Internet-Draft Opus Speech Coding Enhancement July 2026
Buethe & Valin Expires 22 January 2027 [Page]
Workgroup:
Machine Learning for Audio Coding
Internet-Draft:
draft-ietf-mlcodec-opus-speech-coding-enhancement-04
Updates:
6716 (if approved)
Published:
Intended Status:
Standards Track
Expires:
Authors:
J. Buethe, Ed.
Meta Platforms Inc.
J.-M. Valin
Google

Integration of Speech Codec Enhancement Algorithms into the Opus Codec

Abstract

This document proposes a set of requirements for integrating a speech codec enhancement algorithm into the Opus codec [RFC6716].

Status of This Memo

This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.

Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at https://datatracker.ietf.org/drafts/current/.

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This Internet-Draft will expire on 2 January 2027.

Table of Contents

1. Introduction

Since the specification of the original Opus codec [RFC6716], new data-driven speech codec enhancement algorithms have emerged that outperform classical enhancement algorithms by a large margin. Using such enhancement algorithms to improve the quality of the Opus speech codec SILK requires an update of [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 [RFC6716] to enable the use of a speech codec enhancement algorithm. 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

(1)
consistent performance of the enhancement algorithm itself,
(2)
preservation of decoder performance (e.g. seamless mode switching), and
(3)
preservation of basic interoperability when tuning the Opus encoder for use with an enhanced decoder.

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 interfere 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.

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 algorithm.

1.1. Requirements Language

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 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.

2. An Illustrative Example

We use the linear-adaptive coding enhancer (LACE) [lace-paper] as an illustrative example to highlight the specific challenges of integrating a speech codec enhancement algorithm into the Opus decoder. LACE is trained to enhance the output signal of the SILK decoder, the speech coding mode of Opus, and Figure 1 depicts a high-level overview of the Opus decoder with LACE added as an enhancement algorithm.

The first requirement for a speech coding enhancement algorithm concerns the performance of the algorithm itself. In this example it relates to the question of how the SILK decoder output compares to the LACE output. In [lace-paper] this has been evaluated on clean speech samples using a P.808 listening test [p.808] as well as the objective method PESQ, which showed consistent improvement for all tested bitrates. For a general enhancement algorithm it will be necessary to specify testing material and performance criteria to prevent unintended quality degradation of the Opus codec.

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 algorithm, 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 algorithms are non-causal and non-phase-preserving, these requirements may be too strict for a general enhancement algorithm.

The third requirement concerns interoperability. The Opus specification provides significant freedom for tuning the encoder and the presence of an enhancement algorithm 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 the LACE-enhanced decoder while the quality of a legacy decoder is significantly worse. To make full use of these new enhancement algorithms, such encoder tunings should be allowed but basic interoperability with legacy decoders or other enhanced decoders needs to be ensured.


                 ┌──────────────────────────────┐
                 │           Bitstream          │
                 └─────┬──────────────────┬─────┘
                       │                  │
                       ▼                  ▼
                 ┌───────────┐      ┌───────────┐
                 │   CELT    │      │   SILK    │
                 │  decoder  │      │  decoder  │
                 └─────┬─────┘      └─────┬─────┘
                       │                  │
                       │                  ▼
                       │            ┌───────────┐
                       │            │   LACE    │
                       │            └─────┬─────┘
                       │                  │
                       │                  ▼
                       │            ┌───────────┐
                       │            │ Resampler │
                       │            └─────┬─────┘
                       │                  │
                       ▼                  ▼
                 ┌──────────────────────────────┐
                 │        Mode Handling         │
                 └──────────────┬───────────────┘
                                │
                                ▼
                         decoded  signal

Figure 1: A simplified Opus decoder diagram including LACE as enhancement module

LACE has meanwhile been superseded by the non-linear adaptive coding enhancer (NoLACE) [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 algorithm over specifying the algorithm itself.

3. Definition of a SILK Enhancement Algorithm

A SILK enhancement algorithm denotes any algorithm that replaces the native SILK decoded output with an enhanced output signal, an example of which is depicted in Figure 1. Here, the native SILK decoded output refers to the signal that an unmodified SILK decoder would produce. A SILK enhancement algorithm is not restricted to operating on the native SILK decoded output alone; it may equally draw on internal quantities of the SILK decoder (such as gains, LSFs, pitch or excitation), modify such internal values, or replace parts of the SILK decoder altogether. The requirements in this document therefore constrain only the relation between the native SILK decoded output and the output of the enhancement algorithm. If the encoded content is wideband and the decoder sampling rate allows for a higher bandwidth than wideband, a SILK enhancement algorithm may also produce an output signal of higher bandwidth than the native SILK decoded output, replacing the resampler in Figure 1, or it may modify the combination of a SILK-decoded wideband signal with a CELT decoded highband signal in hybrid mode. Such bandwidth extension is only permitted for wideband speech; narrowband and mediumband content MUST NOT be extended. In any case, a SILK enhancement algorithm MUST NOT modify the output of pure CELT frames. A SILK enhancement algorithm whose output has a higher bandwidth than the native SILK decoded output will be referred to as extending, whereas a SILK enhancement algorithm whose output preserves the bandwidth of the native SILK decoded output 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 enhancing nor extending.

4. Qualification of a SILK Enhancement Algorithm

4.1. General Requirements

4.1.1. Subjective Evaluation

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 [p.800] or [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 algorithm SHOULD significantly improve quality for at least one encoder operating point while showing no significant degradation for other operating points.

4.1.2. Delay and Phase Considerations

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.

4.1.3. Encoder Requirements

The Opus specification [RFC6716] provides much freedom for encoding an audio signal and the presence of a powerful enhancement algorithm 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.

4.1.4. Training Data

To keep the objective tests meaningful, an enhancement algorithm MUST NOT be trained on any of the clips used in the tests of this document, and SHOULD NOT be trained on the datasets from which those clips are drawn (the Common Voice dataset for the wideband speech clips and additionally the VCTK corpus for the fullband speech clips).

4.2. Requirements Specific to Non-Extending SILK Enhancement Algorithms for Wideband Speech

4.2.1. Lowband Test (Non-Extending)

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 algorithm.

The tests are based on comparing a degradation score for audio samples decoded from a list of bitstreams contained in https://media.xiph.org/opus/ietf/osce_testvectors_v1.zip (FIXME: find final location) to a reference degradation score computed from audio decoded with a reference decoder. Any conforming Opus decoder [RFC6716] is admissible as a reference decoder. This latitude is deliberate and necessary: decoding a bitstream may involve behavior that is not fully specified by [RFC6716], such as packet loss concealment, so different conforming decoders may produce slightly different reference scores. Each test corresponds to an encoder operating point and the test names follow the scheme

osce_test_BITRATE_BITRATEMODE_FRAMESIZEms_BANDWIDTH_cCOMPLEXITY_MODE

where

(1)
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,
(2)
BITRATEMODE is either vbr or cbr indicating variable bitrate or constant bitrate encoding,
(3)
FRAMESIZE is either 10 or 20,
(4)
BANDWIDTH specifies the maximal bandwidth and is always WB for this test (note however that the actual bandwidth can be lower),
(5)
COMPLEXITY is a number from 0 to 10 and specifies the encoder complexity,
(6)
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 fixed rate.

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, obtained by decoding the bitstream with a reference decoder and scoring the result against reference_clips/CLIPNAME.s16 in the same way as the test signal below. 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 osce_compare tool https://gitlab.xiph.org/xiph/opus/-/blob/osce-testing/src/osce_compare.c with the reference signal path as first argument and the test signal path as second argument. The resulting degradation score will be referred to as TESTMOC(CLIPNAME).

All degradation and distortion scores in this document MUST be computed on time-aligned signals: before scoring, the decoder delay MUST be compensated so that the decoded signal is sample-aligned with the reference signal reference_clips/CLIPNAME.s16. The osce_compare tool performs this compensation by discarding a fixed number of leading samples from the decoded (second-argument) signal, given through its -delay option. The decoder delay depends on the Opus implementation and its configuration, and, for an enhanced decoder, on the SILK enhancement algorithm; it may therefore differ between the reference decoder and the decoder under test as well as between sampling rates. The applicable delay MUST be determined for each decoder and sampling rate used, since residual misalignment degrades the scores and can invalidate the test result.

From the reference degradation score REFMOC(CLIPNAME) and the test degradation score TESTMOC(CLIPNAME) a difference score is calculated according to


                    REFMOC(CLIPNAME) - TESTMOC(CLIPNAME)
      D(CLIPNAME) = ------------------------------------
                                               0.5
                        0.1 + REFMOC(CLIPNAME)

To pass the test for group GROUP, the following two criteria MUST be met:

(1)
D(CLIPNAME) is larger than A for every CLIPNAME in GROUP,
(2)
The average of D(CLIPNAME) over GROUP is larger than B.

The thresholds are A = -0.5 and B = -0.052. A test is passed if it is passed for all groups in that test.

4.3. Requirements Specific to Extending SILK Enhancement Algorithms for Wideband Speech

An extending SILK enhancement algorithm extends the bandwidth of the decoded speech beyond the 8 kHz upper limit of wideband. The objective tests in this section therefore evaluate the decoder output at a sampling frequency of 48 kHz. The blind bandwidth extension network (BBWENet) [bbwenet-paper] is an example of an extending SILK enhancement algorithm.

An extending algorithm is subject to two objective tests:

  • a lowband test, which ensures that the quality of the 0-8 kHz lowband is not degraded relative to an unenhanced decoder; and
  • a highband test, which checks that the extended signal has a closer per-band spectral distortion to the fullband reference than the unenhanced wideband-decoded signal has to that reference.

The lowband test is the objective test of Section 4.2.1 applied to the 48 kHz output; a non-extending algorithm is subject only to that test. An extending algorithm MUST pass the lowband test.

Bandwidth extension is blind: the highband content above 8 kHz is not present in the native SILK decoded output, and many plausible highband extensions exist, some of which may violate the highband criterion without being perceptually wrong. The highband test is therefore relaxed in two ways: the criterion is only required to hold for most of the test clips rather than every clip, and an extending algorithm SHOULD, rather than MUST, pass the highband test. Subjective evaluation as described in Section 4.1.1 remains the primary means of assessing the highband of an extending algorithm.

4.3.1. Lowband Test (Extending)

The lowband test reuses the testvectors, naming scheme, grouping, difference score D(CLIPNAME) and group pass criteria of Section 4.2.1, with the following modifications:

(1)
the reference decode (yielding REFMOC(CLIPNAME)) is produced by decoding the bitstream with the reference decoder at a sampling frequency of 48 kHz instead of at 16 kHz;
(2)
the test signal (yielding TESTMOC(CLIPNAME)) is produced by decoding the bitstream with the enhanced decoder under test, i.e. the decoder applying the extending enhancement algorithm, at a sampling frequency of 48 kHz;
(3)
both degradation scores are computed on the 0-8 kHz lowband only.

Both degradation scores are obtained with the osce_compare tool https://gitlab.xiph.org/xiph/opus/-/blob/osce-testing/src/osce_compare.c, invoked with the 16 kHz reference signal reference_clips/CLIPNAME.s16 as first argument and the respective 48 kHz decoded signal as second argument, with the reference and test sampling rates set accordingly (-fs_ref 16000 -fs_test 48000) and with the applicable delay compensation (-delay). Given these rates, osce_compare reduces the 48 kHz decoded signal to the 0-8 kHz lowband before scoring it against the reference signal. Decoding both the reference decode and the test signal at 48 kHz ensures that the 16 kHz to 48 kHz resampling effects on the lowband cancel between the two in the difference score D(CLIPNAME).

In all other respects the lowband test is identical to Section 4.2.1. An extending algorithm MUST pass the lowband test, i.e. it MUST NOT degrade the wideband content.

4.3.2. Highband Test

The highband test verifies that the algorithm adds meaningful signal energy above 8 kHz and does not degrade the highband. It is based on a separate set of fullband (48 kHz) speech reference clips contained in https://media.xiph.org/opus/ietf/osce_testvectors_v1.zip (FIXME: find final location) under the highband subdirectory. Each reference clip is encoded as wideband and the resulting bitstream is decoded with the extending decoder under test at a sampling frequency of 48 kHz. Each test corresponds to an encoder operating point and the test names follow the scheme

osce_hbtest_BITRATE_BITRATEMODE_FRAMESIZEms_BANDWIDTH_cCOMPLEXITY_MODE

with the fields defined as in Section 4.2.1. The bitstreams in TESTNAME follow the naming pattern CLIPNAME_TESTNAME which associates each bitstream uniquely with a fullband reference signal highband/reference_clips/CLIPNAME.s16.

For a decoded test signal and its fullband reference, a per-band spectral distortion DIST(reference, test) is computed for each of the four highbands spanning 8.0-9.6, 9.6-12.0, 12.0-15.6 and 15.6-20.0 kHz, together with the distortion DIST(reference, anchor) of a lowpass anchor. The lowpass anchor is the "no extension" signal and is derived from the fullband reference by zeroing out all STFT bins at and above 8 kHz. The exact computation is defined by the osce_compare tool https://gitlab.xiph.org/xiph/opus/-/blob/osce-testing/src/osce_compare.c, invoked with its -highband option. A clip passes when, in every one of the four highbands BAND, the extended signal is at least as close to the reference as the anchor:


      DIST(reference, test)(BAND) <= DIST(reference, anchor)(BAND)

As with the lowband tests (see Section 4.2.1), the decoded test signal MUST be delay-compensated so that it is sample-aligned with the fullband reference highband/reference_clips/CLIPNAME.s16 before the distortions are computed. The osce_compare tool applies this compensation through its -delay option at the 48 kHz test sampling rate; the applicable decoder delay depends on the Opus implementation and the enhancement algorithm and may differ between decoders.

For each test (encoder operating point) the pass rate is the fraction of clips that pass, and the test is passed if more than 90% of the clips pass. The highband test is only meaningful when the lowband quality is sufficiently high, which requires a sufficiently high bitrate for the given frame size. Therefore, in keeping with the blind nature of the problem described above, an extending algorithm SHOULD pass all tests for bitrates of 9 kb/s and above when the frame size is 20 ms, and it SHOULD pass all tests for bitrates of 12 kb/s and above when the frame size is 10 ms. Lower bitrates are included in the testvectors, but the test result on those operating points is informative only.

5. IANA Considerations

This document updates the media type registration for the audio/opus media subtype defined in [RFC7587] by adding the two optional parameters below to its list of optional parameters. Both are informative parameters by which a receiver signals its decoder's intent to enhance (and, for extendedbandwidth, to extend) the received signal, so that the far-end encoder MAY select an operating point (for example, a lower bitrate or a wideband-only mode) that the enhancement can compensate for. The decoder is not bound by this signaling and may at runtime reduce or disable enhancement, for example when CPU load is high. The parameters do not change which bitstreams a decoder accepts, and by the requirement of Section 4.1.3 any stream that relies on them MUST remain intelligible when decoded by a legacy Opus decoder.

speechenhancement:
an integer from 0 to 40 indicating the effectiveness of the receiver's speech enhancement (see Section 3). It is calibrated at 6 kb/s: if the enhanced decoder at 6 kb/s achieves the quality of an unenhanced decoder at X kb/s, the value is round(10 * (X - 6) / 6). A value of 0 (the default when the parameter is absent) indicates no speech enhancement; 10 indicates that enhancement at 6 kb/s reaches the quality of an unenhanced decoder at 12 kb/s, and 40 (the maximum) corresponds to the quality of an unenhanced decoder at 30 kb/s or better.
extendedbandwidth:
an integer giving the upper limit, in kHz, of the audio bandwidth the receiver's decoder produces after blind bandwidth extension. It applies only to content coded in the speech coding mode; content coded in the general (music) coding mode is not affected by this parameter and is reproduced at the bandwidth selected by the encoder. Typical values follow the Opus bandwidth tiers: 8 (wideband, i.e. no extension and the default when the parameter is absent), 12 (super-wideband) and 20 (fullband). For example, an extending algorithm such as BBWENet [bbwenet-paper] that extends wideband speech to fullband would signal extendedbandwidth=20.

An offerer or answerer MAY include either parameter; a value that is absent, zero (speechenhancement) or 8 (extendedbandwidth) is equivalent to the corresponding baseline capability. An implementation that does not recognize these parameters ignores them, as required by [RFC7587].

6. Security Considerations

A SILK enhancement algorithm runs after the SILK decoder and post-processes (or replaces) its output. It introduces no new bitstream elements and does not change which packets a decoder accepts or rejects, so it does not expand the packet-parser attack surface and the security considerations of [RFC6716] apply unchanged.

The following additional considerations apply to the enhancement itself:

7. References

7.1. Normative References

[RFC2119]
Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, , <https://www.rfc-editor.org/info/rfc2119>.
[RFC8174]
Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, , <https://www.rfc-editor.org/info/rfc8174>.
[RFC6716]
Valin, JM., Vos, K., and T. Terriberry, "Definition of the Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716, , <https://www.rfc-editor.org/info/rfc6716>.
[RFC7587]
Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format for the Opus Speech and Audio Codec", RFC 7587, DOI 10.17487/RFC7587, , <https://www.rfc-editor.org/info/rfc7587>.
[p.800]
ITU-T, "P.800 : Methods for subjective determination of transmission quality", , <https://www.itu.int/rec/T-REC-P.800-199608-I>.
[p.808]
ITU-T, "P.808 : Subjective evaluation of speech quality with a crowdsourcing approach", , <https://www.itu.int/rec/T-REC-P.808-202106-I/en>.

7.2. Informative References

[lace-paper]
Buethe, J., Valin, J.-M., and A. Mustafa, "LACE: A light-weight, causal Model for enhancing coded Speech through Adaptive Convolutions", 2023 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), New Paltz, NY, USA, pp. 1-5, DOI 10.1109/WASPAA58266.2023.10248150, , <https://doi.org/10.1109/WASPAA58266.2023.10248150>.
[nolace-paper]
Buethe, J., Mustafa, A., Valin, J.-M., Helwani, K., and M. Goodwin, "NoLACE: Improving Low-Complexity Speech Codec Enhancement Through Adaptive Temporal Shaping", ICASSP 2024 - 2024 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), Seoul, Republic of Korea, pp. 476-480, DOI 10.1109/ICASSP48485.2024.10448332, , <https://doi.org/10.1109/ICASSP48485.2024.10448332>.
[bbwenet-paper]
Buethe, J. and J.-M. Valin, "A lightweight and robust method for blind wideband-to-fullband extension of speech", 2025 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), Tahoe City, CA, USA, pp. 1-5, DOI 10.1109/WASPAA66052.2025.11230955, , <https://doi.org/10.1109/WASPAA66052.2025.11230955>.

Authors' Addresses

Jan (editor)
Meta Platforms Inc.
United States of America
Jean-Marc
Google
Canada