Update test-resample to handle floating point
Change-Id: Ib34d716fbabcd5eb70f8a5ffcf362e242671d916
Signed-off-by: Andy Hung <hunga@google.com>
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index 6279665..d1de95d 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -27,18 +27,22 @@
#include <inttypes.h>
#include <time.h>
#include <math.h>
+#include <audio_utils/primitives.h>
#include <audio_utils/sndfile.h>
#include <utils/Vector.h>
using namespace android;
-bool gVerbose = false;
+static bool gVerbose = false;
static int usage(const char* 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] [-O csv] [-P csv] [<input-file>]"
+ fprintf(stderr,"Usage: %s [-p] [-f] [-F] [-h] [-v] [-s] [-q {dq|lq|mq|hq|vhq|dlq|dmq|dhq}]"
+ " [-i input-sample-rate] [-o output-sample-rate]"
+ " [-O csv] [-P csv] [<input-file>]"
" <output-file>\n", name);
fprintf(stderr," -p enable profiling\n");
+ fprintf(stderr," -f enable filter profiling\n");
+ fprintf(stderr," -F enable floating point -q {dlq|dmq|dhq} only");
fprintf(stderr," -h create wav file\n");
fprintf(stderr," -v verbose : log buffer provider calls\n");
fprintf(stderr," -s stereo (ignored if input file is specified)\n");
@@ -103,6 +107,7 @@
bool profileResample = false;
bool profileFilter = false;
bool writeHeader = false;
+ bool useFloat = false;
int channels = 1;
int input_freq = 0;
int output_freq = 0;
@@ -111,7 +116,7 @@
Vector<int> Pvalues;
int ch;
- while ((ch = getopt(argc, argv, "pfhvsq:i:o:O:P:")) != -1) {
+ while ((ch = getopt(argc, argv, "pfFhvsq:i:o:O:P:")) != -1) {
switch (ch) {
case 'p':
profileResample = true;
@@ -119,6 +124,9 @@
case 'f':
profileFilter = true;
break;
+ case 'F':
+ useFloat = true;
+ break;
case 'h':
writeHeader = true;
break;
@@ -174,6 +182,12 @@
return -1;
}
}
+
+ if (useFloat && quality < AudioResampler::DYN_LOW_QUALITY) {
+ fprintf(stderr, "float processing is only possible for dynamic resamplers\n");
+ return -1;
+ }
+
argc -= optind;
argv += optind;
@@ -225,22 +239,37 @@
}
}
}
+ size_t frame_size = channels * sizeof(int16_t);
+ size_t input_frames = input_size / frame_size;
+
+ // For float processing, convert input int16_t to float array
+ if (useFloat) {
+ void *new_vaddr;
+
+ frame_size = channels * sizeof(float);
+ input_size = input_frames * frame_size;
+ new_vaddr = malloc(input_size);
+ memcpy_to_float_from_i16(reinterpret_cast<float*>(new_vaddr),
+ reinterpret_cast<int16_t*>(input_vaddr), input_frames * channels);
+ free(input_vaddr);
+ input_vaddr = new_vaddr;
+ }
// ----------------------------------------------------------
class Provider: public AudioBufferProvider {
- int16_t* const mAddr; // base address
+ const void* mAddr; // base address
const size_t mNumFrames; // total frames
- const int mChannels;
+ const size_t mFrameSize; // size of each frame in bytes
size_t mNextFrame; // index of next frame to provide
size_t mUnrel; // number of frames not yet released
const Vector<int> mPvalues; // number of frames provided per call
size_t mNextPidx; // index of next entry in mPvalues to use
public:
- Provider(const void* addr, size_t size, int channels, const Vector<int>& Pvalues)
- : mAddr((int16_t*) addr),
- mNumFrames(size / (channels*sizeof(int16_t))),
- mChannels(channels),
+ Provider(const void* addr, size_t frames, size_t frameSize, const Vector<int>& Pvalues)
+ : mAddr(addr),
+ mNumFrames(frames),
+ mFrameSize(frameSize),
mNextFrame(0), mUnrel(0), mPvalues(Pvalues), mNextPidx(0) {
}
virtual status_t getNextBuffer(Buffer* buffer,
@@ -267,10 +296,10 @@
}
mUnrel = buffer->frameCount;
if (buffer->frameCount > 0) {
- buffer->i16 = &mAddr[mChannels * mNextFrame];
+ buffer->raw = (char *)mAddr + mFrameSize * mNextFrame;
return NO_ERROR;
} else {
- buffer->i16 = NULL;
+ buffer->raw = NULL;
return NOT_ENOUGH_DATA;
}
}
@@ -289,17 +318,18 @@
mUnrel -= buffer->frameCount;
}
buffer->frameCount = 0;
- buffer->i16 = NULL;
+ buffer->raw = NULL;
}
void reset() {
mNextFrame = 0;
}
- } provider(input_vaddr, input_size, channels, Pvalues);
+ } provider(input_vaddr, input_frames, frame_size, Pvalues);
- size_t input_frames = input_size / (channels * sizeof(int16_t));
if (gVerbose) {
printf("%zu input frames\n", input_frames);
}
+
+ int bit_depth = useFloat ? 32 : 16;
size_t output_size = 2 * 4 * ((int64_t) input_frames * output_freq) / input_freq;
output_size &= ~7; // always stereo, 32-bits
@@ -310,7 +340,7 @@
//
// On fast devices, filters should be generated between 0.1ms - 1ms.
// (single threaded).
- AudioResampler* resampler = AudioResampler::create(16, channels,
+ AudioResampler* resampler = AudioResampler::create(bit_depth, channels,
8000, quality);
int looplimit = 100;
timespec start, end;
@@ -348,7 +378,7 @@
}
void* output_vaddr = malloc(output_size);
- AudioResampler* resampler = AudioResampler::create(16, channels,
+ AudioResampler* resampler = AudioResampler::create(bit_depth, channels,
output_freq, quality);
size_t out_frames = output_size/8;
@@ -437,6 +467,13 @@
delete resampler;
resampler = NULL;
+ // For float processing, convert output format from float to Q4.27,
+ // which is then converted to int16_t for final storage.
+ if (useFloat) {
+ memcpy_to_q4_27_from_float(reinterpret_cast<int32_t*>(output_vaddr),
+ reinterpret_cast<float*>(output_vaddr), out_frames * 2); // stereo samples
+ }
+
// 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;