improve resample test
- handle stereo input
- input file can now be ommited, in this case
a linear chirp will be used automatically
- better usage information
Change-Id: I5d62a6c26a9054a1c1a517a065b4df5a2cdcda22
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index 151313b..e6d5cbe 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -25,6 +25,7 @@
#include <sys/stat.h>
#include <errno.h>
#include <time.h>
+#include <math.h>
using namespace android;
@@ -61,31 +62,34 @@
};
static int usage(const char* name) {
- fprintf(stderr,"Usage: %s [-p] [-h] [-q <dq|lq|mq|hq|vhq>] [-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,"-q - resampler quality\n");
- fprintf(stderr," dq : default quality\n");
- fprintf(stderr," lq : low quality\n");
- fprintf(stderr," mq : medium quality\n");
- fprintf(stderr," hq : high quality\n");
- fprintf(stderr," vhq : very high quality\n");
- fprintf(stderr,"-i - input file sample rate\n");
- fprintf(stderr,"-o - output file sample rate\n");
+ fprintf(stderr,"Usage: %s [-p] [-h] [-s] [-q {dq|lq|mq|hq|vhq}] [-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," -s stereo\n");
+ fprintf(stderr," -q resampler quality\n");
+ fprintf(stderr," dq : default quality\n");
+ fprintf(stderr," lq : low quality\n");
+ fprintf(stderr," mq : medium quality\n");
+ fprintf(stderr," hq : high quality\n");
+ fprintf(stderr," vhq : very high quality\n");
+ fprintf(stderr," -i input file sample rate\n");
+ fprintf(stderr," -o output file sample rate\n");
return -1;
}
int main(int argc, char* argv[]) {
+ const char* const progname = argv[0];
bool profiling = false;
bool writeHeader = false;
+ int channels = 1;
int input_freq = 0;
int output_freq = 0;
AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY;
int ch;
- while ((ch = getopt(argc, argv, "phq:i:o:")) != -1) {
+ while ((ch = getopt(argc, argv, "phsq:i:o:")) != -1) {
switch (ch) {
case 'p':
profiling = true;
@@ -93,6 +97,9 @@
case 'h':
writeHeader = true;
break;
+ case 's':
+ channels = 2;
+ break;
case 'q':
if (!strcmp(optarg, "dq"))
quality = AudioResampler::DEFAULT_QUALITY;
@@ -105,7 +112,7 @@
else if (!strcmp(optarg, "vhq"))
quality = AudioResampler::VERY_HIGH_QUALITY;
else {
- usage(argv[0]);
+ usage(progname);
return -1;
}
break;
@@ -117,54 +124,74 @@
break;
case '?':
default:
- usage(argv[0]);
+ usage(progname);
return -1;
}
}
argc -= optind;
+ argv += optind;
- if (argc != 2) {
- usage(argv[0]);
+ const char* file_in = NULL;
+ const char* file_out = NULL;
+ if (argc == 1) {
+ file_out = argv[0];
+ } else if (argc == 2) {
+ file_in = argv[0];
+ file_out = argv[1];
+ } else {
+ usage(progname);
return -1;
}
- argv += optind;
-
// ----------------------------------------------------------
- struct stat st;
- if (stat(argv[0], &st) < 0) {
- fprintf(stderr, "stat: %s\n", strerror(errno));
- return -1;
- }
+ size_t input_size;
+ void* input_vaddr;
+ if (argc == 2) {
+ struct stat st;
+ if (stat(file_in, &st) < 0) {
+ fprintf(stderr, "stat: %s\n", strerror(errno));
+ return -1;
+ }
- int input_fd = open(argv[0], O_RDONLY);
- if (input_fd < 0) {
- fprintf(stderr, "open: %s\n", strerror(errno));
- return -1;
- }
+ int input_fd = open(file_in, O_RDONLY);
+ if (input_fd < 0) {
+ fprintf(stderr, "open: %s\n", strerror(errno));
+ return -1;
+ }
- size_t input_size = st.st_size;
- void* input_vaddr = mmap(0, input_size, PROT_READ, MAP_PRIVATE, input_fd,
- 0);
- if (input_vaddr == MAP_FAILED ) {
- fprintf(stderr, "mmap: %s\n", strerror(errno));
- return -1;
+ input_size = st.st_size;
+ input_vaddr = mmap(0, input_size, PROT_READ, MAP_PRIVATE, input_fd, 0);
+ if (input_vaddr == MAP_FAILED ) {
+ fprintf(stderr, "mmap: %s\n", strerror(errno));
+ return -1;
+ }
+ } else {
+ double k = 1000; // Hz / s
+ double time = (input_freq / 2) / k;
+ size_t input_frames = size_t(input_freq * time);
+ input_size = channels * sizeof(int16_t) * input_frames;
+ input_vaddr = malloc(input_size);
+ int16_t* in = (int16_t*)input_vaddr;
+ for (size_t i=0 ; i<input_frames ; i++) {
+ double t = double(i) / input_freq;
+ double y = sin(M_PI * k * t * t);
+ int16_t yi = floor(y * 32767.0 + 0.5);
+ for (size_t j=0 ; j<channels ; j++) {
+ in[i*channels + j] = yi;
+ }
+ }
}
-// printf("input sample rate: %d Hz\n", input_freq);
-// printf("output sample rate: %d Hz\n", output_freq);
-// printf("input mmap: %p, size=%u\n", input_vaddr, input_size);
-
// ----------------------------------------------------------
class Provider: public AudioBufferProvider {
int16_t* mAddr;
size_t mNumFrames;
public:
- Provider(const void* addr, size_t size) {
+ Provider(const void* addr, size_t size, int channels) {
mAddr = (int16_t*) addr;
- mNumFrames = size / sizeof(int16_t);
+ mNumFrames = size / (channels*sizeof(int16_t));
}
virtual status_t getNextBuffer(Buffer* buffer,
int64_t pts = kInvalidPTS) {
@@ -174,47 +201,61 @@
}
virtual void releaseBuffer(Buffer* buffer) {
}
- } provider(input_vaddr, input_size);
+ } provider(input_vaddr, input_size, channels);
- size_t output_size = 2 * 2 * ((int64_t) input_size * output_freq)
- / input_freq;
+ size_t input_frames = input_size / (channels * sizeof(int16_t));
+ 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);
- memset(output_vaddr, 0, output_size);
-
- AudioResampler* resampler = AudioResampler::create(16, 1, output_freq,
- quality);
-
- size_t out_frames = output_size/8;
- resampler->setSampleRate(input_freq);
- resampler->setVolume(0x1000, 0x1000);
- resampler->resample((int*) output_vaddr, out_frames, &provider);
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_HR, &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_HR, &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;
+ 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);
+
+ memset(output_vaddr, 0, output_size);
+ resampler->resample((int*) output_vaddr, out_frames, &provider);
+
// down-mix (we just truncate and keep the left channel)
int32_t* out = (int32_t*) output_vaddr;
- int16_t* convert = (int16_t*) malloc(out_frames * sizeof(int16_t));
+ int16_t* convert = (int16_t*) malloc(out_frames * channels * sizeof(int16_t));
for (size_t i = 0; i < out_frames; i++) {
- int32_t s = out[i * 2] >> 12;
- if (s > 32767) s = 32767;
- else if (s < -32768) s = -32768;
- convert[i] = int16_t(s);
+ 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;
+ convert[i * channels + j] = int16_t(s);
+ }
}
// write output to disk
- int output_fd = open(argv[1], O_WRONLY | O_CREAT | O_TRUNC,
+ int output_fd = open(file_out, O_WRONLY | O_CREAT | O_TRUNC,
S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH);
if (output_fd < 0) {
fprintf(stderr, "open: %s\n", strerror(errno));
@@ -222,11 +263,11 @@
}
if (writeHeader) {
- HeaderWav wav(out_frames*sizeof(int16_t), 1, output_freq, 16);
+ HeaderWav wav(out_frames * channels * sizeof(int16_t), channels, output_freq, 16);
write(output_fd, &wav, sizeof(wav));
}
- write(output_fd, convert, out_frames * sizeof(int16_t));
+ write(output_fd, convert, out_frames * channels * sizeof(int16_t));
close(output_fd);
return 0;