I want to copy a part of the audio file, given the starting and ending point(in terms of time which I'll convert to packets or frames-is the conversion right?), and create a new audio file for the copied snippet. How do I copy?
Please advice.
Regards,
Namratha
For what file format(s)?
For the simplest and common .WAV (RIFF) file format, you can just copy the canonical 44-byte header (after checking to make sure the file is using only this simple format), update with the target file length, and then copy a selected sub-range of bytes (multiply time by sample rate by frame size) from the source file, and append that PCM data to the copied header. Apple's codec does not complain about audio files patched together this way.
For other formats, you might be able to convert them either to a simple WAVE file, or to an array of raw PCM samples of suitable sample rate, data type and endianess, and then do the above.
Related
Is it possible to create a PNG file with a predefined CRC? (kind of a programming challenge..)
I have a python script to generate hex codes with the target CRC, but I'm not sure how to make a valid PNG out of it.
BTW - it may be that I'm talking nonsense, but it sounds possible on theory (right?)
You can use spoof.c to do that, either at the level of a PNG chunk or at the level of the entire file. (Note that a PNG file does not contain a CRC of the whole thing, only CRCs of the chunks.)
I need to append 2 video files by appending their NSMutableData as I have already done this with audio files and it is done correctly but not with video files.
It may be because data bytes contain some header info and I will need to remove these bytes from the 2nd video but I don't know that how many bytes should I remove?
You haven't told us the file format in question, but generally:
Look up the specification for the file format in question and find out what the header looks like. Then code a solution based on this information.
There are many resources on the net with file format information, but one place you might want to look is http://www.wotsit.org/.
I would like to play some kind of text-to-speech with only numbers. I can record 10 wav files, but how can I combine them programmatically ?
For instance, the user types 1234, and the text-to-speech combines 1.wav with 2.wav, 3.wav and 4.wav to produce 1234.wav that plays "one two three four".
1) create a new destination sample buffer (you will want to know the sizes).
2) read the samples (e.g. using AudioFile and ExtAudioFile APIs) and write them in sequence to the buffer. You may want to add silence between the files.
It will help if your files are all the same bit depth (the destination bit depth - 16 should be fine) and sample rate.
Alternatively, if you have fixed, known, sample rates and bit depths for all files, you could just save them as raw sample data and be done in much less time because you could simply append the data as is without writing all the extra audio file reading programs.
The open source project wavtools provides a good reference for this sort of work, if you're ok with perl. Otherwise there is a similar question with some java examples.
The simplist common .wav (RIFF) file format just has a 44 byte header in front of raw PCM samples. So, for these simple types of .wav files, you could just try reading the files as raw bytes, removing the 44 byte header from all but the first file, and concatening the samples. Or just play the concatenated samples directly using the Audio Queue API.
I m building a voice recording application in iPhone. The recorded file is transfer to linux server and it need to be converted to wav file.
However, when I try to convert caf file using libsnd, it gives an error.
Error : Not able to open input file testfile.caf
For testing I converted some wave file to caf using libsnd and vice versa.
So I think that there is a problem of my recorded file in iphone.
Anyone has got such an experience ?
I hope can someone help me.
Thanks.
If you use AVAudioRecorder, AudioFile, or any of the other Core Audio APIs, you should be able to record directly to a WAV file, and skip the entire conversion process altogether.
But if you need to convert audio files, first check if the CAF file is a valid file. Does it play? Is the header correct? What is the data format? Is it compressed? Does libsnd support the data format?
(The data format is separate from the file format, which is just a container for various bits of data as well as the sample data. The data format could be PCM, or it could be compressed in any format such as MP3, AAC, uLaw, etc.)
I am using AVAudioRecorder in my app in order to create a .wav file based on user voice input. I want to be able to stuff" silence in the beginning of an audio file for some time before the actual recording is done.
How can I do this using AVAudioRecorder? Can I mention a time for which I want the "silence" to be recorded?
Thanks.
I haven't worked with the AVAudioRecorder in particular. But silence in PCM audio is just zeros as sample values. You could set the encoding of the AVAudioRecorder to PCM, store the file and then edit the file by prepending the desired number of zeros in the beginning. E.g. at 44100 Hz, 8-bit encoding, you'd add 44100 zero-bytes to the beginning of the file for each second of silence you would like to have. Hope this helps as an idea.
Note: Keep the file header of the PCM file intact and edit the "data chunk".
I'm sure there's probably a better way to do it, but how about simply pre-recording the silent stream you want and then concatenating that file with a file of the user generated content before saving out the final version?