How to convert Opus audio file to mp3 with maximum quality use FFmpeg? Linux. Source audio file is `Opus, stereo 48 kHz, 138 kbps, Bits per sample:32` I used ffmpeg -i audio.opus -c:a libmp3lame -q:a 0 newfile.mp3 which gave me VBR audio with an average bitrate about 273 kb/s.

I can embed an mp4 file from an ogg file with embedded subtitles from an srt file alright, thanks to stack*** forums and other internet resources, in two steps: First create a black png file for background against white subtitle. Then join the audio with 1 pic to create an mp4 file. ffmpeg -loop 1 -i 1.png -i input.ogg -shortest output.mp4 On the second round, use that mp4 file and hard embed the subtitles into the file. ffmpeg -i output.mp4 -vf subtitles=input.srt out.mp4 My next objective would have been to combine the two steps into one. There are some resources that are available on the internet, notably, Linux Subtitles ffmpeg , from which I have the code: ffmpeg -i input.ogg -loop 1 -i 1.png -c:v libx264 -r 24000/1001 -vf "srt='input.srt'" -pix_fmt yuv240p -c:a copy -map 1:v -map 0:a -shortest output.mp4 There are errors: ffmpeg version 4.3.5-0+deb11u1 Copyright (c) 2000-2022 the FFmpeg developers built with gcc 10 (Debian 10.2.1-6) configuration: --prefix=/usr --extra-version=0+deb11u1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-pocketsphinx --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared libavutil 56. 51.100 / 56. 51.100 libavcodec 58. 91.100 / 58. 91.100 libavformat 58. 45.100 / 58. 45.100 libavdevice 58. 10.100 / 58. 10.100 libavfilter 7. 85.100 / 7. 85.100 libavresample 4. 0. 0 / 4. 0. 0 libswscale 5. 7.100 / 5. 7.100 libswresample 3. 7.100 / 3. 7.100 libpostproc 55. 7.100 / 55. 7.100 Input #0, ogg, from 'input.ogg': Duration: 00:09:56.73, start: 0.000000, bitrate: 53 kb/s Stream #0:0: Audio: vorbis, 44100 Hz, mono, fltp, 140 kb/s Input #1, png_pipe, from '1.png': Duration: N/A, bitrate: N/A Stream #1:0: Video: png, rgba(pc), 639x288, 25 fps, 25 tbr, 25 tbn, 25 tbc File 'output.mp4' already exists. Overwrite? [y/N] y Stream mapping: Stream #1:0 -> #0:0 (png (native) -> h264 (libx264)) Stream #0:0 -> #0:1 (copy) Press [q] to stop, [?] for help [AVFilterGraph @ 0x5623a8c54480] No such filter: 'srt' Error reinitializing filters! Failed to inject frame into filter network: Invalid argument Error while processing the decoded data for stream #1:0 Conversion failed! I am sure that an expert will easily be able to solve the problem faced, by suitably correcting the above line of code. But my final objective is still beyond: Since my input content file is only an audio file, I want to have the subtitles spread out across the entire screen, not restricted only to the bottom of the screen. I have seen such videos on YouTube, but the resources and code presently evade me.

Question ======== I have 2 files: - audio.opus (extension also possible: .opus.ogg), - cover.jpg. **How can I use ffmpeg to put cover.jpg as an embedded "cover" image into audio.opus?** Constraints ----------- - I don't want to re-encode the audio (so unfortunately, opusenc --picture seems out of scope), - I don't want to re-encode the image (jpeg), although that's not much of a problem, - I don't want to transform the image into a video (!), - I'd really prefer doing it with ffmpeg and not another tool, - I don't want to hear about mp3. Expected result --------------- Here is how it should look like when it's properly done:

$ opusinfo audio_with_cover.opus
Processing file "audio_with_cover.opus"...

New logical stream (#1, serial: 39a74bbb): type opus
Encoded with libopus 1.3.1
User comments section follows...
	ENCODER=opusenc from opus-tools 0.1.10
	METADATA_BLOCK_PICTURE=3|image/jpeg||1280x720x24|
	ENCODER_OPTIONS=--bitrate 56
Opus stream 1:
	Pre-skip: 312
    [...]

Or seen from ffprobe:

$ ffprobe audio_with_cover.opus 
[ogg @ 0x55675650ca80] 761 bytes of comment header remain
Input #0, ogg, from 'audio_with_cover.opus':
  Duration: 00:02:25.64, start: 0.000000, bitrate: 71 kb/s
  Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
    Metadata:
      ENCODER         : opusenc from opus-tools 0.1.10
      ENCODER_OPTIONS : --bitrate 56
  Stream #0:1: Video: mjpeg (Baseline), yuvj420p(pc, bt470bg/unknown/unknown), 1280x720, 90k tbr, 90k tbn (attached pic)
    Metadata:
      comment         : Cover (front)

Not possible? ------------- If you know it is not possible, please let me know, I will open a feature request at ffmpeg.

My girlfriend accidently bought an audio book on the internet and she saved it on my hard disk in windows media audio format (.wma) and now she asks me how to listen to it. I'm a bit puzzled this format still survived. I'm on ArchLinux, kernel 4.2.1-1-ARCH. My first idea was simply trying to read it with VLC as it usually is a general problem solver for audio files and comes with a couple of codecs. But it didn't work. Second idea was converting, I usually use Audacity for quick format swaps, but after several attempts of exporting to ogg/vorbis or ogg/opus I noticed Audacity already failed to read the proprietary file format. I looked up the codecs section on arch wiki but it does not contain any hints for WMA. How to proceed? How to convert audio/x-ms-wma to audio/ogg, audio/vorbis, or audio/opus format?

I am searching a command-line tagging tool that supports ogg- and mp3-tagging and allows to add a picture to the tags. Currently I use lltag but that does not support pictures. dagger is the tool I use before but it is lacking picture support as well.

I am using youtube-dl to download ogg audio files with their respective thumbnail, but I am not sure how to embed the thumbnail into the audio file.

I have a directory with a bunch of CD quality (16bit 44100Hz) wave-files. How can I batch decode those into different formats (lets say FLAC, OGG and MP3) using ffmpeg? ---------- **Update:** here are the commands one by one as suggested by @StephenHarris ffmpeg -i input.wav output.ogg ffmpeg -i input.wav output.mp3 ffmpeg -i input.wav output.flac

Starting recently, about 2-3 weeks ago, I believe after some update, Clementine stopped playing OGG files. All other formats that I have play without any issues. At the same time VLC can play these same OGG files. When it fails, Clementine pops an error window with the following message only: **Internal data stream error.** At the same time, this error message gets printed to the terminal: (clementine:647736): GStreamer-CRITICAL **: 11:27:26.111: gst_pad_get_range_unchecked: assertion '!*buffer || res_buf == *buffer' failed (clementine:647736): GStreamer-CRITICAL **: 11:27:26.111: gst_mini_object_unref: assertion 'GST_MINI_OBJECT_REFCOUNT_VALUE (mini_object) > 0' failed 11:27:26.186 WARN unknown QTimeLine::start: already running 11:27:26.691 WARN unknown QTimeLine::start: already running 11:27:26.691 WARN unknown QTimeLine::start: already running 11:27:26.695 WARN unknown QTimeLine::start: already running Version info: - Fedora 33 5.9.11-200.fc33.x86_64 - clementine.x86_64: 1.4.0-3.rc1.20200617gitedb8c3b.fc33.1 - gstreamer1.x86_64: 1.18.2-1.fc33 - gstreamer1-libav.x86_64: 1.18.1-1.fc33 - gstreamer1-plugin-openh264.x86_64: 1.16.2-2.fc33 - gstreamer1-plugins-bad-free.x86_64: 1.18.2-1.fc33 - gstreamer1-plugins-bad-free-extras.x86_64: 1.18.2-1.fc33 - gstreamer1-plugins-base.x86_64: 1.18.2-1.fc33 - gstreamer1-plugins-good.x86_64: 1.18.2-1.fc33 - gstreamer1-plugins-good-gtk.x86_64: 1.18.2-1.fc33 - gstreamer1-plugins-good-qt.x86_64: 1.18.2-1.fc33 - gstreamer1-plugins-ugly.x86_64: 1.18.1-1.fc33 - gstreamer1-plugins-ugly-free.x86_64: 1.18.2-1.fc33 Any idea on what caused it and how to fix it?

my ~/.bashrc contains the following function webcamrecord () { vlc v4l2:// :v4l-vdev="/dev/video0" :v4l-adev="/dev/audio2" :v4l-norm=3 :v4l-frequency=-1 :v4l-caching=300 :v4l-chroma="" :v4l-fps=-1.000000 :v4l-samplerate=44100 :v4l-channel=0 :v4l-tuner=-1 :v4l-audio=-1 :v4l-stereo :v4l-width=640 :v4l-height=480 :v4l-brightness=-1 :v4l-colour=-1 :v4l-hue=-1 :v4l-contrast=-1 :no-v4l-mjpeg :v4l-decimation=1 :v4l-quality=100 --sout="#transcode{vfilter=adjust{gamma=1.5},vcodec=theo,vb=2000,scale=0.67,acodec=vorb,ab=128,channels=2}:standard{access=file,mux=ogg,dst="output_file.ogg"}" } The command works beautifully to record directly from my webcam, with one or two flaws. First, the video produced moves too quickly, between twice and three times the normal rate. Also, there is no audio. This is less of an issue, but I would still like to fix this. Can someone help me debug this command, so as to fix these issues?

I've got some OPUS files I've created with FFMPEG on Linux (from MP3 originals I've deleted to save storage space). I'd like to make them playable on a Mac and, if possible, on an iPhone via iTunes. As far as I know Apple hardware and software (not the web browsers though) support OPUS but don't support its native OGG container format and need it to be contained in Core Audio files instead. How do I perform this conversion on Linux and/or on MacOS? I know I can just convert back to MP3 or AAC and know how to do this but I'm curious about creating Core Audio files without re-encoding the actual audio data.

I am using Vorbis/ogg to record audio (using arecord and then pipping to oggenc). I have downloaded oggvideotools, which gave me oggCat, which will let me join two previously recorded audio files into one file. But, my problem is this: The device I am working on is known to have the "plug pulled", but it is recording while the power is on (swapping out batteries etc.). Once the device is powered back on, I need to continue to record to the previous file (automated recording). Right now, I can use arecord to record two separate files, and then once the recording is done, I can oggCat the two, but that doesn't get me what I want 100%, I need to be able to continue recording where the first file got cut off. I need python and busybox solutions. Unfortunately, other solutions outside Python and Busybox are not available to me - furthermore, downloading additional packages/programs is also not available, I have oggenc, arecord, and oggvideotools. Any advise on where to look? Can I somehow pipe arecord data from a temp file to oggCat? Like: oggCat NewFile.ogg File1.ogg (arecord file2.ogg ...) Where (arecord file2.ogg ...) is the file currently being recorded and simultaneously being appended to Newfile.ogg? PS. The audio files will have the exact same parameters - i.e. same channels, rates, etc.

In order to run ogg123 (for getting wav from ogg vorbis) I need to get (not found) or compile static build. I tried this on Amazon Linux (the same version as current AWS Lambda): ./configure --disable-shared --enable-static make LDFLAGS=-lm SHARED=0 CC='gcc -static' Produced ogg123 filesize is 288K but when I copied that file to another Amazon Linux and tried to run I get error while loading shared libraries: libvorbisfile.so.3: cannot open shared object file: No such file or directory

I am running Amarok 2.5.0 on Fedora 16 and it is displaying the wrong time (zero or a very large number) instead of the actual duration of the song, for all ogg audio files. Is there a way to fix this issue?

When trying to convert either an mp3 or flac file to ogg, the output ogg file is actually a flac file with a large file size. For instance: running for file in *.mp3; do ffmpeg -i "${file}" "${file/%mp3/ogg}"; done and then checking the file with mediainfo output.ogg gives: General Complete name : 06 - Magma.ogg Format : Ogg Format/Info : Free Lossless Audio Codec File size : 47.0 MiB Duration : 6mn 42s Overall bit rate mode : Variable Overall bit rate : 980 Kbps Audio ID : 1238237382 (0x49CE00C6) Format : FLAC Format/Info : Free Lossless Audio Codec Duration : 6mn 42s Bit rate mode : Variable Channel(s) : 2 channels Channel positions : Front: L R Sampling rate : 44.1 KHz Bit depth : 16 bits Writing library : Lavf57.37.100 The input mp3 in my case was 6 megabytes, but the ogg file is for some reason 47 megabytes. Why is this happening, and is there a why to convert files to ogg without this happening?

I am running an online radio based on Icecast. I wrote the source client myself, so that I can be flexible in schedule of the tracks and so on. In order to play tracks on the radio, inside my python code I run 2 subprocesses: ffmpeg to decode file from disk into WAV and then redirect it's output is encoded by oggenc. Something like this, for each track: ffmpeg -i foo_bar.mp3 -acodec pcm_s16le -ac 2 -f wav pipe:1 | oggenc - -b 128 --managed -o - It works fine, until there is a new track to play. Sometimes players start to behave strange, for example mplayer simply stops and exits, vlc can change speed and start producing creepy sounds and so on. The simplest solution would be keep oggenc running forever and somehow substitute the waveform it reads from standart input, but I have no idea how to do it in a not so tricky manner. I would like to avoid complicated things here, because it impacts the stability of the source client. What do you think? Thanks

I have a lot number audio files in MP3 the proprietary format, I want to convert them to 'opus' the free and high quality format, with keep metadata also. My selection command-line programs are SoX (Sound eXchange) for convert MP3 files to 'AIFF' format in order to keep quality and metadata* and 'opusenc ' the official encoder for convert 'AIFF' files to 'opus' format. Most important part of the work is use of 'mediainfo' program for exporting metadata tags of MP3 files and giving them to 'opusenc' program for importing metadata tags to encoding opus files, also giving the original bit-rate value of MP3 files to 'opusenc' for encoding large size AIFF files in order to achieve the opus files with same bit-rate of original MP3 files, so I need a script for automation listed below operations: 1. Use sox for convert MP3 files in a directory with many sub-directory to the files with AIFF format with same name. 2. Use 'mediainfo' for exporting metadata tags (Album, Performer, Track name, Recorded date, Genre) and bit-rate value of original MP3 files. 3. Giving the previous step information to 'opusenc' (artist, title, album, date YYYY-MM-DD, genre) and run it. 4. Delete MP3 and AIFF files after encode successfully done. I want mentioned operations run consecutive on directories one by one. My goal of this work is achieve to more freedom. Thank you very much for help and time that you devote to this work. *: I choose AIFF format because it supports ID3v2 tags but when for test I convert a MP3 file contains metadata to 'AIFF' format with 'SoX' program and convert it to 'opus' with 'opusenc', result was the output opus file lacks metadata (don't known what is the problem exactly) also there was need to specify bit-rate value of MP3 file, so use of 'mediainfo' is necessary.

I am programming a Rasberry to Pi to become a radio player. It is meant to be used to play the [local college radio](http://radiocampus.se/) in the student union hangout. INPUT: - Power is turn on. OUTPUT: - Audio is played through the 3,5 mm jack. PROCESS: 1. Device boot Raspdebian 2. ... 3. Start playing an ogg stream (/radiocampus_local.m3u) How can I do this?

How can I add an audio file (OGG vorbis) into a shell script making the script as small as possible and at the same time being able to execute it on as most systems as possible? I want the audio file be played. I don’t want it to be uuencoded, because sharutils are not installed on many systems. base64 is nice, but makes the file/script bigger, which I don’t want. Should I use mplayer or the play command from the sox package? Is it more common on Linux/BSD systems? Or is there another media player mostly installed? Should I convert the audio file to another format/codec (if that doesn’t make the file much bigger) to have more possible players to play the file back or is OGG vorbis a good choice?

How can I get this "something" out of the .ogg file? How can I clear comments/infos like this from the file? $ strings foo.ogg |grep -i SOMETHING ARTIST=SOMETHING $ OS: Scientific Linux 6.3

Is there any utility to search MP3/Ogg files by tags (or other characteristics) from the command line? (e.g. finding all audio files longer than 10 minutes with genre set to 'Rock'). I found this , which suggests using mp3info, but it's only for MP3 files and the proposed solution isn't the most elegant thing. Any other suggestions?