[FFmpeg-user] need help to convert wav to mp3
Ashtiani
ashtiani.alireza at gmail.com
Wed Sep 22 16:08:28 EEST 2021
lame :
Unsupported data format: 0x0701
Can't init infile 'input.wav'
On Wed, Sep 22, 2021 at 4:06 PM Reindl Harald <h.reindl at thelounge.net>
wrote:
>
>
> Am 22.09.21 um 14:09 schrieb Ashtiani:
> > It is created by a type of recorder but I need to play it in My Web
> > Application .
> > browser just support webm, mp3 and ogg format :(
>
> i wouldn't use ffmpeg for mp3 conversion to begin with
>
> [harry at srv-rhsoft:~]$ lame --help
> LAME 64bits version 3.100 (http://lame.sf.net)
>
> usage: lame [options] <infile> [outfile]
>
> <infile> and/or <outfile> can be "-", which means stdin/stdout.
>
> RECOMMENDED:
> lame -V2 input.wav output.mp3
>
> OPTIONS:
> -b bitrate set the bitrate, default 128 kbps
> -h higher quality, but a little slower.
> -f fast mode (lower quality)
> -V n quality setting for VBR. default n=4
> 0=high quality,bigger files. 9.999=smaller files
> --preset type type must be "medium", "standard", "extreme",
> "insane",
> or a value for an average desired bitrate and
> depending
> on the value specified, appropriate quality
> settings will
> be used.
> "--preset help" gives more info on these
>
> --help id3 ID3 tagging related options
>
> --longhelp full list of options
>
> --license print License information
>
> > On Wed, Sep 22, 2021 at 3:34 PM Paul B Mahol <onemda at gmail.com> wrote:
> >
> >> On Wed, Sep 22, 2021 at 1:57 PM Ashtiani <ashtiani.alireza at gmail.com>
> >> wrote:
> >>
> >>> yes the sound is correct and play with MPC-HC <https://mpc-hc.org/>
> >>> player ;
> >>> Format : Wave
> >>> File size : 356 KiB
> >>> Duration : 1 min 50 s
> >>> Overall bit rate : 26.4 kb/s
> >>>
> >>> Audio
> >>> Format : 701
> >>> Codec ID : 701
> >>> Duration : 1 min 50 s
> >>> Bit rate : 26.4 kb/s
> >>> Channel(s) : 2 channels
> >>> Sampling rate : 8 000 Hz
> >>> Stream size : 356 KiB (100%)
> >>>
> >>
> >>
> >> Hm, what created such wav file? Because that adpcm variant should have
> >> 0x0017 twocc and not 0x1700 one.
> >>
> >>
> >>>
> >>> On Wed, Sep 22, 2021 at 3:17 PM Paul B Mahol <onemda at gmail.com> wrote:
> >>>
> >>>> On Wed, Sep 22, 2021 at 1:22 PM Ashtiani <ashtiani.alireza at gmail.com>
> >>>> wrote:
> >>>>
> >>>>> thanks paul :
> >>>>>
> >>>>> *" ffplay -avcodec adpcm_ima_oki input.wav" : *
> >>>>> Failed to set value 'adpcm_ima_oki' for option 'avcodec': Option not
> >>>> found
> >>>>>
> >>>>
> >>>> Sorry, i meant acodec. not avcodec.
> >>>>
> >>>>
> >>>>
> >>>>>
> >>>>>
> >>>>> *"ffmpeg -c:a adpcm_ima_oki -i input.wav out.mp3"*
> >>>>> ffmpeg version 4.4 Copyright (c) 2000-2021 the FFmpeg developers
> >>>>> built with gcc 8 (GCC)
> >>>>> configuration: --arch=x86_64 --bindir=/usr/bin
> >>>>> --datadir=/usr/share/ffmpeg --disable-debug --disable-static
> >>>>> --disable-stripping --enable-avcodec --enable-avdevice
> >>> --enable-avfilter
> >>>>> --enable-avformat --enable-avresample --enable-alsa --enable-bzlib
> >>>>> --enable-chromaprint --enable-decklink --enable-frei0r
> >> --enable-gcrypt
> >>>>> --enable-gmp --enable-gpl --enable-gray --enable-iconv
> >> --enable-ladspa
> >>>>> --enable-libass --enable-libaom --enable-libbluray --enable-libbs2b
> >>>>> --enable-libcaca --enable-libcdio --enable-libcodec2
> >> --enable-libdc1394
> >>>>> --enable-libdav1d --enable-libdavs2 --enable-libdrm
> >> --enable-libfdk-aac
> >>>>> --enable-libfontconfig --enable-libfreetype --enable-libfribidi
> >>>>> --enable-libgme --enable-libgsm --enable-libiec61883 --enable-libilbc
> >>>>> --enable-libjack --enable-libkvazaar --enable-libmodplug
> >>>>> --enable-libmp3lame --enable-libndi_newtek --enable-libopencore-amrnb
> >>>>> --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg
> >>>>> --enable-libopenmpt --enable-libopus --enable-libpulse
> >>>> --enable-librabbitmq
> >>>>> --enable-librsvg --enable-librtmp --enable-librubberband
> >>>>> --enable-libsmbclient --enable-libsnappy --enable-libsoxr
> >>>> --enable-libspeex
> >>>>> --enable-libssh --enable-libtesseract --enable-libtheora
> >>>>> --enable-libtwolame --enable-libv4l2 --enable-libvidstab
> >>>>> --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
> >>>> --enable-libwebp
> >>>>> --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxcb
> >>>>> --enable-libxcb-shape --enable-libxcb-shm --enable-libxcb-xfixes
> >>>>> --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzvbi
> >>>>> --enable-lzma --enable-nonfree --enable-openal --enable-opencl
> >>>>> --enable-opengl --enable-openssl --enable-postproc --enable-sdl2
> >>>>> --enable-shared --enable-swresample --enable-swscale --enable-vaapi
> >>>>> --enable-version3 --enable-vdpau --enable-xlib --enable-zlib
> >>>>> --incdir=/usr/include/ffmpeg --libdir=/usr/lib64
> >>> --mandir=/usr/share/man
> >>>>> --optflags='-O2 -g -pipe -Wall -Werror=format-security
> >>>>> -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fexceptions
> >>>>> -fstack-protector-strong -grecord-gcc-switches
> >>>>> -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1
> >>>>> -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic
> >>>>> -fasynchronous-unwind-tables -fstack-clash-protection
> >> -fcf-protection'
> >>>>> --prefix=/usr --shlibdir=/usr/lib64 --enable-libsrt --enable-libzmq
> >>>>> --enable-v4l2-m2m --enable-vapoursynth --enable-vulkan --enable-cuda
> >>>>> --enable-cuvid --enable-ffnvcodec --enable-libmfx --enable-libnpp
> >>>>> --enable-libsvtav1 --enable-libsvthevc --enable-libsvtvp9
> >>>> --enable-libvmaf
> >>>>> --enable-nvdec --enable-nvenc --extra-cflags=-I/usr/include/cuda
> >>>>> --cpu=x86_64
> >>>>> libavutil 56. 70.100 / 56. 70.100
> >>>>> libavcodec 58.134.100 / 58.134.100
> >>>>> libavformat 58. 76.100 / 58. 76.100
> >>>>> libavdevice 58. 13.100 / 58. 13.100
> >>>>> libavfilter 7.110.100 / 7.110.100
> >>>>> libavresample 4. 0. 0 / 4. 0. 0
> >>>>> libswscale 5. 9.100 / 5. 9.100
> >>>>> libswresample 3. 9.100 / 3. 9.100
> >>>>> libpostproc 55. 9.100 / 55. 9.100
> >>>>> 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: adpcm_ima_oki ([1][7][0][0] / 0x0701), 8000
> Hz,
> >>>>> stereo, s16, 64 kb/s
> >>>>> File 'out.mp3' already exists. Overwrite? [y/N] y
> >>>>> Stream mapping:
> >>>>> Stream #0:0 -> #0:0 (adpcm_ima_oki (native) -> mp3 (libmp3lame))
> >>>>> Press [q] to stop, [?] for help
> >>>>> Output #0, mp3, to 'out.mp3':
> >>>>> Metadata:
> >>>>> TSSE : Lavf58.76.100
> >>>>> Stream #0:0: Audio: mp3, 8000 Hz, stereo, s16p
> >>>>> Metadata:
> >>>>> encoder : Lavc58.134.100 libmp3lame
> >>>>> size= 134kB time=00:00:45.58 bitrate= 24.2kbits/s speed= 134x
> >>>>> video:0kB audio:134kB subtitle:0kB other streams:0kB global
> >> headers:0kB
> >>>>> muxing overhead: 0.189990%
> >>>>>
> >>>>>
> >>>>>
> >>>> Is sound correct?
> >>>>
> >>>>
> >>>>>
> >>>>> On Wed, Sep 22, 2021 at 2:32 PM Paul B Mahol <onemda at gmail.com>
> >> wrote:
> >>>>>
> >>>>>> 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
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