A mode tuned for 44.1 kHz (pretending it's actually 48 kHz) could be added, but it would be a lot of work without benefit. You needn't be concerned about resampling. There are many transparent (without audible artefacts) resamplers which will do a fine job converting 44.1 kHz input to 48 kHz for encoding in Opus.

Put another way, if you have an application where resampling causes quality loss, you should use a lossless compressor like FLAC instead of Opus.

I'm really concerned about the resampling...So 2 lossy operations....for which I'm still sticking with Vorbis.

The losses introduced by resampling are below audible thresholds. The whole workings of a lossy codec are a stack of lossy operations! Why focus on the one that has about the least loss of all of them? You're worrying about it in Opus, when in reality the quality loss due to it is far below the quality loss you get from Vorbis('s stack of lossy operations) as compared to Opus's.

Opus gives better quality at 44.1kHz than Vorbis does. And Opus does support 44.1kHz, unlike the bug title implies. Using a resampler doesn't mean it's not supported, on the contrary.

From a compatibility perspective, a 44.1 kHz Opus would be like a new codec and it would be incompatible with existing deployment. Breaking compatibility because of a thought experiment (assuming that resampling degrades quality despite no evidence existing) is not something we're going to do.

Existing OPUS decoders can't follow a 44100 instruction? If not, adding 44.1 kHz output wouldn't break compatibility.

They could not play it, which is exactly what breaking compatibility means. Worse, the quality would almost certainly be worse as a result of losing the benefit of the careful tuning of the codec. Not better. With a good resampler the conversion to 48kHz is essentially lossless, far far more so than the codec itself (technically a matched 44.1>48>44.1 conversion set could be made actually lossless if anyone cared to ever create such a thing). Opus doesn't do 44.1k because there is literally no point to do so, and supporting different rates makes life harder for decoders (especially ones that have to deal with streams from multiple sources simultaneously).

Existing OPUS decoders can't follow a 44100 instruction? If not, adding 44.1 kHz output wouldn't break compatibility.

They could not play it, which is exactly what breaking compatibility means.

What does --raw-rate do?

  parec -v --raw -d alsa_output.pci-0000_00_1b.0.analog-stereo.monitor | opusenc --raw-rate 44100 - $HOME/localscripts/output.opus
  mkvmerge -w --enable-durations -o $HOME/localscripts/output.webm $HOME/localscripts/output.opus

Firefox 76, Nightly 78, Chromium 81, mpv, ffplay each play the file.

mkvinfo -t output.webm

+ EBML head
|+ EBML version: 1
|+ EBML read version: 1
|+ Maximum EBML ID length: 4
|+ Maximum EBML size length: 8
|+ Document type: webm
|+ Document type version: 4
|+ Document type read version: 1
+ Segment: size 618781
|+ Seek head (subentries will be skipped)
|+ EBML void: size 4029
|+ Segment information
| + Timestamp scale: 22674
| + Multiplexing application: libebml v1.3.10 + libmatroska v1.5.2
| + Writing application: mkvmerge v46.0.0 ('No Deeper Escape') 32-bit
| + Duration: 00:01:00.784753962
| + Date: Wed May 27 03:32:52 2020 UTC
|+ Tracks
| + Track
|  + Track number: 1 (track ID for mkvmerge & mkvextract: 0)
|  + Track UID: 7354171215355932620
|  + Track type: audio
|  + Codec ID: A_OPUS
|  + Seek pre-roll: 00:00:00.080000000
|  + Codec's private data: size 19
|  + Language: und
|  + Codec-inherent delay: 00:00:00.006500000
|  + Audio track
|   + Sampling frequency: 44100
|   + Channels: 2

$ mkvmerge -J output.opus

{
  "attachments": [],
  "chapters": [],
  "container": {
    "properties": {
      "container_type": 23,
      "is_providing_timestamps": true
    },
    "recognized": true,
    "supported": true,
    "type": "Ogg/OGM"
  },
  "errors": [],
  "file_name": "output.opus",
  "global_tags": [],
  "identification_format_version": 12,
  "track_tags": [],
  "tracks": [
    {
      "codec": "Opus",
      "id": 0,
      "properties": {
        "audio_channels": 2,
        "audio_sampling_frequency": 44100,
        "number": 833194209,
        "tag_encoder": "opusenc from opus-tools 0.1.10"
      },
      "type": "audio"
    }
  ],
  "warnings": []
}

$ mediainfo output.opus

General
Complete name                            : output.opus
Format                                   : Ogg
File size                                : 560 KiB
Duration                                 : 1 min 0 s
Overall bit rate                         : 75.4 kb/s
Writing application                      : opusenc from opus-tools 0.1.10

Audio
ID                                       : 833194209 (0x31A988E1)
Format                                   : Opus
Duration                                 : 1 min 0 s
Channel(s)                               : 2 channels
Channel positions                        : Front: L R
Sampling rate                            : 44.1 kHz
Compression mode                         : Lossy
Writing library                          : libopus 1.1.2

ffplay output.opus 

[ogg @ 0x9d200600] 693 bytes of comment header remain    0B f=0/0   
Input #0, ogg, from 'output.opus':
  Duration: 00:01:00.78, start: 0.000000, bitrate: 75 kb/s
    Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
    Metadata:
      ENCODER         : opusenc from opus-tools 0.1.10

ffprobe output.opus

[ogg @ 0x156b7a0] 693 bytes of comment header remain
Input #0, ogg, from 'output.opus':
  Duration: 00:01:00.78, start: 0.000000, bitrate: 75 kb/s
    Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
    Metadata:
      ENCODER         : opusenc from opus-tools 0.1.10

$ mpv output.opus

Playing: output.opus
 (+) Audio --aid=1 (opus 2ch 48000Hz)
AO: [pulse] 48000Hz stereo 2ch float
A: 00:01:00 / 00:01:00 (98%)

Не, ну ёпта, пацаны. Зачем вам 48000 вообще изначально в этом кодеке? Вы делаете его в первую очередь для передачи речи? Тогда так и ответьте автору топика, чтобы музыку свою кодировал во что-то другое, т.к. очередной мега-супер-кодек, который многие эксперты опробовав с помощью своих невероятно чувствительных ушей и указали на его неоспоримое превосходство над другими, попросту не предназначен для меломанов.
Бро, конвертируй всё в AAC через iTunes и больше не парься ни о чём, всё равно поделия в виде Vorbis и Opus не поддерживаются таким большим перечнем железа, как MP3 и AAC, что требует дополнительных ресурсов для их декодирования.
Удачи.

According to Google's translation service, the Russian comment recommends MPEG codecs as better for music because they are more broadly supported and don't require resampling to/from 44.1 kHz.

If you'd like to discuss the objective, testable and measurable merits of resampling, you want HydrogenAudio.

If you'd like to start an opinionated flame war, there's Reddit, AVSForum, etc.

If there's a bug in the code or an entirely overlooked feature you've noticed, then Github is the right place.