I recorded an audio using audacity, then increased its volume 10 times using ffmpeg. When I am sending this audio as attachment to my friends to test, then people with older i-phones are able to hear it fine (volume is still low though), but people on i-phones 14 and higher are not hearing my voice in the mp3 at all.
Actually, on Apple 14 and higher, my voice is there in audio, but its so low in volume that its almost inaudible, impossible to hear. I can hear it if I strain really really a lot. Other things on the phone are playing fine, audio settings are showing full.
If I add a background music to the audio using ffmpeg, then people on i-phones 14 and higher are only hearing the music, not my voice.
What is happening here?
Do I need to adjust some frequency etc settings in the recorded audio when using ffmpeg to increase the volume?
Here is the ffmpeg command I am using to increase volume in my recorded audio. Output mp3 file from below command is uploaded here:
https://tbrjar.com/audio_test/z1.mp3
ffmpeg command and output:
ffmpeg -ss 00:00:00 -i input.mp3 -to 00:00:05 -filter:a "volume=10" -c:a libmp3lame z1.mp3
ffmpeg version 2024-03-04-git-e30369bc1c-full_build-www.gyan.dev Copyright (c) 2
000-2024 the FFmpeg developers
built with gcc 13.2.0 (Rev5, Built by MSYS2 project)
configuration: --enable-gpl --enable-version3 --enable-static --pkg-config=pkg
conf --disable-w32threads --disable-autodetect --enable-fontconfig --enable-icon
v --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --e
nable-libsnappy --enable-zlib --enable-librist --enable-libsrt --enable-libssh -
-enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sd
l2 --enable-libaribb24 --enable-libaribcaption --enable-libdav1d --enable-libdav
s2 --enable-libuavs3d --enable-libzvbi --enable-librav1e --enable-libsvtav1 --en
able-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvi
d --enable-libaom --enable-libjxl --enable-libopenjpeg --enable-libvpx --enable-
mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-li
bfribidi --enable-libharfbuzz --enable-liblensfun --enable-libvidstab --enable-l
ibvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-
ffnvcodec --enable-nvdec --enable-nvenc --enable-dxva2 --enable-d3d11va --enable
-libvpl --enable-libshaderc --enable-vulkan --enable-libplacebo --enable-opencl
--enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enabl
e-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --e
nable-libtwolame --enable-libvo-amrwbenc --enable-libcodec2 --enable-libilbc --e
nable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --ena
ble-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmys
ofa --enable-librubberband --enable-libsoxr --enable-chromaprint
libavutil 58. 40.100 / 58. 40.100
libavcodec 60. 41.100 / 60. 41.100
libavformat 60. 23.100 / 60. 23.100
libavdevice 60. 4.100 / 60. 4.100
libavfilter 9. 17.100 / 9. 17.100
libswscale 7. 6.100 / 7. 6.100
libswresample 4. 14.100 / 4. 14.100
libpostproc 57. 4.100 / 57. 4.100
Input #0, mp3, from 'luck1.mp3':
Duration: 00:00:54.02, start: 0.025057, bitrate: 133 kb/s
Stream #0:0: Audio: mp3 (mp3float), 44100 Hz, stereo, fltp, 133 kb/s
Metadata:
encoder : LAME3.100
Stream mapping:
Stream #0:0 -> #0:0 (mp3 (mp3float) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, mp3, to 'z1.mp3':
Metadata:
TSSE : Lavf60.23.100
Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp
Metadata:
encoder : Lavc60.41.100 libmp3lame
[out#0/mp3 @ 00000000001a7340] video:0KiB audio:79KiB subtitle:0KiB other stream
s:0KiB global headers:0KiB muxing overhead: 0.313639%
size= 79KiB time=00:00:05.00 bitrate= 129.5kbits/s speed=18.2x
