Audio resampler update to add S16 filters
This does not affect the existing resamplers.
New resampler accessed through additional quality settings:
DYN_LOW_QUALITY = 5
DYN_MED_QUALITY = 6
DYN_HIGH_QUALITY = 7
Change-Id: Iebbd31871e808a4a6dee3f3abfd7e9dcf77c48e1
Signed-off-by: Andy Hung <hunga@google.com>
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index 403bb6d..3f2ce55 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -33,8 +33,9 @@
bool gVerbose = false;
static int usage(const char* name) {
- fprintf(stderr,"Usage: %s [-p] [-h] [-v] [-s] [-q {dq|lq|mq|hq|vhq}] [-i input-sample-rate] "
- "[-o output-sample-rate] [<input-file>] <output-file>\n", name);
+ fprintf(stderr,"Usage: %s [-p] [-h] [-v] [-s] [-q {dq|lq|mq|hq|vhq|dlq|dmq|dhq}]"
+ " [-i input-sample-rate] [-o output-sample-rate] [<input-file>]"
+ " <output-file>\n", name);
fprintf(stderr," -p enable profiling\n");
fprintf(stderr," -h create wav file\n");
fprintf(stderr," -v verbose : log buffer provider calls\n");
@@ -45,6 +46,9 @@
fprintf(stderr," mq : medium quality\n");
fprintf(stderr," hq : high quality\n");
fprintf(stderr," vhq : very high quality\n");
+ fprintf(stderr," dlq : dynamic low quality\n");
+ fprintf(stderr," dmq : dynamic medium quality\n");
+ fprintf(stderr," dhq : dynamic high quality\n");
fprintf(stderr," -i input file sample rate (ignored if input file is specified)\n");
fprintf(stderr," -o output file sample rate\n");
return -1;
@@ -86,6 +90,12 @@
quality = AudioResampler::HIGH_QUALITY;
else if (!strcmp(optarg, "vhq"))
quality = AudioResampler::VERY_HIGH_QUALITY;
+ else if (!strcmp(optarg, "dlq"))
+ quality = AudioResampler::DYN_LOW_QUALITY;
+ else if (!strcmp(optarg, "dmq"))
+ quality = AudioResampler::DYN_MED_QUALITY;
+ else if (!strcmp(optarg, "dhq"))
+ quality = AudioResampler::DYN_HIGH_QUALITY;
else {
usage(progname);
return -1;
@@ -137,6 +147,8 @@
channels = info.channels;
input_freq = info.samplerate;
} else {
+ // data for testing is exactly (input sampling rate/1000)/2 seconds
+ // so 44.1khz input is 22.05 seconds
double k = 1000; // Hz / s
double time = (input_freq / 2) / k;
size_t input_frames = size_t(input_freq * time);
@@ -148,7 +160,7 @@
double y = sin(M_PI * k * t * t);
int16_t yi = floor(y * 32767.0 + 0.5);
for (size_t j=0 ; j<(size_t)channels ; j++) {
- in[i*channels + j] = yi / (1+j);
+ in[i*channels + j] = yi / (1+j); // right ch. 1/2 left ch.
}
}
}
@@ -170,6 +182,7 @@
}
virtual status_t getNextBuffer(Buffer* buffer,
int64_t pts = kInvalidPTS) {
+ (void)pts; // suppress warning
size_t requestedFrames = buffer->frameCount;
if (requestedFrames > mNumFrames - mNextFrame) {
buffer->frameCount = mNumFrames - mNextFrame;
@@ -205,6 +218,9 @@
buffer->frameCount = 0;
buffer->i16 = NULL;
}
+ void reset() {
+ mNextFrame = 0;
+ }
} provider(input_vaddr, input_size, channels);
size_t input_frames = input_size / (channels * sizeof(int16_t));
@@ -215,37 +231,29 @@
output_size &= ~7; // always stereo, 32-bits
void* output_vaddr = malloc(output_size);
-
- if (profiling) {
- AudioResampler* resampler = AudioResampler::create(16, channels,
- output_freq, quality);
-
- size_t out_frames = output_size/8;
- resampler->setSampleRate(input_freq);
- resampler->setVolume(0x1000, 0x1000);
-
- memset(output_vaddr, 0, output_size);
- timespec start, end;
- clock_gettime(CLOCK_MONOTONIC, &start);
- resampler->resample((int*) output_vaddr, out_frames, &provider);
- resampler->resample((int*) output_vaddr, out_frames, &provider);
- resampler->resample((int*) output_vaddr, out_frames, &provider);
- resampler->resample((int*) output_vaddr, out_frames, &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 time = (end_ns - start_ns)/4;
- printf("%f Mspl/s\n", out_frames/(time/1e9)/1e6);
-
- delete resampler;
- }
-
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;
+ 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
+ }
+ 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);
+ resampler->reset();
+ }
+
memset(output_vaddr, 0, output_size);
if (gVerbose) {
printf("resample() %u output frames\n", out_frames);
@@ -259,14 +267,18 @@
printf("reset() complete\n");
}
- // down-mix (we just truncate and keep the left channel)
+ // 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));
+
for (size_t i = 0; i < out_frames; i++) {
- for (int j=0 ; j<channels ; j++) {
+ 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;
+ if (s > 32767)
+ s = 32767;
+ else if (s < -32768)
+ s = -32768;
convert[i * channels + j] = int16_t(s);
}
}