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@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 <one...@gmail.com> wrote:

On Wed, Sep 22, 2021 at 1:57 PM Ashtiani <ashtiani.alir...@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 <one...@gmail.com> wrote:

On Wed, Sep 22, 2021 at 1:22 PM Ashtiani <ashtiani.alir...@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 <one...@gmail.com>
wrote:

On Wed, Sep 22, 2021 at 11:06 AM Arif Driessen <arif...@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.alir...@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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