Re: Trying to understand frames/samples, frames within samples etc.
Re: Trying to understand frames/samples, frames within samples etc.
- Subject: Re: Trying to understand frames/samples, frames within samples etc.
- From: Ash Gibson <email@hidden>
- Date: Mon, 03 Sep 2012 10:22:20 +1000
Hi Aran,
Thanks for replying. I understand about making sure you are using the correct type etc. In the case of my current project, this is my setup:
AVAssetReader with an AVAssetReaderAudioMixOutput that is configured for 16-bit PCM samples. I read the buffers into a circular buffer (TPCircularBuffer).
I then have an audio graph with remoteIO and a mixer and a callback on the input to the mixer where I pull the values out of the circular buffer and push them into ioData. The ASBD is configured to be 16bit LPCM as well. This is all working perfectly and my audio plays.
My next step is to perform some analysis on the audio in order to draw a visualisation of the data, sampled every .5 seconds for example. I also want to somehow get a reading of the lower frequencies (bass) so that I can do some visualisations that react to the bass only. (low pass filter?)
In terms of the visualisation, I am not trying to build a visualisation of the entire audio file from 0 secs -> 3 mins, I want to take a snapshot in time and visualise the frequency range
----
So, in order to progress into the complicated world of FFT and the Accelerate framework I am trying to actually understand what the data in an audio buffer represents. What I can gather from your reply is that if i get an audio buffer at 1.00 seconds, each frame within the 16bit integer sample is a moment in time from 1.00 onwards. For example, sample[0] is 0.00, sample[1] is 0.01 etc.
Each of these frames is a numeric value and depending on the data type, it will be a value within a certain range. What is this numeric value? Is it a result of an algorithm/formula or similar?
Just to confirm, when I get an audio buffer and convert it to a 16bit integer and then loop through it using the inNumberFrames, each of the values are a different point in time?
I come from a background in flash development where I could use computeSpectrum to generate an array of values with [0] being the lowest frequency(bass) and [length] being the highest frequency. I understand that there is more work to be done in ios but am not sure how to bridge the gap in my learning.
Any help is greatly appreciated.
Cheers,
Ash
On 3 September 2012 09:19, Aran Mulholland
<email@hidden> wrote:
It is like this.
sample[0] sample[1] sample[2]
time ------------------------------------------------------------------>
Sometimes (depending on your sample format) you can get a stereo
interleaved stream where you will get
sample[0](left) sample[0](right) sample[1](left) sample[1](right) etc..
If the streams are not interleaved and you have a stereo signal you
can access the left and right channels by
SInt16* samples = (SInt16*)(ioData->mBuffers[0].mData); //left channel
SInt16* samples = (SInt16*)(ioData->mBuffers[1].mData); //right channel
Also when you are doing the cast to (SInt16 *) you have to make sure
the type you are casting to is correct. Some sample formats use 32 bit
ints, some use floats, some use other formats.
On Sun, Sep 2, 2012 at 9:46 PM, Ash Gibson <
email@hidden> wrote:
> In a callback function, I get the inNumberFrames and ioData args. I then
> write something like this:
>
> SInt16* samples = (SInt16*)(ioData->mBuffers[0].mData);
>
> which will give me a 16-bit signed integer which is like an array of
> integers?
>
> I can then loop through it with the inNumberFrames as the total number of
> elements e.g.
>
> for (int i=0; i < inNumberFrames; i++) {
>
> something = samples[i];
>
> }
>
>
> When looking at each of these samples that are created inside the for loop
> e.g. sample[0], sample[1], sample[2] etc. I am trying to visualise what they
> represent in terms of sound and time.
>
> Is it like this?
>
> sample[0] sample[1] sample[2]
> time ------------------------------------------------------------------>
>
> or are these samples actually different parts (e.g. bass, mid range, high)
> of a sound caught at a point in time e.g.
>
>
> sample[2] sample[2] sample[2]
> sample[1] sample[1] sample[1]
> sample[0] sample[0] sample[0]
> time ------------------------------------------------------------------>
>
>
>
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--
Ash Gibson
0410 329 355
www.smartface.com.au
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