struct divecomputer *fake_dc(struct divecomputer *dc)
{
static struct sample fake[6];
static struct divecomputer fakedc;
fakedc = (*dc);
fakedc.sample = fake;
fakedc.samples = 6;
/* The dive has no samples, so create a few fake ones */
int max_t = dc->duration.seconds;
int max_d = dc->maxdepth.mm;
int avg_d = dc->meandepth.mm;
memset(fake, 0, sizeof(fake));
fake[5].time.seconds = max_t;
if (!max_t || !max_d)
return &fakedc;
/*
* We want to fake the profile so that the average
* depth ends up correct. However, in the absence of
* a reasonable average, let's just make something
* up. Note that 'avg_d == max_d' is _not_ a reasonable
* average.
* We explicitly treat avg_d == 0 differently */
if (avg_d == 0) {
/* we try for a sane slope, but bow to the insanity of
* the user supplied data */
fill_samples_no_avg(fake, max_d, max_t, MAX(2.0 * max_d / max_t, 5000.0 / 60));
if (fake[3].time.seconds == 0) { // just a 4 point profile
fakedc.samples = 4;
fake[3].time.seconds = max_t;
}
return &fakedc;
}
if (avg_d < max_d / 10 || avg_d >= max_d) {
avg_d = (max_d + 10000) / 3;
if (avg_d > max_d)
avg_d = max_d * 2 / 3;
}
if (!avg_d)
avg_d = 1;
/*
* Ok, first we try a basic profile with a specific ascent
* rate (5 meters per minute) and d_frac (1/3).
*/
if (fill_samples(fake, max_d, avg_d, max_t, 5000.0 / 60, 0.33))
return &fakedc;
/*
* Ok, assume that didn't work because we cannot make the
* average come out right because it was a quick deep dive
* followed by a much shallower region
*/
if (fill_samples(fake, max_d, avg_d, max_t, 10000.0 / 60, 0.10))
return &fakedc;
/*
* Uhhuh. That didn't work. We'd need to find a good combination that
* satisfies our constraints. Currently, we don't, we just give insane
* slopes.
*/
if (fill_samples(fake, max_d, avg_d, max_t, 10000.0, 0.01))
return &fakedc;
/* Even that didn't work? Give up, there's something wrong */
return &fakedc;
}
int main(int argc, char **argv)
{
int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
int src_rate = 48000, dst_rate = 44100;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
const char *fmt;
struct SwrContext *swr_ctx;
double t;
int ret;
if (argc != 2) {
fprintf(stderr, "Usage: %s output_file\n"
"API example program to show how to resample an audio stream with libswresample.\n"
"This program generates a series of audio frames, resamples them to a specified "
"output format and rate and saves them to an output file named output_file.\n",
argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create resampler context */
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* set options */
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
goto end;
}
/* allocate source and destination samples buffers */
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
ret = alloc_samples_array_and_data(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
goto end;
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
max_dst_nb_samples = dst_nb_samples =
av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = alloc_samples_array_and_data(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
goto end;
}
t = 0;
do {
/* generate synthetic audio */
fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_free(dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0)
break;
max_dst_nb_samples = dst_nb_samples;
}
/* convert to destination format */
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
goto end;
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, dst_sample_fmt, 1);
printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
} while (t < 10);
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt) < 0))
goto end;
fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
"ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
end:
if (dst_file)
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);
av_freep(&src_data);
if (dst_data)
av_freep(&dst_data[0]);
av_freep(&dst_data);
swr_free(&swr_ctx);
return ret < 0;
}
