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.
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I have to play various mp3 files on a given software without showing any information that could lead to the recognition of the played track (some kind of quiz). For that, I want to change the displayed track length to an arbitrary value. I can easily change up standard ID3 tags like "name", "artist" and so on. But changing the displayed track length seems to be more tricky, though...
Edit (after Vikram's response):
So far, I was able to manipulate the displayed track length by modifying the 'xing' header in a vbr encoded mp3 file. More precisely, I changed the bytes in the 'number of frames' section with a hex editor which lead to an mp3 that showed a modified track length according to:
Track length = Number of Frames * Samples Per Frame / Sampling Rate
with the file still being correctly played. This approach seems to work for winamp, vlc player and windows in general. Unfortunately, it does not seem to work for the proprietary software I have to use. When using that software, somehow the original track duration is still identified because a different calculation method is applied.
Any other ideas on how the track duration could be calculated resp. fooled into displaying an arbitrary value?
Thanks!
YES and NO.
Most of the .mp3 files have this extra info besides ID3 in XING header, which contains duration of the file.
You can modify this header to put wrong info.
Or you can simply remove this XING header!
There are two types of .mp3 files: CBR and VBR.
CBR is most common. So, using bitrate info, players still can estimate length of the audio for CBR.
For VBR this is not always correct!
So, the audio file you had was most likely an VBR encoded mp3 without XING header.
The track length is calculated by parsing the whole MPEG audio stream, a fairly straight forward process. XING (or similar) headers (correctly: frames) exist as a help like an additional index for seeking in the file, but it is not mandatory; it also only exists because 25 years ago in most cases it would have taken too much performance to fully parse files and keeping relevant data in memory when VBR was "invented". Metadata, where you could define incorrect track lengths (like the TLEN frame thru ID3v2) aren't mandatory either.
So: it's not possible. You rather found software/players that opt for performance without assuring everything they spot. Also no other file/stream format comes to my mind where a track length is mandatory and cannot be calculated by parsing the file.
How do you split an audio file into multiple parts using MATLAB ? Like I have an audio file composed of ten-twenty notes of a Piano, I need to split it into individual notes and store each note in a separate variable . Is it possible to do this with MATLAB ,if so how ? Can anyone please help me on this ?
If you want to do the splitting visually you can try "Simple Audio Editor" available in the file exchange.
http://www.mathworks.com/matlabcentral/fileexchange/19873-simple-audio-editor
I am the author of this program. Let me know if something does not work with that.
You can also try a free audio editor like audacity and export individual pieces to audio files. You can read each piece in MATLAB separately.
If you are looking to achieve this automatically you might need to ask this question in a Signal processing group.
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 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.
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.)