diff options
Diffstat (limited to 'services/audioflinger/test-resample.cpp')
| -rw-r--r-- | services/audioflinger/test-resample.cpp | 139 |
1 files changed, 118 insertions, 21 deletions
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp index 3f2ce55..66fcd90 100644 --- a/services/audioflinger/test-resample.cpp +++ b/services/audioflinger/test-resample.cpp @@ -57,7 +57,8 @@ static int usage(const char* name) { int main(int argc, char* argv[]) { const char* const progname = argv[0]; - bool profiling = false; + bool profileResample = false; + bool profileFilter = false; bool writeHeader = false; int channels = 1; int input_freq = 0; @@ -65,10 +66,13 @@ int main(int argc, char* argv[]) { AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY; int ch; - while ((ch = getopt(argc, argv, "phvsq:i:o:")) != -1) { + while ((ch = getopt(argc, argv, "pfhvsq:i:o:")) != -1) { switch (ch) { case 'p': - profiling = true; + profileResample = true; + break; + case 'f': + profileFilter = true; break; case 'h': writeHeader = true; @@ -230,27 +234,108 @@ int main(int argc, char* argv[]) { size_t output_size = 2 * 4 * ((int64_t) input_frames * output_freq) / input_freq; output_size &= ~7; // always stereo, 32-bits - void* output_vaddr = malloc(output_size); - AudioResampler* resampler = AudioResampler::create(16, channels, - output_freq, quality); - size_t out_frames = output_size/8; - resampler->setSampleRate(input_freq); - resampler->setVolume(0x1000, 0x1000); - - if (profiling) { - const int looplimit = 100; + if (profileFilter) { + // Check how fast sample rate changes are that require filter changes. + // The delta sample rate changes must indicate a downsampling ratio, + // and must be larger than 10% changes. + // + // On fast devices, filters should be generated between 0.1ms - 1ms. + // (single threaded). + AudioResampler* resampler = AudioResampler::create(16, channels, + 8000, quality); + int looplimit = 100; timespec start, end; clock_gettime(CLOCK_MONOTONIC, &start); for (int i = 0; i < looplimit; ++i) { - resampler->resample((int*) output_vaddr, out_frames, &provider); - provider.reset(); // reset only provider as benchmarking + resampler->setSampleRate(9000); + resampler->setSampleRate(12000); + resampler->setSampleRate(20000); + resampler->setSampleRate(30000); } clock_gettime(CLOCK_MONOTONIC, &end); int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec; int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec; int64_t time = end_ns - start_ns; - printf("time(ns):%lld channels:%d quality:%d\n", time, channels, quality); - printf("%f Mspl/s\n", out_frames * looplimit / (time / 1e9) / 1e6); + printf("%.2f sample rate changes with filter calculation/sec\n", + looplimit * 4 / (time / 1e9)); + + // Check how fast sample rate changes are without filter changes. + // This should be very fast, probably 0.1us - 1us per sample rate + // change. + resampler->setSampleRate(1000); + looplimit = 1000; + clock_gettime(CLOCK_MONOTONIC, &start); + for (int i = 0; i < looplimit; ++i) { + resampler->setSampleRate(1000+i); + } + clock_gettime(CLOCK_MONOTONIC, &end); + start_ns = start.tv_sec * 1000000000LL + start.tv_nsec; + end_ns = end.tv_sec * 1000000000LL + end.tv_nsec; + time = end_ns - start_ns; + printf("%.2f sample rate changes without filter calculation/sec\n", + looplimit / (time / 1e9)); + resampler->reset(); + delete resampler; + } + + void* output_vaddr = malloc(output_size); + AudioResampler* resampler = AudioResampler::create(16, channels, + output_freq, quality); + size_t out_frames = output_size/8; + + /* set volume precision to 12 bits, so the volume scale is 1<<12. + * This means the "integer" part fits in the Q19.12 precision + * representation of output int32_t. + * + * Generally 0 < volumePrecision <= 14 (due to the limits of + * int16_t values for Volume). volumePrecision cannot be 0 due + * to rounding and shifts. + */ + const int volumePrecision = 12; // in bits + + resampler->setSampleRate(input_freq); + resampler->setVolume(1 << volumePrecision, 1 << volumePrecision); + + if (profileResample) { + /* + * For profiling on mobile devices, upon experimentation + * it is better to run a few trials with a shorter loop limit, + * and take the minimum time. + * + * Long tests can cause CPU temperature to build up and thermal throttling + * to reduce CPU frequency. + * + * For frequency checks (index=0, or 1, etc.): + * "cat /sys/devices/system/cpu/cpu${index}/cpufreq/scaling_*_freq" + * + * For temperature checks (index=0, or 1, etc.): + * "cat /sys/class/thermal/thermal_zone${index}/temp" + * + * Another way to avoid thermal throttling is to fix the CPU frequency + * at a lower level which prevents excessive temperatures. + */ + const int trials = 4; + const int looplimit = 4; + timespec start, end; + int64_t time; + + for (int n = 0; n < trials; ++n) { + clock_gettime(CLOCK_MONOTONIC, &start); + for (int i = 0; i < looplimit; ++i) { + resampler->resample((int*) output_vaddr, out_frames, &provider); + provider.reset(); // during benchmarking reset only the provider + } + clock_gettime(CLOCK_MONOTONIC, &end); + int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec; + int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec; + int64_t diff_ns = end_ns - start_ns; + if (n == 0 || diff_ns < time) { + time = diff_ns; // save the best out of our trials. + } + } + // Mfrms/s is "Millions of output frames per second". + printf("quality: %d channels: %d msec: %lld Mfrms/s: %.2lf\n", + quality, channels, time/1000000, out_frames * looplimit / (time / 1e9) / 1e6); resampler->reset(); } @@ -266,19 +351,31 @@ int main(int argc, char* argv[]) { if (gVerbose) { printf("reset() complete\n"); } + delete resampler; + resampler = NULL; // mono takes left channel only // stereo right channel is half amplitude of stereo left channel (due to input creation) int32_t* out = (int32_t*) output_vaddr; int16_t* convert = (int16_t*) malloc(out_frames * channels * sizeof(int16_t)); + // round to half towards zero and saturate at int16 (non-dithered) + const int roundVal = (1<<(volumePrecision-1)) - 1; // volumePrecision > 0 + for (size_t i = 0; i < out_frames; i++) { for (int j = 0; j < channels; j++) { - int32_t s = out[i * 2 + j] >> 12; - if (s > 32767) - s = 32767; - else if (s < -32768) - s = -32768; + int32_t s = out[i * 2 + j] + roundVal; // add offset here + if (s < 0) { + s = (s + 1) >> volumePrecision; // round to 0 + if (s < -32768) { + s = -32768; + } + } else { + s = s >> volumePrecision; + if (s > 32767) { + s = 32767; + } + } convert[i * channels + j] = int16_t(s); } } |