can someone define/explain briefly (or with some detail-> up to you) the following terms:

  1. Block Processing? and what are the differences between it and Sample Processing?
  2. (removed)
  3. What is a 3D Audio System and what is relation between it & Head Related Transfer Function (HRTF)? and how can HRTF be applied in Headphones?
  4. (removed)
  5. The relationship between bit rate, sample rate and bit depth? Is there a formula that combines all terms?
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    $\begingroup$ wow!! good questions! $\endgroup$ – robert bristow-johnson Feb 5 at 23:14
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    $\begingroup$ ok but points 3 and 4 can hardly be brief... $\endgroup$ – Fat32 Feb 5 at 23:30
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    $\begingroup$ Welcome to DSP.se Leo.! I think this post would have deserved separate questions. $\endgroup$ – anpar Feb 6 at 7:58
  1. Block Processing (sometimes called Vector Processing) is a programming technique similar to, but not exactly the same as First-In-First-Out (FIFO) buffering. It is double buffering of the samples coming into the process (be they from an A/D converter or a stream) and going out (to a D/A converter or another stream). Block processing divides the processing time into slots of the entire block. I.e. in audio, 32-sample blocks might be common in 48 kHz samplerate data. So in one block time (which is 32/48000 second), the DSP algorithm is presented with 32 valid input samples at the beginning of the time period, and has the entire block time (of 32 sample periods) to compute 32 valid output samples. This allows the cost of some of the work, like loading/saving states and loading coefficients and initializing accumulators and such, to be amortized over all 32 samples. This makes the processing more efficient. But the cost is 2 blocks of delay, due to the double-buffering. And if there is feedback from one processing algorithm to another, that block of samples being fed back is from the previous block of samples, not the current block. So feedback has a delay of 1 block of samples.
  2. HRTF, I'll have to come back to that. Maybe someone else can pick it up.
  3. Sample rate is a well-known definition. It is the number of samples per unit time expressed as frequency. Usually it's samples-per-second expressed as Hz. Bit depth is the same as the word-width of the samples. CDs have 16-bit depth. Decent audio processing DAWs have 24 or 32-bit depth. Bit rate for uncompressed audio is simply the sample rate times the bit depth (times the number of channels). It's the data rate of the audio stream. For compressed audio, the bit rate is much reduces and continues to be the data rate of the audio stream.
  • $\begingroup$ Thanks Robert for your help. Will see if someone answers the rest. $\endgroup$ – Leo Feb 6 at 0:05
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    $\begingroup$ @Leo let's actually NOT do that. Usually, we try to maintain a one-question-per-post ratio, so that we don't end up in this situation. rbj has given an excellent answer to most of your questions. If anyone, he'd be deserving of your acceptance of his answer. I propose the following: I'm not sure what L* means, but I'd simply edit your question to replace these two bullet points (2. and 4.) with --- and ask them, separately, in two new questions here (add a link to these new questions to your question here). That way, you can still get answers, whilst being able to accept this answer. $\endgroup$ – Marcus Müller Feb 6 at 7:10
  • $\begingroup$ the L* was just a place holder. i'll try to get back to HTRF later. but it really is a good idea to split multiple questions into separate questions. $\endgroup$ – robert bristow-johnson Feb 6 at 19:57
  • $\begingroup$ @MarcusMüller: Totally agree. It's much better when it comes to searching and linking answers. $\endgroup$ – jojek Feb 12 at 9:09
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    $\begingroup$ If we all agree on this, I'll edit @Leo's question and rbj's answer to exclude 2 and 4. (I'll simply add (removed) in place to not disturb the counting) $\endgroup$ – Marcus Müller Feb 12 at 11:59

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