SPO600 Algorithm Selection Lab
Lab 4
Background
- Digital sound is typically represented, uncompressed, as signed 16-bit integer signal samples. There is one stream of samples for the left and right stereo channels, at typical sample rates of 44.1 or 48 thousand samples per second, for a total of 88.2 or 96 thousand samples per second (kHz). Since there are 16 bits (2 bytes) per sample, the data rate is 88.2 * 1000 * 2 = 176,400 bytes/second (~172 KiB/sec) or 96 * 1000 * 2 = 192,000 bytes/second (~187.5 KiB/sec).
- To change the volume of sound, each sample can be scaled by a volume factor, in the range of 0.00 (silence) to 1.00 (full volume).
- On a mobile device, the amount of processing required to scale sound will affect battery life.
Basic Sound Scale Program
Get the files for this lab on one of the SPO600 Servers -- but you can perform the lab wherever you want.
- Unpack the archive
/public/spo600-algorithm-selection-lab.tgz
- Examine the
vol1.c
source code. This program:- Creates 5,000,000 random "sound samples" in an input array (the number of samples is set in the
vol.h
file). - Scales those samples by the volume factor 0.75 and stores them in an output array.
- Sums the output array and prints the sum.
- Creates 5,000,000 random "sound samples" in an input array (the number of samples is set in the
- Build and test this file.
- Does it produce the same output each time?
- Test the performance of this program. Adjust the number of samples as necessary to get measurable results.
- How long does it take to run the scaling?
- How much time is spent scaling the sound samples? Be sure to eliminate the time taken for the non-scaling part of the program (e.g., random sample generation). (How will you do this?)
- Do multiple runs take the same time? How much variation do you observe? What is the likely cause of this variation?
- Is there any difference in the results produced by the various algorithms?
Alternate Approaches
The sample program uses the most basic, obvious algorithm for the problem. Let's call this "Algorithm 0", or the "Naive Algorithm".
Try these alternate algorithms for scaling the sound samples by modifying copies of vol1.c
. Edit the Makefile
to build your modified programs as well as the original. Test each approach to see the performance impact:
- Pre-calculate a lookup table (array) of all possible sample values multiplied by the volume factor, and look up each sample in that table to get the scaled values.
- Convert the volume factor 0.75 to a fix-point integer by multiplying by a binary number representing a fixed-point value "1". For example, you could use 0b100000000 (= 256 in decimal) to represent 1.00, and therefore use 0.75 * 256 = 192 for your volume factor. Multiply this fixed-point integer volume factor by each sample, then shift the result to the right the required number of bits after the multiplication (>>8 if you're using 256 as the multiplier).
Deliverables
Blog about your experiments with a detailed analysis of your results, including memory usage, time performance, accuracy, and trade-offs.
Important! -- explain what you're doing so that a reader coming across your blog post understands the context (in other words, don't just jump into a discussion of optimization results -- give your post some context).
Optional - Recommended: Compare results across several implementations of AArch64 and x86_64 systems. Note that on different implementations, the relative performance of different algorithms will vary; for example, table lookup may outperform other algorithms on a system with a fast memory system (cache), but not on a system with a slower memory system.
- For AArch64, you could compare the performance on AArchie against another 64-bit ARM system such as a Raspberry Pi 3 or an ARM Chromebook.
- For x86_64, you could compare the performance of different processors, such as xerxes, your own laptop or desktop, and Seneca systems such as Matrix, Zenit, or lab desktops.
Things to consider
Design of Your Tests
- Most solutions for a problem of this type involve generating a large amount of data in an array, processing that array using the function being evaluated, and then storing that data back into an array. Make sure that you measure the time taken in the test function only -- you need to be able to remove the rest of the processing time from your evaluation.
- You may need to run a very large amount of sample data through the function to be able to detect its performance. Feel free to edit the sample count in
vol.h
as necessary. - If you do not use the output from your calculation (e.g., do something with the output array), the compiler may recognize that, and remove the code you're trying to test. Be sure to process the results in some way so that the optimizer preserves the code you want to test. It is a good idea to calculate some sort of verification value to ensure that both approaches generate the same results.
- Be aware of what other tasks the system is handling during your test run.
Analyzing Results
- What is the impact of various optimization levels on the software performance? (For example, compiling with -O0 / -O1 / -O2 / -O3)
- Does the distribution of data matter? (e.g., is there any difference if there are no absolute large numbers, or no negative numbers?)
- If samples are fed at CD rate (44100 samples per second x 2 channels x 2 bytes per sample), can each of the algorithms keep up?
- What is the memory footprint of each approach?
- What is the performance of each approach?
- What is the energy consumption of each approach? (What information do you need to calculate this?)
- Various machines within an architecture have very different performance profiles, energy consumption, and hardware costs -- so it's not reasonable to compare performance between machines, but it is reasonable to compare the relative performance of the algorithms in each context. Does the ratio of performance of the various approaches remain constant across the machines? Why or why not?
- What other optimizations can be applied to this problem?