Copy wav files

Bytes, or number of samples? My suspicion is that the code you represent by [ How do you read the data into the header? Do you have to worry about big-endian vs little-endian encoding for the file header? You've also omitted some global variables and some other stuff. Please reread about how to create an MCVE minimal reproducible example and make this into one.

What are you actually trying to achieve? Do you just want to make a verbatim copy of the wav file? If yes, your question boils down to "how can I make a copy of a file". JonathanLeffler This is right, I didn't need to completely understand the structure to copy. But, later, my goal is to make some operation like add a chanel, reverse a wave This is why, for copy, I do this code. This is the size of the data, without the header.

Drop files here. Codec "Without reencoding" copies the audio stream from the input file into output without re-encoding if possible. Audio Channels: Mono 1. This setting is most useful when downmixing channels e. For Mac users, iMovie can suit your needs. In terms of easiness, wide format support, the ability to combine audio files without quality compress, and accompany features to edit MP3 ID3 tags, audio metadata, the best way to efficiently merge audio files is to use VideoProc Converter.

No matter you only want to splice two songs together, or join hundreds of audio into one track, the tool works nicely for Windows and Mac users. Cecilia Hwung is the marketing manager of Digiarty Software and the editor-in-chief of VideoProc team. She pursues common progress with her team and expects to share more creative content and useful information to readers. She has strong interest in copywriting and rich experience in video editing tips. Create cinematic videos and beyond. Learn More.

VideoProc Converter One-stop video processing software. Convert, transcode, compress, download and record. VideoProc Converter Convert, transcode, compress, download and record.

You have audiobooks but the player cannot play the chapters in the correct order. The solution is to combine audio files orderly or edit their ID3 tags, both can be done with method 1 below. One is mono, the other is stereo audio , or differs in bitrate, sample rate, and codec. In these cases, you can convert them to stereo and parameter settings, and combine audio files into one.

In method 1, you can use one tool to complete all the tasks in one go. Photo by: whathifi 1. About The Author. To put the value into this range, we multiply by and, with the floor function, round down. Only sound card manufacturers deal with the actual process of creating sound, so the info about the software logic is a little hard to get and a lot of device manufacturers keep their code a proprietary secret.

But for people like us who want to learn the actual process of making sound, this is the holy grail. So negative values in the signal can mess this up? I was able to run the code, and tried on a piano sequence, but it comes out noisy.. My data range is from , Not really. In line , you are reading into the data buffer the audio data for all channels I was using a 2 channel example and it was 4 bytes ; however inside the for loop [ line ] looping over each channel you are always using the data for the first channel.

I had to modify this so that each channel data was recorded correctly. Can you clarify? I still get a noisy data from my player, but thats a different issue. Its been a few years since I wrote the code so I will have to go over it once to see if there is any issue. At first glance, what you say could be correct, because the channel loop is using the same data extract in every iteration. I believe the pointer within the data array should be incremented within the channel loop.

If you have already made the change and its working fine , you could probably post the fix. I will go over the code in the next couple of days and apply your fix. No, I dont have it working satisfactorily. Once I have it working, I will be happy to share — in fact we are hoping to use this this for an open source educational toolkit. I got the parser working yesterday. There are two issues with your original code.

One I mentioned above, where for multiple channels, you need to update the pointer. Secondly, this code wont work correctly for negative signal values.

For this you need to do the following where you do the endian conversion:. I have a version for C I can send you with these changes. I will send a link to that with acknowledgements for your approval.

We would like to use that in an open source educational toolkit we are building. To check the correct results from matlab, each sample in matlab is stored as a double instead of an int in this example. Also, the values still differ from the ones parsed by matlab.