services/audioflinger/test-resample.cpp - platform/frameworks/av - Git at Google

 /*
  * Copyright (C) 2012 The Android Open Source Project
  *
  * Licensed under the Apache License, Version 2.0 (the "License");
  * you may not use this file except in compliance with the License.
  * You may obtain a copy of the License at
  *
  *      http://www.apache.org/licenses/LICENSE-2.0
  *
  * Unless required by applicable law or agreed to in writing, software
  * distributed under the License is distributed on an "AS IS" BASIS,
  * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
  * See the License for the specific language governing permissions and
  * limitations under the License.
  */

 #include "AudioResampler.h"
 #include <media/AudioBufferProvider.h>
 #include <unistd.h>
 #include <stdio.h>
 #include <stdlib.h>
 #include <fcntl.h>
 #include <string.h>
 #include <sys/mman.h>
 #include <sys/stat.h>
 #include <errno.h>
 #include <time.h>
 #include <math.h>

 using namespace android;

 struct HeaderWav {
     HeaderWav(size_t size, int nc, int sr, int bits) {
         strncpy(RIFF, "RIFF", 4);
         chunkSize = size + sizeof(HeaderWav);
         strncpy(WAVE, "WAVE", 4);
         strncpy(fmt,  "fmt ", 4);
         fmtSize = 16;
         audioFormat = 1;
         numChannels = nc;
         samplesRate = sr;
         byteRate = sr * numChannels * (bits/8);
         align = nc*(bits/8);
         bitsPerSample = bits;
         strncpy(data, "data", 4);
         dataSize = size;
     }

     char RIFF[4];           // RIFF
     uint32_t chunkSize;     // File size
     char WAVE[4];        // WAVE
     char fmt[4];            // fmt\0
     uint32_t fmtSize;       // fmt size
     uint16_t audioFormat;   // 1=PCM
     uint16_t numChannels;   // num channels
     uint32_t samplesRate;   // sample rate in hz
     uint32_t byteRate;      // Bps
     uint16_t align;         // 2=16-bit mono, 4=16-bit stereo
     uint16_t bitsPerSample; // bits per sample
     char data[4];           // "data"
     uint32_t dataSize;      // size
 };

 static int usage(const char* name) {
     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, "phsq:i:o:")) != -1) {
         switch (ch) {
         case 'p':
             profiling = true;
             break;
         case 'h':
             writeHeader = true;
             break;
         case 's':
             channels = 2;
             break;
         case 'q':
             if (!strcmp(optarg, "dq"))
                 quality = AudioResampler::DEFAULT_QUALITY;
             else if (!strcmp(optarg, "lq"))
                 quality = AudioResampler::LOW_QUALITY;
             else if (!strcmp(optarg, "mq"))
                 quality = AudioResampler::MED_QUALITY;
             else if (!strcmp(optarg, "hq"))
                 quality = AudioResampler::HIGH_QUALITY;
             else if (!strcmp(optarg, "vhq"))
                 quality = AudioResampler::VERY_HIGH_QUALITY;
             else {
                 usage(progname);
                 return -1;
             }
             break;
         case 'i':
             input_freq = atoi(optarg);
             break;
         case 'o':
             output_freq = atoi(optarg);
             break;
         case '?':
         default:
             usage(progname);
             return -1;
         }
     }
     argc -= optind;
     argv += optind;

     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;
     }

     // ----------------------------------------------------------

     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(file_in, O_RDONLY);
         if (input_fd < 0) {
             fprintf(stderr, "open: %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<(size_t)channels ; j++) {
                 in[i*channels + j] = yi / (1+j);
             }
         }
     }

     // ----------------------------------------------------------

     class Provider: public AudioBufferProvider {
         int16_t* mAddr;
         size_t mNumFrames;
     public:
         Provider(const void* addr, size_t size, int channels) {
             mAddr = (int16_t*) addr;
             mNumFrames = size / (channels*sizeof(int16_t));
         }
         virtual status_t getNextBuffer(Buffer* buffer,
                 int64_t pts = kInvalidPTS) {
             buffer->frameCount = mNumFrames;
             buffer->i16 = mAddr;
             return NO_ERROR;
         }
         virtual void releaseBuffer(Buffer* buffer) {
         }
     } provider(input_vaddr, input_size, channels);

     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);

     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)/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 * channels * sizeof(int16_t));
     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;
             convert[i * channels + j] = int16_t(s);
         }
     }

     // write output to disk
     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));
         return -1;
     }

     if (writeHeader) {
         HeaderWav wav(out_frames * channels * sizeof(int16_t), channels, output_freq, 16);
         write(output_fd, &wav, sizeof(wav));
     }

     write(output_fd, convert, out_frames * channels * sizeof(int16_t));
     close(output_fd);

     return 0;
 }
	/*
	* Copyright (C) 2012 The Android Open Source Project
	*
	* Licensed under the Apache License, Version 2.0 (the "License");
	* you may not use this file except in compliance with the License.
	* You may obtain a copy of the License at
	*
	* http://www.apache.org/licenses/LICENSE-2.0
	*
	* Unless required by applicable law or agreed to in writing, software
	* distributed under the License is distributed on an "AS IS" BASIS,
	* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
	* See the License for the specific language governing permissions and
	* limitations under the License.
	*/

	#include "AudioResampler.h"
	#include <media/AudioBufferProvider.h>
	#include <unistd.h>
	#include <stdio.h>
	#include <stdlib.h>
	#include <fcntl.h>
	#include <string.h>
	#include <sys/mman.h>
	#include <sys/stat.h>
	#include <errno.h>
	#include <time.h>
	#include <math.h>

	using namespace android;

	struct HeaderWav {
	HeaderWav(size_t size, int nc, int sr, int bits) {
	strncpy(RIFF, "RIFF", 4);
	chunkSize = size + sizeof(HeaderWav);
	strncpy(WAVE, "WAVE", 4);
	strncpy(fmt, "fmt ", 4);
	fmtSize = 16;
	audioFormat = 1;
	numChannels = nc;
	samplesRate = sr;
	byteRate = sr * numChannels * (bits/8);
	align = nc*(bits/8);
	bitsPerSample = bits;
	strncpy(data, "data", 4);
	dataSize = size;
	}

	char RIFF[4]; // RIFF
	uint32_t chunkSize; // File size
	char WAVE[4]; // WAVE
	char fmt[4]; // fmt\0
	uint32_t fmtSize; // fmt size
	uint16_t audioFormat; // 1=PCM
	uint16_t numChannels; // num channels
	uint32_t samplesRate; // sample rate in hz
	uint32_t byteRate; // Bps
	uint16_t align; // 2=16-bit mono, 4=16-bit stereo
	uint16_t bitsPerSample; // bits per sample
	char data[4]; // "data"
	uint32_t dataSize; // size
	};

	static int usage(const char* name) {
	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, "phsq:i:o:")) != -1) {
	switch (ch) {
	case 'p':
	profiling = true;
	break;
	case 'h':
	writeHeader = true;
	break;
	case 's':
	channels = 2;
	break;
	case 'q':
	if (!strcmp(optarg, "dq"))
	quality = AudioResampler::DEFAULT_QUALITY;
	else if (!strcmp(optarg, "lq"))
	quality = AudioResampler::LOW_QUALITY;
	else if (!strcmp(optarg, "mq"))
	quality = AudioResampler::MED_QUALITY;
	else if (!strcmp(optarg, "hq"))
	quality = AudioResampler::HIGH_QUALITY;
	else if (!strcmp(optarg, "vhq"))
	quality = AudioResampler::VERY_HIGH_QUALITY;
	else {
	usage(progname);
	return -1;
	}
	break;
	case 'i':
	input_freq = atoi(optarg);
	break;
	case 'o':
	output_freq = atoi(optarg);
	break;
	case '?':
	default:
	usage(progname);
	return -1;
	}
	}
	argc -= optind;
	argv += optind;

	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;
	}

	// ----------------------------------------------------------

	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(file_in, O_RDONLY);
	if (input_fd < 0) {
	fprintf(stderr, "open: %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<(size_t)channels ; j++) {
	in[i*channels + j] = yi / (1+j);
	}
	}
	}

	// ----------------------------------------------------------

	class Provider: public AudioBufferProvider {
	int16_t* mAddr;
	size_t mNumFrames;
	public:
	Provider(const void* addr, size_t size, int channels) {
	mAddr = (int16_t*) addr;
	mNumFrames = size / (channels*sizeof(int16_t));
	}
	virtual status_t getNextBuffer(Buffer* buffer,
	int64_t pts = kInvalidPTS) {
	buffer->frameCount = mNumFrames;
	buffer->i16 = mAddr;
	return NO_ERROR;
	}
	virtual void releaseBuffer(Buffer* buffer) {
	}
	} provider(input_vaddr, input_size, channels);

	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);

	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)/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 channels * sizeof(int16_t));
	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;
	convert[i * channels + j] = int16_t(s);
	}
	}

	// write output to disk
	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));
	return -1;
	}

	if (writeHeader) {
	HeaderWav wav(out_frames * channels * sizeof(int16_t), channels, output_freq, 16);
	write(output_fd, &wav, sizeof(wav));
	}

	write(output_fd, convert, out_frames * channels * sizeof(int16_t));
	close(output_fd);

	return 0;
	}