder capability to the encoder and for transmitting side information
        to guide a speech enhancment method from the encoder to the decoder. However, it needs to be proven that side information is useful. </li>
        <li> update <xref target="opus-extension"/> to include extension IDs beyond 127 for data-less extensions </li>
      </ol>
    </section>
    
    <section anchor="Security">
      <!-- All drafts are required to have a security considerations section. See RFC 3552 for a guide. -->
      <name>Security Considerations</name>
      <t>TBD</t>
    </section>
    
    <!-- NOTE: The Acknowledgements and Contributors sections are at the end of this template -->
  </middle>

  <back>
    <references>
      <name>References</name>
      <references>
        <name>Normative References</name>
        <xi:include href="https://www.rfc-editor.org/refs/bibxml/reference.RFC.2119.xml"/>
        <xi:include href="https://www.rfc-editor.org/refs/bibxml/reference.RFC.8174.xml"/>
        <xi:include href="https://www.rfc-editor.org/refs/bibxml/reference.RFC.6716.xml"/>
        <xi:include href="https://www.rfc-editor.org/refs/bibxml/reference.RFC.7587.xml"/>
        <xi:include href="https://www.rfc-editor.org/refs/bibxml/reference.RFC.9512.xml"/>
        <!-- The recommended and simplest way to include a well known reference -->
        <reference anchor="opus-extension">
        <!-- Manually added reference -->
          <front>
            <title>Extension Formatting for the Opus Codec (draft-valin-opus-extension)</title>
            <author initials="J.-M." surname="Valin" fullname="Jean-Marc Valin">
              <organization/>
            </author>
            <date year="2023" month="April"/>
            <abstract>
              <t>Opus extension format.
              </t>
            </abstract>
          </front>
        </reference>
      </references>
 
      <references>
        <name>Informative References</name>
        <reference anchor="lace-paper">
          <front>
            <title>LACE: A light-weight, causal Model for enhancing coded Speech through Adaptive Convolutions</title>
            <author initials="J." surname="Buethe"/>
            <author initials="J.-M." surname="Valin"/>
            <author initials="A." surname="Mustafa"/>
            <date year="2023"/>
          </front>
        </reference>
        <reference anchor="nolace-paper">
          <front>
            <title>NoLACE: Improving Low-Complexity Speech Codec Enhancement Through Adaptive Temporal Shaping</title>
            <author initials="J." surname="Buethe"/>
            <author initials="A." surname="Mustafa"/>
            <author initials="J.-M." surname="Valin"/>
            <author initials="K." surname="Helwani"/>
            <author initials="M." surname="Goodwin"/>
            <date year="2024"/>
          </front>
        </reference>
        <reference anchor="p.800" target="https://www.itu.int/rec/T-REC-P.800-199608-I">
          <front>
            <title>P.800 : Methods for subjective determination of transmission quality</title>
            <author><organization abbrev="ITU-T">International Telecommuniaction Union - Telecommuniaction Standardization Sector</organization></author>
            <date month="August" year="1996"/>
          </front>
        </reference>
        <reference anchor="p.808" target="https://www.itu.int/rec/T-REC-P.808-202106-I/en">
          <front>
            <title>P.808 : Subjective evaluation of speech quality with a crowdsourcing approach</title>
            <author><organization abbrev="ITU-T">International Telecommuniaction Union - Telecommuniaction Standardization Sector</organization></author>
            <date month="June" year="2021"/>
          </front>
        </reference>
      </references>
    </references>
    

 </back>
</rfc>
