What is the actual format of this audio file?

I am collecting some data from jotform .wav widget. All the audio files are in .wav format. But when I tried to open the file using python

from pydub import AudioSegment s = AudioSegment.from_wav(audio_path) 

But it was giving error, So I tried to check whether .wav files are correctly encoded. I tried to convert existing .wav files into .wav format using FFmpeg in macOS.

This is the output I got :

$ ffmpeg -i count.wav new.wav ffmpeg version 4.2.2 Copyright (c) 2000-2019 the FFmpeg developers built with Apple clang version 11.0.0 (clang-1100.0.33.17) configuration: --prefix=/usr/local/Cellar/ffmpeg/4.2.2_2 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack libavutil 56. 31.100 / 56. 31.100 libavcodec 58. 54.100 / 58. 54.100 libavformat 58. 29.100 / 58. 29.100 libavdevice 58. 8.100 / 58. 8.100 libavfilter 7. 57.100 / 7. 57.100 libavresample 4. 0. 0 / 4. 0. 0 libswscale 5. 5.100 / 5. 5.100 libswresample 3. 5.100 / 3. 5.100 libpostproc 55. 5.100 / 55. 5.100 [mp3 @ 0x7ffa44807200] Estimating duration from bitrate, this may be inaccurate Input #0, mp3, from 'count.wav': Duration: 00:00:09.36, start: 0.000000, bitrate: 160 kb/s Stream #0:0: Audio: mp3, 48000 Hz, stereo, fltp, 160 kb/s Stream mapping: Stream #0:0 -> #0:0 (mp3 (mp3float) -> pcm_s16le (native)) Press [q] to stop, [?] for help Output #0, wav, to 'new.wav': Metadata: ISFT : Lavf58.29.100 Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s Metadata: encoder : Lavc58.54.100 pcm_s16le size= 1755kB time=00:00:09.36 bitrate=1536.1kbits/s speed= 339x video:0kB audio:1755kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.004340% 

Confusing part is if it is already .wav file then FFmpeg won't increase or decrease size while converting in same format, but after I used FFmpeg to convert this .wav to .wav it's size increased from 187KB to 1.8 MB

I thought maybe that's how FFmpeg works, so I downloaded one .wav format from internet and tried same command, but on this .wav file the size didn't increase or decrease, the size was same.

$ ffmpeg -i file_example_WAV_1MG.wav check2.wav ffmpeg version 4.2.2 Copyright (c) 2000-2019 the FFmpeg developers built with Apple clang version 11.0.0 (clang-1100.0.33.17) configuration: --prefix=/usr/local/Cellar/ffmpeg/4.2.2_2 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack libavutil 56. 31.100 / 56. 31.100 libavcodec 58. 54.100 / 58. 54.100 libavformat 58. 29.100 / 58. 29.100 libavdevice 58. 8.100 / 58. 8.100 libavfilter 7. 57.100 / 7. 57.100 libavresample 4. 0. 0 / 4. 0. 0 libswscale 5. 5.100 / 5. 5.100 libswresample 3. 5.100 / 3. 5.100 libpostproc 55. 5.100 / 55. 5.100 [wav @ 0x7f9726007400] Discarding ID3 tags because more suitable tags were found. Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, wav, from 'file_example_WAV_1MG.wav': Metadata: title : Impact Moderato album : YouTube Audio Library artist : Kevin MacLeod genre : Cinematic Duration: 00:00:33.53, bitrate: 256 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, stereo, s16, 256 kb/s Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native)) Press [q] to stop, [?] for help Output #0, wav, to 'check2.wav': Metadata: INAM : Impact Moderato IPRD : YouTube Audio Library IART : Kevin MacLeod IGNR : Cinematic ISFT : Lavf58.29.100 Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, stereo, s16, 256 kb/s Metadata: encoder : Lavc58.54.100 pcm_s16le size= 1048kB time=00:00:33.52 bitrate= 256.0kbits/s speed=2.83e+03x video:0kB audio:1048kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.016031% 

I am confused about which format jotform is using to store the audio files because the extension says .wav but it's not opening in pyaudio and also FFmpeg is converting in .wav with bigger size, which indicates that the raw format is not actually .wav it's the compressed version?

I am uploading the .wav file here