Re: Convert Waveform Audio To Mp3
Joseph Zyiuahndy
This free WAV to MP3 converter can help you convert WAV (Waveform Audio) audio to MP3 (MPEG Audio Layer 3) audio. The tool will try to maintain the audio quality of the source WAV file and create a high quality MP3 file as much as possible.
That sounds like a bit of a headache you've got on your hands! Converting WAV to MP3 should be straightforward, but it's super annoying when it doesn't go as planned, especially with the quality taking a hit or it dragging on forever.
Have you tried using AudioKies? It's an easy-to-use audio editing software that's pretty great for tasks like this. It's user-friendly and does a solid job of converting files without losing too much quality.
I don't know how much you know about computers, if you are computer savvy, I recommend you to use FFmpeg, it is a powerful software that can handle batch conversions quickly. You can use the following command to convert WAV to MP3 on Windows 10 computer.
To use FormatFactory click on the "Audio" button click on "Add File" select your WAV file and choose MP3 as the output format click on "Start". Start the conversion process and save your MP3 file after the conversion is finished.
One problem could be that you're casting your floats (doubles or singles) to integers (type unknown), and then to an I8. A 4 or 8 byte float won't fit in an I8, so you'll loose 1/4th or 1/8th of your data. We can't see this from an image.
It would be easier to spot problems if the code is clean. Bugs hide in messy code. 'Highlight' of the mess is the init array wire going to the right, then up, to the left, and down again. Can't imagine this works well for you, imagine how it is for us, seeing this for the first time...
Put a probe on the dbls and fxps to see that the fxps don't have anything to do with the dbls. I was assuming you'd want to send the binary doubles, but if you want to send values with a range from -128 to +128 casting is the wrong tool. The fxps are not really needed, you can simply convert (not cast) the dbls to i8s.
This is something new to me. I just assumed all audio formats were formatted as integer numbers representing sample amplitude at that integer type - e.g., a byte for 8-bit encoding, a word for 16-bit encoding, etc...
Is there a way to convert these lines that I made with audio waveform effect in After Effect, to be more round on the corners. When I increase displayed samples I get more roundnes, but to many lines, I also tryed Wave Warp, and that did't work for me. I wolud like only to round theses edges of the sound wave, one sine wave with rounded corners, that is moving accordingly to the audio. Thank you
Not really. You would have to write a whole bunch of convoluted expressions to a) extract the relevant info from converted audio keyframes and b) control the positions of the "waveform" points on a shape layer or similar oto which then a Rounded Corners modifier could be applied. Similar stuff would apply to alternate solutions using third-party plug-ins. That is to say unless there were a plug-in that does this right off the bat, it's pretty complicated or even impossible. I at least don#t know of any such plug-in, though naturally that doesn't rule out something like that exists. However, it's more likely that dedicated VJ-ing software offers more along the lines of these audio visualizations, so maybe look into that. Other than that it may be possible to get the results by processing the underlying audio file heavily, but that, too, could be infinitely complicated and time-consuming.
The "v" variable is a linear method that reads the minimum and maximum values from the Both Channels slider values. This is how that works: linear(t, tMin, tMax, value1, value2) where t is the value of the Both Channels Slider, tMin is the minimum value of the Both Channels slider as set by looking at the Graph Editor and checking the value graph, tMin is the maximum value. You'll have to put in your own numbers. The ones in the expression fit my audio file.
value1 and value2 are going to be multiplied by the Index value (layer number) of the Null layers. The more points/nulls you have in your comp, the lower the numbers need to be. Plus and minus 30 worked for my simple line, you'll have to make adjustments for your audio file and the number of points you want in your wave.
Come to think of it, you could use the layer name instead of Index and then just enter in the value you want to multiply for the layer name. That would probably be easier to work with, but it would require you to rename all of the null layers. Here's how that expression would look:
The posted expression is slightly different than the screenshot. The value of the layer name determines how far the null moves. The MaxT is the only value you need to edit in the expression for your audio amplitude.
Hello!
The easiest way that I've found to achieve this is by changing the Display Option to Analog Dots and maxing out the number of frequency bands. You essentially get a solid line when you do this.
Hello everyone.
I am doing a project where I want Arduino to detect audio (using an audio sensor), transform that audio to the corresponding waveform and then display that waveform on some external output (TFT LCD, PC). What would be your advice in achieving this?
Depending on the ambition You might need to check the performance of various controllers. The controllers use some time for an analog reading. That might limit the beauty of the curve You can plot. The plotting device also takes some little time to access.
Specify Your needs.
Thank you for answering.
Arduino in question is MEGA 2560 R3. I am aware that the analog reading will take some time, the goal is to minimize that period as much as the hardware can provide. The looks of the waveform are not a priority.
My goal is to show the waveform of the input signal realtime, will need some FFT library for the Arduino. Where do I go from there?
Thanks, will do. I read that the Arduino has a built in A/D converter and with the audio sensor, I guess I should be fine with the analog to digital conversion. The thing that is confusing me about this project is how can I show the now digital data I got as a waveform?
Most people draw a graph of the data on a screen of some sort. For Arduino there is the Serial Plotter, which produces output similar to the graph below. The blue curve is the input signal, the red curve is a severe sampling artifact, caused by sampling the 440 Hz tone at 352 Hz.
Do you know where the Sound functions are on the Block Diagram Palette? Look at the Help for Sound Output Write, which tells you most of what you need to know. It also should point you to an Example file, such as Generate Sound, which you can examine, play with, and modify to suit your needs (I recommend modifying a copy of the Example).
Oops -- I just realized I answered the Wrong Question -- I showed you how to play a data file as though it were a Sound File. However, to play a sound, the first step(s) are/is to create some Sound Data, which is just what the Sound File functions want. So if you can Play it, you can Write it. Again, use LabVIEW's Help and check out the Examples (with links at the bottom of the Help document).
Thank you for ur answer. Actually my qestion is convert a txt file( i convert signal which showing in labview graph to txt file )which include numbers convert to sound file or just signal convert to audio file in labview. Is that possible?
I'm just wondering: Could I be saving a significant amount of disk space by converting all of my WAV originals to FLAC? Or would I lose audio data that might be important in the future if I want to (re)process my recordings?
The main reason not to encode everything as FLAC is simple convenience. For many applications (using them in most DAWs, for example) you'll have to transcode them back to .WAV which takes some time. Not much, but some. Whether this matters or not is entirely up to your workflow.
The only reason I can think of not to save things as FLAC is compatibility. Not all software will understand a FLAC file (example: iTunes, Logic) but I've never met software that couldn't handle WAV files. You'll definitely save on space at no cost to quality but you may find yourself converting from FLAC back to WAV to work with the files. If that takes time, the question becomes what's worth more? Your time or your disk space?
Flac saves you about 60% of the space and gives you spiffing tag info, but at a price I am not going to pay at any moment in my recording-workflow: it costs CPU power to decode them. This is highly inconvenient when you are working on a project, especially when you need your CPU to do DSP-related things. If you would convert a FLAC file back to WAV (disk access time not counting) you get a general idea of CPU time this process takes during the playback of the entire track.
There are DJ programs that will add 'hotcues' to sections of audio, and you could probably create a cue sheet as well. But if you mark a section of audio as bad, and decide to delete it, any cues behind that section will stay with the audio. If you cue up a flac file and then edit it, all the cues after the edit will be out of sync.
Uncompressed WAV or AIFF will sound better than FLAC for a very simple reason. FLAC requires your processors to decompress and this produces more electrical noise in your system. The only way to avoid this is to properly isolate the DAC stage from the analogue output, something only a few brands like PS Audio does.
Have you ever noticed that when you play music on a sound platform, you may also bring up a visual? Audio waveforms are often these wave-like lines that provide psychedelic-looking visuals. You're looking at a waveform that depicts and moves in tandem with an audio stream that amplitude fluctuates over time.
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