Amplify effect is not working, for Apple 14 plus users

Special Interest Groups Audio Processing
1

I downloaded audacity recently, so am thinking my version of audacity is latest one. I recorded an audio using Audacity, increased its volume (amplified effects). When I am sending this audio as attachment to my friends to test, then people with older iPhones are able to hear the increased volume in mp3 file, but people on iPhone 14 and higher are not hearing my voice in the mp3 at all. It sounds like an insect buzzing.

With audacity, I imported the original recorded audio, then did a select all, then effects>volume and compress>amplify.
Here I kept the settings which audacity showed, and did APPLY.
Within audacity, the volume increased. Playing in VLC etc also the volume increased.
But playing on iphone 14+, the volume is not audible at all.

I tried increasing the volume using ffmpeg then. Increased volume 10 times. Still same result. No voice audio audible in the file. If I add a background music to the audio using ffmpeg, then people on iPhone 14 and higher are only hearing the music, not my voice. What is happening here?
Why is the music audio playing fine, but the recorded audio is not?

I asked this same question on SuperUser forums, one guy uploaded the fixed AIFF file but he is not answering how he fixed the audio. He has even deleted the audio which he had sent me to test. It played fine on iphone 14 also, and on older iphones also.

Here’s the original input file as recorded by audacity:
tbrjar (dot) com (slash) audio_test (slash) input.mp3

And here is the fixed file which the person from SuperUser forums uploaded, which he deleted later:
tbrjar (dot) com (slash) audio_test (slash) hydranix_AIFF_input.aiff

Below is the ffmpeg command I am using to increase the volume, and its 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) 2000-2024 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

2

The left & right channels are (approximately) identical but out-of-phase. When they are combined to mono they cancel. That might have been done as a “stereo widening” effect or it could be a foul-up.

Click the little drop-down arrow to the left of the waveform and select Split Stereo to Mono.

To the left of the drop-down arrows, you’ll see the file name and an “X”. Click one of those X’s to delete one of the waveforms.

Now, you have a mono file that will play through both speakers. You can Amplify and Export.

1 Like
3

Hi @ DVDdoug Is there a keyboard friendly way of doing what you explained? I am blind person, using a screen reader to use laptop and audacity. If there is another way, keyboard friendly way, or even if via command line, please let me know. Thanks

4

Hopefully, someone else can help with that.

BUT, I wasn’t reading carefully and I missed that! It’s your recording so you didn’t do it intentionally .

You need to prevent it rather than fix it after recording.

The most common way for that to happen is if you are using a “pro” microphone with a balanced XLR connector and the wrong kind of adapter.

Pro (stage and studio) microphones are not compatible/interchangeable with computer mics. People doing pro or semi-pro recording usually use USB audio interface with XLR inputs.

Another option (other than an analog computer mic) is a USB microphone. There are some good “studio style” “podcast mics”.

Technically, what’s happening is - The balanced connection has two differential (out-of-phase) signals and a ground. (That’s done because it minimizes electrical noise pick-up when fed-into a balanced-differential input). A normal unbalanced connection has one signal and one ground. (With stereo, the left & right share the ground so a 3.5mm plug also has 3 connections.) If you plug that (mono) balanced signal into a regular stereo connection, one connection goes to the left and the other reverse-phase connection goes to the right and you’ve got a screwy out-of-phase stereo recording.

FYI - A single voice/instrument/microphone is mono . You can record in stereo with two microphones, one for the left and another for the right.

1 Like
5

Hi, many thanks for this explanation. I deeply appreciate. So I need to change the mic to prevent this from happening? I was using my laptop’s inbuilt mic so far. Will a 3.5mm jack mic prevent this from happening? Or will I need a USB mic? Which one is it, you have mentioned both and I didn’t understand this part properly.

Could I turn off a setting in audacity to prevent this from happening?

The recorded files seem to be working fine with all devices, except with the new iphone 14 and above versions, like iphone 14+.
I was earlier recording with laptop’s inbuilt mic and microsoft sound recorder that comes in windows accessories. Then some older android devices were not playing the audio.
Switching to audacity solved that problem.
But now newer iphones are not playing the sound…

6

Hi, I found this post on this same forums:

It says to convert stereo to mono, import audio file in audacity, open Tracks menu > Mix > then select first option (stereo down to mono)
When I did this and exported file as AIFF file, then played it in windows media player, it sounded exactly as how original input file was sounding in iphone 14+, like an insect buzzing.
This is the exact same problem I am facing on iphones, now the same is happening on windows also if the audio is converted to mono.
This is what needs to be fixed somehow.
Thanks