Using libsamplerate with libsndfile

Question 1

With kind help from Erik, I finally got this test code working. My problem was a misunderstanding of audio frames versus audio samples. A frame is composed of one sample per channel. A sample is just that, a number indicating the audio signal at an instant. I thought I knew the difference, but forgot when applying Erik's examples code seen in the main loop. That code is from sndfile-resample.c in the examples directory in the libsamplerate distribution tarball. The consequence of this misunderstanding was that in stereo samples, the last few samples (around 23 to 60 depending on the buffer size) would be zero. That caused jittery playback. If I reduced the buffer size to 512, I got distortion that sounded like a ring modulator on an analogue synthesizer. Note the change from sf_read_float() to sf_readf_float(). The loop in floattopcm16() was incorrectly testing count <= length. I have corrected it to count < length.

For those who are also having trouble, here is code that works and passes -Wall.

/*
 * Objective: sample rate conversion
 * compile with
 * "gcc -o glurp glurp.c -lao -lsndfile -lsamplerate"
 *
 */

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <limits.h>
#include <ao/ao.h>
#include <sndfile.h>
#include <samplerate.h>

#define DEFAULT_CONVERTER SRC_SINC_MEDIUM_QUALITY
#define NEW_RATE 44100

#define BUFFSIZE 4096
#define MAX(x,y) ((x)>(y)) ? (x) : (y)
#define MIN(x,y) ((x)<(y)) ? (x) : (y)

int playfile(FILE *);
void floattopcm16(short *, float *, int);
void pcm16tofloat(float *, short *, int);

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

    if (argc != 2) {
        printf("usage: %s <input>\n", argv[0]);
    exit(1);
    }

    fp = fopen(argv[1], "rb");
    if (fp == NULL) {
        printf("Cannot open %s.\n", argv[1]);
    exit(1);
    }

    playfile(fp);
    fclose(fp);

    return 0;
}

int playfile(FILE *fp)
{
    int default_driver;
    float *floatbuffer;
    float *floatbuffer2;
    short *shortbuffer;
    long filestart;

    int newrate = NEW_RATE;

    ao_device *device;
    ao_sample_format format;
    SNDFILE     *sndfile;
    SF_INFO sf_info;

    SRC_STATE   *src_state;
    SRC_DATA    src_data;
    int     error;
    sf_count_t  output_count = 0;

    ao_initialize();
    default_driver = ao_default_driver_id();

    sf_info.format = 0;

    filestart = ftell(fp);

    sndfile = sf_open_fd(fileno(fp), SFM_READ, &sf_info, 0);

    memset(&format, 0, sizeof(ao_sample_format));

    format.byte_format = AO_FMT_NATIVE;
    format.bits = 16;
    format.channels = sf_info.channels;
    format.rate = newrate;

    printf("Channels:           %d\n", sf_info.channels);
    printf("Start sample rate:  %d\n", sf_info.samplerate);
    printf("Ending sample rate: %d\n", newrate);

    device = ao_open_live(default_driver, &format, NULL /* no options */);
    if (device == NULL) {
        printf("Error opening sound device.\n");
        return 1;
    }

    floatbuffer = malloc(BUFFSIZE * sf_info.channels * sizeof(float));
    floatbuffer2 = malloc(BUFFSIZE * sf_info.channels * sizeof(float));
    shortbuffer = malloc(BUFFSIZE * sf_info.channels * sizeof(short));

    /* Set up for conversion */
    if ((src_state = src_new(DEFAULT_CONVERTER, sf_info.channels, &error)) == NULL) {
        printf("Error: src_new() failed: %s.\n", src_strerror(error));
        exit(1);
    }
    src_data.end_of_input = 0;
    src_data.input_frames = 0;
    src_data.data_in = floatbuffer;
    src_data.src_ratio = (1.0 * newrate) / sf_info.samplerate;
    src_data.data_out = floatbuffer2;
    src_data.output_frames = BUFFSIZE / sf_info.channels;

    while (1) {
        /* if floatbuffer is empty, refill it */
        if (src_data.input_frames == 0) {
            src_data.input_frames = sf_readf_float(sndfile, floatbuffer, BUFFSIZE / sf_info.channels);
            src_data.data_in = floatbuffer;

            /* mark end of input */
            if (src_data.input_frames < BUFFSIZE / sf_info.channels)
                src_data.end_of_input = SF_TRUE;
        }

        if ((error = src_process(src_state, &src_data))) {
            printf("Error: %s\n", src_strerror(error));
            exit(1);
        }

        /* terminate if done */
        if (src_data.end_of_input && src_data.output_frames_gen == 0)
            break;

        /* write output */
        floattopcm16(shortbuffer, floatbuffer2, src_data.output_frames_gen * sf_info.channels);
        ao_play(device, (char *)shortbuffer, src_data.output_frames_gen * sizeof(short) * sf_info.channels);

        output_count += src_data.output_frames_gen;
        src_data.data_in += src_data.input_frames_used * sf_info.channels;
        src_data.input_frames -= src_data.input_frames_used;
    }

    src_state = src_delete(src_state);

    free(shortbuffer);
    free(floatbuffer);
    free(floatbuffer2);
    fseek(fp, filestart, SEEK_SET);
    ao_close(device);
    sf_close(sndfile);
    ao_shutdown();
    printf("Finished\n");

    return 0;
}


/* Convert back to shorts */
void floattopcm16(short *outbuf, float *inbuf, int length)
{
    int   count;

    const float mul = (32768.0f);
    for (count = 0; count < length; count++) {
        int32_t tmp = (int32_t)(mul * inbuf[count]);
        tmp = MAX( tmp, -32768 ); // CLIP < 32768
        tmp = MIN( tmp, 32767 );  // CLIP > 32767
        outbuf[count] = tmp;
    }
}

Question 2

Libsndfile doesn't deinterlace stereo audio, you have to do it manually.