Objective Your task is to read a data file with sound data and scale the amplitude and then create a.WAV sound file with the slower audio. WAV files are a Microsoft sound file that stores data samples as raw data - in other words, there is no compression of the data. I have provided you with a test datafile that has 16-bit samples recorded at 22050 samples/second. 65535. This anows Provided Code I have provided you with the following skeleton main function int main(int argc, char **argv) { if (argc = 4) { } printf("%s datafile wavfile scale\n", argv[0]); exit(-1); char datafile = argv[1]; char wavfile = argv[2]; double scale atof (argv[3]); = int16_t* sounddata; // TODO allocate space for 1000000 int16_t's // TODO open the datafile and read the data into sounddata // TODO iterate through the array and scale the values wavdata_t wav; // TODO fill in the wav struct write_wav (wavfile, &wav); Writing the code The code takes in three arguments - an input datafile, the name of a .WAV file to write the data with the scaled signal, and a scale value. You will need to read all the data from datafile into an array. You can assume that the file is no bigger than 1000000 sound samples long- that means you should allocate an sounddata array that is big enough 1000000 sound samples where each sample is a int16_t. You should then read the entire file into sounddata with one read of 1000000 int16_t's. However, the read may return less bytes than that. Keep track of the number of bytes that the read returns, since you can use to calculate how many samples you read into the array - each sample is 2 bytes long (the size of an int16_t type). In this case, it will be easier to use open and read instead of fopen and fread. Once you figure out the number of samples in sounddata, you can iterate through the samples in the sounddata array and multiply the values by scale. You just have to make sure that the resulting values doesn't exceed 32767 or go below -32768 which are the bounds of an int16_t type. I have provided you with a write_wav function in wav.c that helps write the data to a .WAV file. However, before you call write_wav, you need to fill in the way data structure. The wavdata_t struct is defined as follows: typedef struct { uint16_t nchannels; uint16 t bits_per_sample; uint32_t sample_rate; uint32_t datasize; int16_t *data; } wavdata_t; For the provided testfile, nchannels=1, bits_per_sample=16, and sample_rate=22050. datasize is the number of bytes in the data set, which is equal to the number of bytes you read from the datafile. data is a pointer to an array that holds the data samples. Once way data structure is filled in, you can call write_wav. Testing the code I have included an expected.5.wavand an expected2.wav file that contains the expected output for the following runs of the code. You can compare your output with the expected output using the diff command as shown below. If the files are exactly the same, then the diff command will not show any differences. $ gcc -o hw5 hw5.c wav.c $ ./hw5 datafile my.wav 2 $ diff my.wav expected2.wav $ ./hw5 datafile my.wav 0.5 $ diff my.wav expected.5.wav You should be able to play your output .WAV file and hear the change in volume in the audio. IMPORTANT: make sure that you get no differences with your output.
Objective Your task is to read a data file with sound data and scale the amplitude and then create a.WAV sound file with the slower audio. WAV files are a Microsoft sound file that stores data samples as raw data - in other words, there is no compression of the data. I have provided you with a test datafile that has 16-bit samples recorded at 22050 samples/second. 65535. This anows Provided Code I have provided you with the following skeleton main function int main(int argc, char **argv) { if (argc = 4) { } printf("%s datafile wavfile scale\n", argv[0]); exit(-1); char datafile = argv[1]; char wavfile = argv[2]; double scale atof (argv[3]); = int16_t* sounddata; // TODO allocate space for 1000000 int16_t's // TODO open the datafile and read the data into sounddata // TODO iterate through the array and scale the values wavdata_t wav; // TODO fill in the wav struct write_wav (wavfile, &wav); Writing the code The code takes in three arguments - an input datafile, the name of a .WAV file to write the data with the scaled signal, and a scale value. You will need to read all the data from datafile into an array. You can assume that the file is no bigger than 1000000 sound samples long- that means you should allocate an sounddata array that is big enough 1000000 sound samples where each sample is a int16_t. You should then read the entire file into sounddata with one read of 1000000 int16_t's. However, the read may return less bytes than that. Keep track of the number of bytes that the read returns, since you can use to calculate how many samples you read into the array - each sample is 2 bytes long (the size of an int16_t type). In this case, it will be easier to use open and read instead of fopen and fread. Once you figure out the number of samples in sounddata, you can iterate through the samples in the sounddata array and multiply the values by scale. You just have to make sure that the resulting values doesn't exceed 32767 or go below -32768 which are the bounds of an int16_t type. I have provided you with a write_wav function in wav.c that helps write the data to a .WAV file. However, before you call write_wav, you need to fill in the way data structure. The wavdata_t struct is defined as follows: typedef struct { uint16_t nchannels; uint16 t bits_per_sample; uint32_t sample_rate; uint32_t datasize; int16_t *data; } wavdata_t; For the provided testfile, nchannels=1, bits_per_sample=16, and sample_rate=22050. datasize is the number of bytes in the data set, which is equal to the number of bytes you read from the datafile. data is a pointer to an array that holds the data samples. Once way data structure is filled in, you can call write_wav. Testing the code I have included an expected.5.wavand an expected2.wav file that contains the expected output for the following runs of the code. You can compare your output with the expected output using the diff command as shown below. If the files are exactly the same, then the diff command will not show any differences. $ gcc -o hw5 hw5.c wav.c $ ./hw5 datafile my.wav 2 $ diff my.wav expected2.wav $ ./hw5 datafile my.wav 0.5 $ diff my.wav expected.5.wav You should be able to play your output .WAV file and hear the change in volume in the audio. IMPORTANT: make sure that you get no differences with your output.
Chapter13: File Input And Output
Section: Chapter Questions
Problem 6PE
Related questions
Question
wave.c:
#include <stdio.h>
#include <fcntl.h>
#include <unistd.h>
#include <stdlib.h>
#include <math.h>
#include "wav.h"
int write_wav(char *wavfile, wavdata_t *wav)
{
int fd = open(wavfile, O_WRONLY | O_CREAT | O_TRUNC, 0644);
if (fd == -1) {
printf("Could not open file %s\n", wavfile);
return -1;
}
write(fd, "RIFF", 4);
uint32_t filesize = wav->datasize + 36;
uint8_t data[20];
data[0] = filesize & 0xff;
data[1] = (filesize>>8) & 0xff;
data[2] = (filesize>>16) & 0xff;
data[3] = (filesize>>24) & 0xff;
write(fd, data, 4);
write(fd, "WAVEfmt ", 8);
data[0] = 16;
data[1] = 0;
data[2] = 0;
data[3] = 0;
data[4] = 1;
data[5] = 0;
data[6] = wav->nchannels & 0xff;
data[7] = wav->nchannels >> 8;
data[8] = wav->sample_rate & 0xff;
data[9] = (wav->sample_rate>>8) & 0xff;
data[10] = (wav->sample_rate>>16) & 0xff;
data[11] = (wav->sample_rate>>24) & 0xff;
uint32_t bytes_per_second =
wav->sample_rate * wav->nchannels * wav->bits_per_sample/8;
data[12] = bytes_per_second & 0xff;
data[13] = (bytes_per_second>>8) & 0xff;
data[14] = (bytes_per_second>>16) & 0xff;
data[15] = (bytes_per_second>>24) & 0xff;
uint16_t bytes_per_sample = wav->nchannels * wav->bits_per_sample/8;
data[16] = bytes_per_sample & 0xff;
data[17] = (bytes_per_sample>>8) & 0xff;
data[18] = wav->bits_per_sample & 0xff;
data[19] = wav->bits_per_sample >> 8;
write(fd, data, 20);
write(fd, "data", 4);
data[0] = wav->datasize & 0xff;
data[1] = (wav->datasize>>8) & 0xff;
data[2] = (wav->datasize>>16) & 0xff;
data[3] = (wav->datasize>>24) & 0xff;
write(fd, data, 4);
write(fd, wav->data, wav->datasize);
close(fd);
return 0;
}
wave.h:
#include <stdint.h>
typedef struct {
uint16_t nchannels;
uint16_t bits_per_sample;
uint32_t sample_rate;
uint32_t datasize;
int16_t* data;
} wavdata_t;
int read_wav(char *wavfile, wavdata_t *);
int write_wav(char *wavfile, wavdata_t *);
typedef struct {
double r;
double i;
} complex_t;
void dft(int16_t *x, complex_t *X, int N);
![Objective
Your task is to read a data file with sound data and scale the amplitude and then create a.WAV sound file
with the slower audio. WAV files are a Microsoft sound file that stores data samples as raw data - in other
words, there is no compression of the data. I have provided you with a test datafile that has 16-bit samples
recorded at 22050 samples/second.
65535. This anows
Provided Code
I have provided you with the following skeleton main function
int main(int argc, char **argv)
{
if (argc = 4) {
}
printf("%s datafile wavfile scale\n", argv[0]);
exit(-1);
char datafile = argv[1];
char
wavfile = argv[2];
double scale atof (argv[3]);
=
int16_t* sounddata;
// TODO allocate space for 1000000 int16_t's
// TODO open the datafile and read the data into sounddata
// TODO iterate through the array and scale the values
wavdata_t wav;
// TODO fill in the wav struct
write_wav (wavfile, &wav);
Writing the code
The code takes in three arguments - an input datafile, the name of a .WAV file to write the data with the
scaled signal, and a scale value. You will need to read all the data from datafile into an array. You can
assume that the file is no bigger than 1000000 sound samples long- that means you should allocate an
sounddata array that is big enough 1000000 sound samples where each sample is a int16_t. You
should then read the entire file into sounddata with one read of 1000000 int16_t's. However, the
read may return less bytes than that. Keep track of the number of bytes that the read returns, since
you can use to calculate how many samples you read into the array - each sample is 2 bytes long (the size
of an int16_t type). In this case, it will be easier to use open and read instead of fopen and fread.
Once you figure out the number of samples in sounddata, you can iterate through the samples in the
sounddata array and multiply the values by scale. You just have to make sure that the resulting
values doesn't exceed 32767 or go below -32768 which are the bounds of an int16_t type.
I have provided you with a write_wav function in wav.c that helps write the data to a .WAV file.
However, before you call write_wav, you need to fill in the way data structure.
The wavdata_t struct is defined as follows:
typedef struct {
uint16_t nchannels;
uint16 t bits_per_sample;
uint32_t sample_rate;
uint32_t datasize;
int16_t *data;
} wavdata_t;
For the provided testfile, nchannels=1, bits_per_sample=16, and sample_rate=22050.
datasize is the number of bytes in the data set, which is equal to the number of bytes you read from
the datafile. data is a pointer to an array that holds the data samples.
Once way data structure is filled in, you can call write_wav.
Testing the code
I have included an expected.5.wavand an expected2.wav file that contains the expected output
for the following runs of the code. You can compare your output with the expected output using the
diff command as shown below. If the files are exactly the same, then the diff command will not
show any differences.
$ gcc -o hw5 hw5.c wav.c
$ ./hw5 datafile my.wav 2
$ diff my.wav expected2.wav
$ ./hw5 datafile my.wav 0.5
$ diff my.wav expected.5.wav
You should be able to play your output .WAV file and hear the change in volume in the audio.
IMPORTANT: make sure that you get no differences with your output.](/v2/_next/image?url=https%3A%2F%2Fcontent.bartleby.com%2Fqna-images%2Fquestion%2F3f696f88-7326-4ac1-a64b-52bf788d01ee%2F17de5910-21ee-4120-8e88-81a97d30a65d%2Fhj3ibdc_processed.png&w=3840&q=75)
Transcribed Image Text:Objective
Your task is to read a data file with sound data and scale the amplitude and then create a.WAV sound file
with the slower audio. WAV files are a Microsoft sound file that stores data samples as raw data - in other
words, there is no compression of the data. I have provided you with a test datafile that has 16-bit samples
recorded at 22050 samples/second.
65535. This anows
Provided Code
I have provided you with the following skeleton main function
int main(int argc, char **argv)
{
if (argc = 4) {
}
printf("%s datafile wavfile scale\n", argv[0]);
exit(-1);
char datafile = argv[1];
char
wavfile = argv[2];
double scale atof (argv[3]);
=
int16_t* sounddata;
// TODO allocate space for 1000000 int16_t's
// TODO open the datafile and read the data into sounddata
// TODO iterate through the array and scale the values
wavdata_t wav;
// TODO fill in the wav struct
write_wav (wavfile, &wav);
Writing the code
The code takes in three arguments - an input datafile, the name of a .WAV file to write the data with the
scaled signal, and a scale value. You will need to read all the data from datafile into an array. You can
assume that the file is no bigger than 1000000 sound samples long- that means you should allocate an
sounddata array that is big enough 1000000 sound samples where each sample is a int16_t. You
should then read the entire file into sounddata with one read of 1000000 int16_t's. However, the
read may return less bytes than that. Keep track of the number of bytes that the read returns, since
you can use to calculate how many samples you read into the array - each sample is 2 bytes long (the size
of an int16_t type). In this case, it will be easier to use open and read instead of fopen and fread.
Once you figure out the number of samples in sounddata, you can iterate through the samples in the
sounddata array and multiply the values by scale. You just have to make sure that the resulting
values doesn't exceed 32767 or go below -32768 which are the bounds of an int16_t type.
I have provided you with a write_wav function in wav.c that helps write the data to a .WAV file.
However, before you call write_wav, you need to fill in the way data structure.
The wavdata_t struct is defined as follows:
typedef struct {
uint16_t nchannels;
uint16 t bits_per_sample;
uint32_t sample_rate;
uint32_t datasize;
int16_t *data;
} wavdata_t;
For the provided testfile, nchannels=1, bits_per_sample=16, and sample_rate=22050.
datasize is the number of bytes in the data set, which is equal to the number of bytes you read from
the datafile. data is a pointer to an array that holds the data samples.
Once way data structure is filled in, you can call write_wav.
Testing the code
I have included an expected.5.wavand an expected2.wav file that contains the expected output
for the following runs of the code. You can compare your output with the expected output using the
diff command as shown below. If the files are exactly the same, then the diff command will not
show any differences.
$ gcc -o hw5 hw5.c wav.c
$ ./hw5 datafile my.wav 2
$ diff my.wav expected2.wav
$ ./hw5 datafile my.wav 0.5
$ diff my.wav expected.5.wav
You should be able to play your output .WAV file and hear the change in volume in the audio.
IMPORTANT: make sure that you get no differences with your output.
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