Update test-resample to handle floating point (78136683) · Commits · e / os / android_frameworks_av

services/audioflinger/test-resample.cpp

+54 −17

Original line number	Diff line number	Diff line
		@@ -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 @@ int main(int argc, char* argv[]) {
		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 @@ int main(int argc, char* argv[]) {
		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 @@ int main(int argc, char* argv[]) {
		case 'f':
		profileFilter = true;
		break;
		case 'F':
		useFloat = true;
		break;
		case 'h':
		writeHeader = true;
		break;
		@@ -174,6 +182,12 @@ int main(int argc, char* argv[]) {
		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 @@ int main(int argc, char* argv[]) {
		}
		}
		}
		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 @@ int main(int argc, char* argv[]) {
		}
		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 @@ int main(int argc, char* argv[]) {
		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 @@ int main(int argc, char* argv[]) {
		//
		// 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 @@ int main(int argc, char* argv[]) {
		}

		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 @@ int main(int argc, char* argv[]) {
		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;