I am working on a project in matlab to take a predetermined audio file and change the sample rate dynamically from data generated in real time. I have hit a very stubborn roadblock with the dsp.audioplayer object. It doesn't allow change in either the sample rate or the sample size once it's state is locked. My thoughts right now are to vary the sample size that I pull from the wav file and scale it using a fir rate conversion filter. Is this an option worth perusing? Are there any other ways around this problem?
In the latest MATLAB release samplerate is tunable in dsp.audioplayer. Tunable Means you can change the property value after the object is locked.
Your workaround is good when this is not possible.
Related
I have a wav file pulled up in MATLAB, and I can see it's sample rate. All I need to do is change this 1 number. Everything else in the file will remain uncahnged. (The resulting sound would play at a different speed but would have an identical array of sample data.)
The reason I need to do this is because MATLAB seems to freak out when I tell it to open something sampled at anything other than 8k. All I need MATLAB for is to edit the file, so the sample rate really doesn't matter at all, since I'll be putting it back into a wav file when I'm done. So I either need to be able to change the value in the wav file that stores the sample rate, or to get MATLAB to change the sample rate it prefers from 8k to the sample rate that my files were recorded at.
if you just want to change the sampling frequency, here is the code, but it would distort the original wav file. If you decrease the sampling frequency, then the beat and music would be very slow.
Code:
[y, fs, nbits]=wavread('stego_lab');
fs2=11025;
wavwrite(y,fs2,nbits,'stego2_lab.wav');
sound(y,fs2,nbits)
you can hear it but the samples will remain the same.
Hope it helps.
There is the SOX tool, which should help you in that respect, and it comes on almost any platform - http://sox.sourceforge.net
There is also libsndrate, libsamplerate, libsndfile and others, that might have executables too.
Try this solution
[x,fs] = wavread('infile.wav');
<br>[p,q] = rat(16000/fs) % to convert to 16k sample rate</br>
<br>y = resample(x,p,q); % signal package require
wavwrite(x,16000,'outfile.wav');
I have a video in which the frame rate is not constant over the length of video. How can i get the actual time information from the video using MATLAB?
Use the function mmread() available from the file exchange: (link). More generally, the best way is to be using a Unix system and issue system calls to a better program, like ffmpeg or mencoder, which can give you frame-specific timing information. Doing this with Matlab only would be a real pain. There may be a toolbox function for it, but then your code only works for other people who also have the toolbox...
I'm not sure if it's possible to achieve what I want, but basically I have a NSDictionary which represents a recording. It's a timeline of what sound id was played at what point in time.
I have it so that you can play back this timeline/recording, and it works perfectly.
I'm wondering if there is anyway to take this timeline, and export it as a single sound that could be saved to a computer if the device was synced with iTunes.
So basically I'm asking if I can take a timeline of sounds, play it back and have these sounds stitched together as a single sound, that can then be exported.
I'm using OpenAL as my sound framework and the sound files are all CAFs.
Any help or guidance is appreciated.
Thanks!
You will need:
A good understanding of linear PCM audio format (See Wikipedia's Linear PCM page).
A good understanding of audio sample-rates and some basic maths to convert your timings into sample-offsets.
An awareness of how two's-complement binary numbers (signed/unsigned, 16-bit, 32-bit, etc.) are stored in computers, and how the endian-ness of a processor affects this.
Patience, interest in learning, and a strong desire to get this working.
Here's what to do:
Enable file sharing in your app (UIFileSharingEnabled=YES in info.plist and write files to /Documents directory).
Render the used sounds into memory buffers containing linear PCM audio data (if they are not already, i.e. if they are compressed). You can do this using the offline rendering functionality of Audio Queues (see Apple audio queue docs). It will make things a lot easier if you render them all to the same PCM format and sample rate (For example 16-bit signed samples #44,100Hz, I'll use this format for all examples), and use the same format for your output. I recommend starting off with a Mono format then adding stereo once you get it working.
Choose an uncompressed output format and mix your sounds into a single stream:
3.1. Allocate a buffer large enough, or open a file stream to write to.
3.2. Write out any headers (for example if using WAV format output instead of raw PCM) and write zeros (or the mid-point of your sample range if not using a signed sample format) for any initial silence before your first sound starts. For example if you want 0.1 seconds silence before your first sound, write 4410 (0.1 * 44100) zero-samples i.e. write 4410 shorts (16-bit) all with zero.
3.3. Now keep track of all 'currently playing' sounds and mix them together. Start with an empty list of 'currently playing sounds and keep track of the 'current time' of the sample you are mixing, for each sample you write out increment the 'current time' by 1.0/sample_rate. When it gets time for another sound to start, add it to the 'currently playing' list with a sample offset of 0. Now to do the mixing, you iterate through all of the 'currently playing' sounds and add together their current sample, then increment the sample offset for each of them. Write the summed value into the output buffer. For example if soundA starts at 0.1 seconds (after the silence) and soundB starts at 0.2 seconds, you will be doing the equivalent of output[8820] = soundA[4410] + soundB[0]; for sample 8820 and then output[8821] = soundA[4411] + soundB[1]; for sample 8821, etc. As a sound ends (you get to the end of its samples) simply remove it from the 'currently playing' list and keep going until the end of your audio data.
3.4. The simple mixing (sum of samples) described above does have some problems. For example if two samples have values that add up to a number larger than 32767, this cannot be stored in a signed-16-bit number, this is called clipping. For now, just clamp the value to 32767, and get it working... later on come back and implement a simple limiter (see description at end).
Now that you have a mixed version of your track in an uncompressed linear PCM format, that might be enough, so write it to /Documents. If you want to write it in a compressed format, you will need to get the source for an audio encoder and run your linear PCM output through that.
Simple limiter:
Let's choose to limit the top 10% of the sample range, so if the absolute value is greater than 29490 (int limitBegin = (int)(32767 * 0.9f);) we will scale down the value. The maximum possible peak would be int maxSampleValue = 32767 * numPlayingSounds; and we want to scale values above limitBegin to peak at 32767. So do the summation into sampleValue as per the very simple mixer described above, then:
if(sampleValue > limitBegin)
{
float overLimit = (sampleValue - limitBegin) / (float)(maxSampleValue - limitBegin);
sampleValue = limitBegin + (int)(overLimit * (32767 - limitBegin));
}
If you're paying attention, you will have noticed that when numPlayingSounds changes (for example when a new sound starts), the limiter becomes more (or less) harsh and this may result in abrupt volume changes (within the limited range) to accommodate the extra sound. You can use the maximum number of playing sounds instead, or devise some clever way to ramp up the limiter over a few milliseconds.
Remember that this is operating on the absolute value of sampleValue (which may be negative in signed formats), so the code here is just to demonstrate the idea. You'll need to write it properly to handle limiting at both ends (peak and trough) of your sample range. Also, there are some tricks you can do to optimize all of the above during the mixing - you will probably spot these while you're writing the mixer, be careful and get it working first, then go back and refactor/optimize if needed.
Also remember to consider the endian-ness of the platform you are using and the file-format you are writing to, as you may need to do some byte-swapping.
One approach which isn't too hard if your files are stored in a simple format is just to combine them together manually. That is, create a new file with the caf format and manually put together the pieces you want.
This will be really easy if the sounds are uncompressed (linear PCM). But, read the documents on the caf file format here:
http://developer.apple.com/library/mac/#documentation/MusicAudio/Reference/CAFSpec/CAF_spec/CAF_spec.html#//apple_ref/doc/uid/TP40001862-CH210-SW1
I want to add a few bytes of data to a sound file (for example a song). The sound file will be transmitted via radio to a received who uses for example the iPhone microphone to pick up the sound, and an application will show the original bytes of data. Preferably it should not be hearable for humans.
What is such technology called? Are there any applications that can do this?
Libraries/apps that can be used on iPhone?
It's audio steganography. There are algorithms to do it. Refer to here.
I've done some research, and it seems the way to go is:
Use low audio frequencies.
Spread the "bits" around randomly - do not use a pattern as it will be picked up by the listener. "White noise" is a good clue. The random pattern is known by the sender and receiver.
Use Fourier transform to pick up frequency and amplitude
Clean up input data.
Use checksum/redundancy-algorithms to compensate for loss.
I'm writing a prototype and am having a bit difficulty in picking up the right frequency as if has a ~4 Hz offset (100 Hz becomes 96.x Hz when played and picked up by the microphone).
This is not the answer, but I hope it helps.
I'm making an Iphone game, we need to use a compressed format for sound, and we want to be able to loop SEAMLESSLY back to a specific sample in the audio file (so there is an intro, then it loops back to an offset)
currently THE ONLY export process I have found that will allow seamless looping (reports the right priming and padding frame numbers, no clicking when looping ect) is using apple's afconvert to a aac format in a caf file.
but when we try and encode to lower bitrates, it automatically re samples the sound! we do NOT want to have the sound re sampled, every other encoder I have encountered has an option to set the output sample rate, but I can't find it for this one.
on another note, if anyone has had any luck with seamless looping of a compressed file format using audio queues, let me know.
currently I'm working off the information found at:
http://developer.apple.com/mac/library/qa/qa2009/qa1636.html
note that this DID work PERFECTLY when I left the bitrate for the encode at default (~128kbs) but when I set it to 32kbps - with the -b option - it resampled, and looping clicks now.
It needs to be at least 48kbps. 32kbps will downsample to a lower sample rate.
I think you are confusing sample rate (typical values: 32kHz, 44.1kHz, 48kHz) and bit rate (typical values: 128kbps, 160kbps, 192kbps).
For a bit rate, 32kbps is extremely low. Sound will have bad quality at this bit rate. You probably intended to set the sample rate to 32kHz instead, which is also not outright typical, but makes more sense.
When compressing to AAC and uncompressing back to WAV, you will not get the same audio file back, because in AAC, the audio data is represented in a completely different format than in raw wave. E.g. you can have shifts by few microseconds, which are necessary to convert to the compressed format. You can not completely get around this with any highly compressed format.
The clicking sound originates from the sudden change between two samples which are played in direct succession. This is likely taking place because the offset to which you jump back in your loop does not end up to be at exactly the same position in the AAC file as it was in the WAV file (as explained above, there can shifts by microseconds).
You will not get around these slight changes when compressing. Instead, you have to compensate for them after compression by adjusting the offset. That means you have to open the compressed sound file in an audio editor, e.g. Audacity, and manually find another offset close to the original one, which is suitable for looping.
How to find an offset which is suitable for looping?
Zoom in to the waveform's end. Look at how the waveform looks there. Then zoom in to the waveform at the original offset and search in its neighbourhood for an offset at which the waveform connects seamlessly to the end of the waveform.
For an example how this shoud look like, open the uncompressed audio file in the audio editor and examine the end of the waveform and the offset there.