[FFmpeg-user] need help to convert wav to mp3

[Paul B Mahol]

On Wed, Sep 22, 2021 at 11:06 AM Arif Driessen <arifd86 at gmail.com> wrote:

> Looking at your output it looks like it can't figure out what your
> input.wav is. Does your wav file work in other applications? Have you tried
> increasing the value for 'analyzeduration'?
> If you know the details you could manually specify them, something like:
> ffmpeg -acodec pcm_s16le -i input.wav out.mp3 (it's likely going to be
> pcm_s16le or pcm_s24le or pcm_s32le)
>
> Perhaps you don't have these codec installed/enabled. You will want
> something with PCM in it. Run: ffmpeg -codecs and look for anything with
> PCM in it. On Linux, and possibly Mac, you could run: ffmpeg -codecs | grep
> -e '\sPCM'
> to filter for lines containing PCM for you.
>
> On Wed, Sep 22, 2021 at 7:41 AM Ashtiani <ashtiani.alireza at gmail.com>
> wrote:
>
> > Try to Convert an Audio wave format file to mp3 with ffmpeg with below
> > command :
> >
> > ffmpeg  -i input.wav output.mp3
> >
> > but ffmpeg say 'could not find codec':
> >
> > ffmpeg version 2021-09-16-git-8f92a1862a-full_build-www.gyan.dev
> > Copyright (c) 2000-2021 the FFmpeg developers
> >   built with gcc 10.3.0 (Rev5, Built by MSYS2 project)
> >   configuration: --enable-gpl --enable-version3 --enable-static
> > --disable-w32threads --disable-autodetect --enable-fontconfig
> > --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp
> > --enable-lzma --enable-libsnappy --enable-zlib --enable-librist
> > --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth
> > --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libdav1d
> > --enable-libzvbi --enable-librav1e --enable-libsvtav1 --enable-libwebp
> > --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom
> > --enable-libopenjpeg --enable-libvpx --enable-libass --enable-frei0r
> > --enable-libfreetype --enable-libfribidi --enable-libvidstab
> > --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm
> > --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc
> > --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libglslang
> > --enable-vulkan --enable-opencl --enable-libcdio --enable-libgme
> > --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb
> > --enable-libmp3lame --enable-libshine --enable-libtheora
> > --enable-libtwolame --enable-libvo-amrwbenc --enable-libilbc
> > --enable-libgsm --enable-libopencore-amrnb --enable-libopus
> > --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b
> > --enable-libflite --enable-libmysofa --enable-librubberband
> > --enable-libsoxr --enable-chromaprint
> >   libavutil      57.  5.100 / 57.  5.100
> >   libavcodec     59.  7.103 / 59.  7.103
> >   libavformat    59.  5.100 / 59.  5.100
> >   libavdevice    59.  0.101 / 59.  0.101
> >   libavfilter     8.  9.100 /  8.  9.100
> >   libswscale      6.  1.100 /  6.  1.100
> >   libswresample   4.  0.100 /  4.  0.100
> >   libpostproc    56.  0.100 / 56.  0.100
> > [wav @ 000001a24340ea40] Could not find codec parameters for stream 0
> > (Audio: none ([1][7][0][0] / 0x0701), 8000 Hz, 2 channels, 26 kb/s):
> > unknown codec
> > Consider increasing the value for the 'analyzeduration' (0) and
> > 'probesize' (5000000) options
> > Guessed Channel Layout for Input Stream #0.0 : stereo
> > Input #0, wav, from 'input.wav':
> >   Duration: 00:01:50.54, bitrate: 26 kb/s
> >   Stream #0:0: Audio: none ([1][7][0][0] / 0x0701), 8000 Hz, stereo, 26
> > kb/s
>

Maybe this is ADPCM variant .

does:

ffplay -avcodec adpcm_ima_oki input.wav

plays anything

> Stream mapping:
> >   Stream #0:0 -> #0:0 (? (?) -> mp3 (libmp3lame))
> > Decoder (codec none) not found for input stream #0:0
> >
> > tnx
> >
> > aras
> > _______________________________________________
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> > ffmpeg-user at ffmpeg.org
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> >
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> >
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