4
PortAudio library을 사용하여 오디오를 캡처 한 코드를 가지고 WAV codec에 패키지하려고합니다. 그런 다음 해당 파일을 FLAC file codec으로 변환하려고 시도하지만 올바른 .wav 파일이 아닙니다.잘못된 WAV 파일 생성
wav.h :
#include <stdint.h>
#include <string.h>
/**
* @struct WaveHeader
* @brief A basic WAVE header
*/
typedef struct
{
char RIFF_marker[4];
uint32_t file_size;
char filetype_header[4];
char format_marker[4];
uint32_t data_header_length;
uint16_t format_type;
uint16_t number_of_channels;
uint32_t sample_rate;
uint32_t bytes_per_second;
uint16_t bytes_per_frame;
uint16_t bits_per_sample;
} WaveHeader;
WaveHeader *genericWAVHeader(uint32_t sample_rate, uint16_t bit_depth, uint16_t channels);
WaveHeader *retrieveWAVHeader(const void *ptr);
int writeWAVHeader(FILE* fd, WaveHeader *hdr);
int recordWAV(const char *fileName, WaveHeader *hdr, uint32_t duration);
capture.c :
#include <stdio.h>
#include <stdlib.h>
#include <portaudio.h>
#include <unistd.h>
#include <math.h>
#include "wav.h"
typedef struct
{
int frameIndex; /* Index into sample array. */
int maxFrameIndex;
char* recordedSamples;
} PAData;
WaveHeader *genericWAVHeader(uint32_t sample_rate, uint16_t bit_depth, uint16_t channels)
{
WaveHeader *hdr = malloc(sizeof(*hdr));
if (!hdr) return NULL;
memcpy(&hdr->RIFF_marker, "RIFF", 4);
memcpy(&hdr->filetype_header, "WAVE", 4);
memcpy(&hdr->format_marker, "fmt ", 4);
hdr->data_header_length = 16;
hdr->format_type = 1;
hdr->number_of_channels = channels;
hdr->sample_rate = sample_rate;
hdr->bytes_per_second = sample_rate * channels * bit_depth/8;
hdr->bytes_per_frame = channels * bit_depth/8;
hdr->bits_per_sample = bit_depth;
return hdr;
}
int writeWAVHeader(FILE* fd, WaveHeader *hdr)
{
if (!hdr) return -1;
fwrite(&hdr->RIFF_marker, sizeof(*hdr->RIFF_marker), sizeof(&hdr->RIFF_marker), fd);
fwrite(&hdr->file_size, sizeof(hdr->file_size), sizeof(&hdr->file_size), fd);
fwrite(&hdr->filetype_header, sizeof(*hdr->filetype_header), sizeof(&hdr->filetype_header), fd);
fwrite(&hdr->format_marker, sizeof(*hdr->format_marker), sizeof(&hdr->format_marker), fd);
fwrite(&hdr->data_header_length, sizeof(hdr->data_header_length), sizeof(&hdr->data_header_length), fd);
fwrite(&hdr->format_type, sizeof(hdr->format_type), sizeof(&hdr->format_type), fd);
fwrite(&hdr->number_of_channels, sizeof(hdr->number_of_channels), sizeof(&hdr->number_of_channels), fd);
fwrite(&hdr->sample_rate, sizeof(hdr->sample_rate), sizeof(&hdr->sample_rate), fd);
fwrite(&hdr->bytes_per_second, sizeof(hdr->bytes_per_second), sizeof(&hdr->bytes_per_second), fd);
fwrite(&hdr->bytes_per_frame, sizeof(hdr->bytes_per_frame), sizeof(&hdr->bytes_per_frame), fd);
fwrite(&hdr->bits_per_sample, sizeof(hdr->bits_per_sample), sizeof(&hdr->bits_per_sample), fd);
fwrite("data", 1, sizeof("data") - 1, fd);
uint32_t data_size = hdr->file_size - 36;
fwrite(&data_size, sizeof(data_size), sizeof(&data_size), fd);
return 0;
}
static int recordCallback(const void *inputBuffer, void *outputBuffer, unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo* timeInfo, PaStreamCallbackFlags statusFlags, void *userData)
{
PAData *data = (PAData*) userData;
const char *rptr = (const char*) inputBuffer;
char *wptr = &data->recordedSamples[data->frameIndex * 2 /*channels*/];
long framesToCalc;
int finished;
unsigned long framesLeft = data->maxFrameIndex - data->frameIndex;
(void) outputBuffer; /* Prevent unused variable warnings. */
(void) timeInfo;
(void) statusFlags;
(void) userData;
if(framesLeft < framesPerBuffer)
{
framesToCalc = framesLeft;
finished = paComplete;
}
else
{
framesToCalc = framesPerBuffer;
finished = paContinue;
}
if(!inputBuffer)
{
for(long i = 0; i < framesToCalc; i++)
{
*wptr++ = 0; /* left */
*wptr++ = 0; /* right */
}
}
else
{
for(long i = 0; i < framesToCalc; i++)
{
*wptr++ = *rptr++; /* left */
*wptr++ = *rptr++; /* right */
}
}
data->frameIndex += framesToCalc;
return finished;
}
int recordWAV(const char *fileName, WaveHeader *hdr, uint32_t duration)
{
PaStreamParameters inputParameters;
PaStream* stream;
PaError err = paNoError;
PAData data;
int totalFrames;
int numSamples;
int numBytes;
char max, val;
double average;
printf("%d", hdr->bytes_per_second);
data.maxFrameIndex = totalFrames = duration * hdr->sample_rate; /* Record for a few seconds. */
data.frameIndex = 0;
numSamples = totalFrames * hdr->number_of_channels;
numBytes = numSamples;
data.recordedSamples = malloc(numBytes);
if(!data.recordedSamples)
{
printf("Could not allocate record array.\n");
goto done;
}
for(int i = 0; i < numSamples; i++) data.recordedSamples[i] = 0;
if((err = Pa_Initialize())) goto done;
inputParameters.device = Pa_GetDefaultInputDevice(); /* default input device */
if (inputParameters.device == paNoDevice) {
fprintf(stderr,"Error: No default input device.\n");
goto done;
}
inputParameters.channelCount = 2; /* stereo input */
inputParameters.sampleFormat = 1;
inputParameters.suggestedLatency = Pa_GetDeviceInfo(inputParameters.device)->defaultLowInputLatency;
inputParameters.hostApiSpecificStreamInfo = NULL;
/* Record some audio. -------------------------------------------- */
err = Pa_OpenStream(&stream, &inputParameters, NULL, hdr->sample_rate, paFramesPerBufferUnspecified, paClipOff, recordCallback, &data);
if(err) goto done;
if((err = Pa_StartStream(stream))) goto done;
puts("Now recording!! Please speak into the microphone.");
while((err = Pa_IsStreamActive(stream)) == 1)
{
Pa_Sleep(1000);
printf("index = %d\n", data.frameIndex);
}
if(err < 0) goto done;
err = Pa_CloseStream(stream);
if(err) goto done;
/* Measure maximum peak amplitude. */
max = 0;
average = 0.0;
for(int i = 0; i < numSamples; i++)
{
val = data.recordedSamples[i];
val = abs(val);
if(val > max)
{
max = val;
}
average += val;
}
average /= (double)numSamples;
{
FILE* fid = fopen(fileName, "wb");
if(!fid) printf("Could not open file.");
else
{
writeWAVHeader(fid, hdr);
fwrite(data.recordedSamples, hdr->number_of_channels, totalFrames, fid);
fclose(fid);
}
}
done:
Pa_Terminate();
if(data.recordedSamples) /* Sure it is NULL or valid. */
free(data.recordedSamples);
if(err)
{
fprintf(stderr, "An error occured while using the portaudio stream\n");
fprintf(stderr, "Error number: %d\n", err);
fprintf(stderr, "Error message: %s\n", Pa_GetErrorText(err));
err = 1; /* Always return 0 or 1, but no other return codes. */
}
return err;
}
I 파일에서 실행 일부 테스트 :
$ xxd -g1 test.wav | head
0000000: 52 49 46 46 00 00 00 30 00 00 00 30 57 41 56 45 RIFF...0...0WAVE
0000010: 66 6d 74 20 10 00 00 00 01 00 02 00 e0 ab 00 00 fmt ............
0000020: 80 af 02 00 04 00 10 00 57 41 56 45 66 6d 74 20 ........WAVEfmt
0000030: 66 6d 74 20 10 00 00 00 10 00 00 00 01 00 02 00 fmt ............
0000040: e0 ab 00 00 80 af 02 00 04 00 10 00 00 00 00 80 ................
0000050: 37 3d 04 10 00 00 03 00 01 00 02 00 e0 ab 00 00 7=..............
0000060: 80 af 02 00 04 00 10 00 02 00 e0 ab 00 00 80 af ................
0000070: 02 00 04 00 10 00 00 00 e0 ab 00 00 80 af 02 00 ................
0000080: 04 00 10 00 00 00 00 80 37 3d 04 10 00 00 03 00 ........7=......
0000090: 00 00 00 00 00 00 00 00 80 af 02 00 04 00 10 00 ................
$ file test.wav
test.wav: RIFF (little-endian) data
$ stat -f %s test.wav
stat: %s: bad format
꽤 확실하지 않다 내가 잘못 가고있는 곳. 데이터를 올바르게 쓰지 않습니까?
WAV 파일 형식이 온라인 상태입니다. 바이트 단위로 크롤링하고 무엇이 잘못되었는지 알아냅니다. –
'fwrite'를주의하여 실행하십시오. 거기에 오류가 있습니다. 특히 세 번째 매개 변수는 주소가되어서는 안됩니다. 실제로는 '1'이어야합니다. – ooga
두 번째 4 바이트의 길이는 리틀 엔디안이어야합니다. "WAVE"는 바이트 8에서 시작하기로되어 있습니다. 그 이상의 것을 시도하지 않았습니다. –