Writing improper number of frames using PortAudio? - c

Question

Running my program, I appear to not be writing the correct amount of frames according to the index.
$ ./test
Now recording!! Please speak into the microphone.
index = 0
Writing to: test.flac
audio.h:
#include <stdint.h>
#include <string.h>
typedef struct
{
uint32_t duration;
uint16_t format_type;
uint16_t number_of_channels;
uint32_t sample_rate;
uint32_t frameIndex; /* Index into sample array. */
uint32_t maxFrameIndex;
char* recordedSamples;
} AudioData;
int recordFLAC(AudioData* data, const char *fileName);
AudioData* initAudioData(uint32_t sample_rate, uint16_t channels, uint32_t duration);
capture.c:
#include <stdio.h>
#include <stdlib.h>
#include <portaudio.h>
#include <sndfile.h>
#include "audio.h"
AudioData* initAudioData(uint32_t sample_rate, uint16_t channels, uint32_t duration)
{
AudioData* data = malloc(sizeof(*data));
if (!data) return NULL;
data->duration = duration;
data->format_type = 1;
data->number_of_channels = channels;
data->sample_rate = sample_rate;
data->frameIndex = 0;
data->maxFrameIndex = sample_rate * duration;
data->recordedSamples = malloc(sizeof(data->maxFrameIndex));
if(!data->maxFrameIndex) return NULL;
return data;
}
static int recordCallback(const void *inputBuffer, void *outputBuffer, unsigned long frameCount, const PaStreamCallbackTimeInfo* timeInfo, PaStreamCallbackFlags statusFlags, void *userData)
{
AudioData* data = (AudioData*)userData;
const char* buffer_ptr = (const char*)inputBuffer;
char* index_ptr = &data->recordedSamples[data->frameIndex];
(void) outputBuffer;
(void) timeInfo;
(void) statusFlags;
long framesToCalc;
long i;
int finished;
unsigned long framesLeft = data->maxFrameIndex - data->frameIndex;
if(framesLeft < frameCount){
framesToCalc = framesLeft;
finished = paComplete;
}else{
framesToCalc = frameCount;
finished = paContinue;
}
if(!inputBuffer){
for(i = 0; i < framesToCalc; i++){
*index_ptr++ = 0;
}
}else{
for(i = 0; i < framesToCalc; i++){
*index_ptr++ = *buffer_ptr++;
}
}
data->frameIndex += framesToCalc;
return finished;
}
int recordFLAC(AudioData* data, const char *fileName)
{
PaStreamParameters inputParameters;
PaStream* stream;
int err = 0;
int totalFrames = data->maxFrameIndex;
int numSamples;
int numBytes;
char max, val;
double average;
numSamples = totalFrames * data->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 = data->number_of_channels; /* stereo input */
inputParameters.sampleFormat = data->format_type;
inputParameters.suggestedLatency = Pa_GetDeviceInfo(inputParameters.device)->defaultLowInputLatency;
inputParameters.hostApiSpecificStreamInfo = NULL;
/* Record some audio. -------------------------------------------- */
err = Pa_OpenStream(&stream, &inputParameters, NULL, data->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;
done:
Pa_Terminate();
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. */
}
else
{
SF_INFO sfinfo;
sfinfo.channels = 1;
sfinfo.samplerate = data->sample_rate;
sfinfo.format = SF_FORMAT_FLAC | SF_FORMAT_PCM_16;
// open to file
printf("Writing to: %s\n", fileName);
SNDFILE * outfile = sf_open(fileName, SFM_WRITE, &sfinfo);
if (!outfile) return -1;
// prepare a 3 second long buffer (sine wave)
const int size = data->sample_rate * 3;
// write the entire buffer to the file
sf_write_raw(outfile, data->recordedSamples, size);
// force write to disk
sf_write_sync(outfile);
// don't forget to close the file
sf_close(outfile);
}
return err;
}
I'm not quite sure where I am going wrong, I know I need to be writing more frames. Any suggestions?

Answer 1

There seems to be something wrong with your assumptions about the sample format. In the callback you are using char * (single bytes) for the sample format, but in your libsndfile call you're opening a 16 bit file with SF_FORMAT_PCM_16.
This is not so good:
data->format_type = 1;
I recommend using one of the symbolic constants in the PortAudio library for sample formatting. Maybe you want a 16 bit one? And if so, you want to be using short* not char* in the PA callback.
Finally, if your channel count is not 1, the copy loops are incorrect:
for(i = 0; i < framesToCalc; i++){
*index_ptr++ = 0;
}
A "frame" contains data for all channels, so for example, if it's a stereo input your iteration needs to deal with both left and right channels like this:
for(i = 0; i < framesToCalc; i++){
*index_ptr++ = 0; // left
*index_ptr++ = 0; // right
}
Same for the other loops.