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I am dealing with an audio file that shows sudden spikes:

When I try to normalize the audio file, the audio program sees this spike, notices that it is at 0 dB and won't normalize the audio any more.

To solve this issue, I have applied a Limiter that limits to -3 dB. I have used both Steinberg Wavelab and Audacity Hard Limiter. Instead, they both diminish the entire audio volume.

Both do not eliminate this spike.

Edit: I have found out that "Hard Clip" in Audacity does what I need, but now I still want to finish my own approach.

So I was thinking that they perhaps do not work correctly.

Then I tried to write my own limiter in VB6 in order to have full control over what's happening.

To do that, I'm loading the audio data of a wav file like this:

(I have stripped the process down very much)

I keep getting the error: "processing abandoned, this video could not be processed" whenever I try to upload a 107 minute video of an audio recording (just myself rambling thoughts into a microphone the entire time, just a black screen for video). The original audio file is .m4a, and I am trying to convert it into a video file to upload to YouTube. I have tried all the methods that I could find mentioned online, and every time I still get the exact same error. I have tried:

• converting from .m4a to .mp4 using VLC and uploading the .mp4
• converting from .m4a to .MOV using VLC and uploading the .MOV
• converting from .m4a to .mp4 using various online conversion tools and uploading the .mp4
• converting from .m4a to .MOV using various online conversion tools and uploading the .MOV
• converting from .m4a to .mp3 using various online conversion tools, then converting from .mp3 to both .mp4 and .MOV and uploading the .mp4 and .MOV
• verifying my youtube account and enabling the verification feature eligibility settings in the youtube creator studio to allow uploads longer than 15 minutes
• re-uploading the same file after the processing failed to try and restart the process
• trying different browsers

I also tried exporting the video file again using the default video editor on Windows (the only video editor I have), but I couldn't upload either the audio file (not an option to choose as an import file type) or the video files I created (gave error: "these files have properties that can't be read". I don't know why this error came up either, but maybe it's because the file is corrupted? This wouldn't make sense to me though because I'm able to play it fine (all file types) in the windows default player, VLC, and Audacity. I was able to convert the audio into the video files in VLC too, so I don't know why I can't do the same in the windows default editor.

I realize that similar questions have been asked before on here, but none actually give a solution that works for my problem. Any insight into why I may be having this problem, and how I may be able to solve it, would be very much appreciated.

67-year-old grannie here. My 12-year-old programmable keyboard died and I will try to use AutoHotkey in its stead to do simple keyboard macros. Here's the first one I need to create. When I press F12, I want it to send these keystrokes to Audacity (but it could be any program; the keystrokes are simply shortcut keys to do specific functions): Alt+G S Enter Enter

How would I program that in AutoHotkey?

Disclaimer : I posted for the same issue (with a different point of view) here

I use a SPH0645 I2S microphone with a custom driver.

It seams properly recognized by alsa :

I need a minimum of 3/4 different tts voice but unfortunatenly I have only one voice.

This because I have only one Italian neural voice (Diego) and the others are all standard voice and the quality is much worse.

The final objective is create a voice over for 3/4 persons minimum and I can't use the some exact voice.

For this reason, I like to create some variant started by the only one neural voice that I have, that gives the impression of a voice of other people all of this without seem unnatural.

Actually I have Adobe Audition, Audacity , Ircam Trax, ffmpeg and apart this I can use SSML with API (in this case microsoft Azure).

I don't known what are the effects and in what measure use it without damage the voices.

In short I ask what is the best way to do using the software that I have or other if I will get better results.

Thanks !

I'm trying to make a record command in my discord.js bot. My code so far is:

I'm trying to record sound from the microphone but it is getting difficult. I have tried several ways and it doesn't work. I created a project only for testing which is going to be implemented in a bigger project later. Here is the code of the project in question:

I'm trying to decode an AAC audio stream in an ADTS container, which is streamed from an external hardware H264 encoder.

I've parsed out the ADTS and it tells me I've got a 2 channel, 44100 AAC Main profile frame. I setup the extra data bytes for the ffmpeg decoder and decode the frame successfully? as follows:

(pseudo c++ code)

setup the decoder:

I am trying to play a byte buffer with NAudio. I created a winforms application for the task.

But when I play the sound (created with Math.sin) subsequently, the volume goes lower and lower for every playback. Why?