Only getting the volume of the bands about nvdsp HOT 7 CLOSED

You first Band Pass filter the data and then measure the volume of the new data? I don't see the problem?

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Would that work to find the volume of each band? That is if I had 10 bands then I want 10 volumes returned back. By filtering the data you wouldn't have the original anymore and in order to get the volume of each you would have to copy that original data 10 times.

Is there away around that?

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I guess you're then looking for a spectrum analyser, which can be done with a FFT algorithm. This has not been implemented in NVDSP.

Have you tried copying the data 10 times? They're small buffers anyway and if you measure one band at a time you can free the copied data immediately when you have the volume.

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Hi Bart,
Hope you would be doing well. Thanks a lot for your support and co-ordination, I really appreciate that. Currently I am working on reverb and echo effect on an audio file. I have read your comments on this issue and tried the Novocaine's sample code in order to achieve a simple delayed effect to mimic echo. I am facing a problem in that there is too much noise. I know I would be doing something wrong so I am sharing my code here with you so you may kindly review it and point out my mistake.

[self.fileReader setReaderBlock:^(float *data, UInt32 numFrames, UInt32 numChannels) {
       wself.ringBuffer->AddNewInterleavedFloatData(data, numFrames, numChannels);
    }];

    [self.fileReader play];
    self.fileReader.currentTime = 0.0;

    //int echoDelay = 11025;
    //float *holdingBuffer = (float *)calloc(16384, sizeof(float));
    [self.audioManager setOutputBlock:^(float *outData, UInt32 numFrames, UInt32 numChannels) {
        //[wself.fileReader retrieveFreshAudio:outData numFrames:numFrames numChannels:numChannels];

        //Grab the play-through audio
        wself.ringBuffer->FetchInterleavedData(outData, numFrames, numChannels);
        float volume = 0.8;
        vDSP_vsmul(outData, 1, &volume, outData, 1, numFrames*numChannels);

        //Seek back, and grab some delayed audio
        wself.ringBuffer->SeekReadHeadPosition(5);
        //wself.ringBuffer->FetchInterleavedData(holdingBuffer, numFrames, numChannels);
        //wself.ringBuffer->SeekReadHeadPosition(echoDelay);

        volume = 0.5;
        //vDSP_vsmul(holdingBuffer, 1, &volume, holdingBuffer, 1, numFrames*numChannels);
        //vDSP_vadd(holdingBuffer, 1, outData, 1, outData, 1, numFrames*numChannels);
    }];

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Hi @faizankhan1909, thank you for the kind words. First of of all, your code currently has the delay parts commented out, so I think you mean that it makes noise when you actually turn it on?

The first thing I noticed is that you combine a 80% volume and 50% volume channel. This results into 130% which is higher than the allowed 100%. What this means is that if the fetched data contains samples that hit the 1.0f (or -1.0f) mark, you will send out something like 0.8 * 1.0f + 0.5 * 1.0f = 1.3f. Samples above 1.0f or below -1.0f will clip horribly, so that' d generate a lot of noise.

I recommend you try to set volume = 0.2; for the delay instead of the 0.5 right now. If this solves it, that means it was the clipping.

Something else I can think of is that you might need to set the ReadHeadPosition of the ringBuffer back after you grab some delayed audio, but it seems you were already thinking about that. Let me know if this works for you.

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Hi @bartolsthoorn,
Thank you for the quick response. Sorry I could not reply earlier, as I was too much indulged in some project. Just wanted to tell you that your solution worked perfectly fine. And now there is no noise, furthermore the delayed samples are giving the impact of a surround sound that I wanted to achieve. Thank you once again :)

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@faizankhan1909 Thanks, that's nice to hear! 👍

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