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FFmpeg/libavresample/avresample-test.c
Justin Ruggles c8af852b97 Add libavresample 
This is a new library for audio sample format, channel layout, and sample rate
conversion.
2012-04-24 21:28:27 -04:00

341 lines
12 KiB
C
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/*
* Copyright (c) 2002 Fabrice Bellard
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdint.h>
#include <stdio.h>
#include "libavutil/avstring.h"
#include "libavutil/lfg.h"
#include "libavutil/libm.h"
#include "libavutil/log.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "avresample.h"
static double dbl_rand(AVLFG *lfg)
{
return 2.0 * (av_lfg_get(lfg) / (double)UINT_MAX) - 1.0;
}
#define PUT_FUNC(name, fmt, type, expr) \
static void put_sample_ ## name(void **data, enum AVSampleFormat sample_fmt,\
int channels, int sample, int ch, \
double v_dbl) \
{ \
type v = expr; \
type **out = (type **)data; \
if (av_sample_fmt_is_planar(sample_fmt)) \
out[ch][sample] = v; \
else \
out[0][sample * channels + ch] = v; \
}
PUT_FUNC(u8, AV_SAMPLE_FMT_U8, uint8_t, av_clip_uint8 ( lrint(v_dbl * (1 << 7)) + 128))
PUT_FUNC(s16, AV_SAMPLE_FMT_S16, int16_t, av_clip_int16 ( lrint(v_dbl * (1 << 15))))
PUT_FUNC(s32, AV_SAMPLE_FMT_S32, int32_t, av_clipl_int32(llrint(v_dbl * (1U << 31))))
PUT_FUNC(flt, AV_SAMPLE_FMT_FLT, float, v_dbl)
PUT_FUNC(dbl, AV_SAMPLE_FMT_DBL, double, v_dbl)
static void put_sample(void **data, enum AVSampleFormat sample_fmt,
int channels, int sample, int ch, double v_dbl)
{
switch (av_get_packed_sample_fmt(sample_fmt)) {
case AV_SAMPLE_FMT_U8:
put_sample_u8(data, sample_fmt, channels, sample, ch, v_dbl);
break;
case AV_SAMPLE_FMT_S16:
put_sample_s16(data, sample_fmt, channels, sample, ch, v_dbl);
break;
case AV_SAMPLE_FMT_S32:
put_sample_s32(data, sample_fmt, channels, sample, ch, v_dbl);
break;
case AV_SAMPLE_FMT_FLT:
put_sample_flt(data, sample_fmt, channels, sample, ch, v_dbl);
break;
case AV_SAMPLE_FMT_DBL:
put_sample_dbl(data, sample_fmt, channels, sample, ch, v_dbl);
break;
}
}
static void audiogen(AVLFG *rnd, void **data, enum AVSampleFormat sample_fmt,
int channels, int sample_rate, int nb_samples)
{
int i, ch, k;
double v, f, a, ampa;
double tabf1[AVRESAMPLE_MAX_CHANNELS];
double tabf2[AVRESAMPLE_MAX_CHANNELS];
double taba[AVRESAMPLE_MAX_CHANNELS];
#define PUT_SAMPLE put_sample(data, sample_fmt, channels, k, ch, v);
k = 0;
/* 1 second of single freq sinus at 1000 Hz */
a = 0;
for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
v = sin(a) * 0.30;
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
a += M_PI * 1000.0 * 2.0 / sample_rate;
}
/* 1 second of varing frequency between 100 and 10000 Hz */
a = 0;
for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
v = sin(a) * 0.30;
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
f = 100.0 + (((10000.0 - 100.0) * i) / sample_rate);
a += M_PI * f * 2.0 / sample_rate;
}
/* 0.5 second of low amplitude white noise */
for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
v = dbl_rand(rnd) * 0.30;
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
}
/* 0.5 second of high amplitude white noise */
for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
v = dbl_rand(rnd);
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
}
/* 1 second of unrelated ramps for each channel */
for (ch = 0; ch < channels; ch++) {
taba[ch] = 0;
tabf1[ch] = 100 + av_lfg_get(rnd) % 5000;
tabf2[ch] = 100 + av_lfg_get(rnd) % 5000;
}
for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
for (ch = 0; ch < channels; ch++) {
v = sin(taba[ch]) * 0.30;
PUT_SAMPLE
f = tabf1[ch] + (((tabf2[ch] - tabf1[ch]) * i) / sample_rate);
taba[ch] += M_PI * f * 2.0 / sample_rate;
}
}
/* 2 seconds of 500 Hz with varying volume */
a = 0;
ampa = 0;
for (i = 0; i < 2 * sample_rate && k < nb_samples; i++, k++) {
for (ch = 0; ch < channels; ch++) {
double amp = (1.0 + sin(ampa)) * 0.15;
if (ch & 1)
amp = 0.30 - amp;
v = sin(a) * amp;
PUT_SAMPLE
a += M_PI * 500.0 * 2.0 / sample_rate;
ampa += M_PI * 2.0 / sample_rate;
}
}
}
/* formats, rates, and layouts are ordered for priority in testing.
e.g. 'avresample-test 4 2 2' will test all input/output combinations of
S16/FLTP/S16P/FLT, 48000/44100, and stereo/mono */
static const enum AVSampleFormat formats[] = {
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_U8P,
AV_SAMPLE_FMT_U8,
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_DBL,
};
static const int rates[] = {
48000,
44100,
16000
};
static const uint64_t layouts[] = {
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_5POINT1,
AV_CH_LAYOUT_7POINT1,
};
int main(int argc, char **argv)
{
AVAudioResampleContext *s;
AVLFG rnd;
int ret = 0;
uint8_t *in_buf = NULL;
uint8_t *out_buf = NULL;
unsigned int in_buf_size;
unsigned int out_buf_size;
uint8_t *in_data[AVRESAMPLE_MAX_CHANNELS] = { 0 };
uint8_t *out_data[AVRESAMPLE_MAX_CHANNELS] = { 0 };
int in_linesize;
int out_linesize;
uint64_t in_ch_layout;
int in_channels;
enum AVSampleFormat in_fmt;
int in_rate;
uint64_t out_ch_layout;
int out_channels;
enum AVSampleFormat out_fmt;
int out_rate;
int num_formats, num_rates, num_layouts;
int i, j, k, l, m, n;
num_formats = 2;
num_rates = 2;
num_layouts = 2;
if (argc > 1) {
if (!av_strncasecmp(argv[1], "-h", 3)) {
av_log(NULL, AV_LOG_INFO, "Usage: avresample-test [<num formats> "
"[<num sample rates> [<num channel layouts>]]]\n"
"Default is 2 2 2\n");
return 0;
}
num_formats = strtol(argv[1], NULL, 0);
num_formats = av_clip(num_formats, 1, FF_ARRAY_ELEMS(formats));
}
if (argc > 2) {
num_rates = strtol(argv[2], NULL, 0);
num_rates = av_clip(num_rates, 1, FF_ARRAY_ELEMS(rates));
}
if (argc > 3) {
num_layouts = strtol(argv[3], NULL, 0);
num_layouts = av_clip(num_layouts, 1, FF_ARRAY_ELEMS(layouts));
}
av_log_set_level(AV_LOG_DEBUG);
av_lfg_init(&rnd, 0xC0FFEE);
in_buf_size = av_samples_get_buffer_size(&in_linesize, 8, 48000 * 6,
AV_SAMPLE_FMT_DBLP, 0);
out_buf_size = in_buf_size;
in_buf = av_malloc(in_buf_size);
if (!in_buf)
goto end;
out_buf = av_malloc(out_buf_size);
if (!out_buf)
goto end;
s = avresample_alloc_context();
if (!s) {
av_log(NULL, AV_LOG_ERROR, "Error allocating AVAudioResampleContext\n");
ret = 1;
goto end;
}
for (i = 0; i < num_formats; i++) {
in_fmt = formats[i];
for (k = 0; k < num_layouts; k++) {
in_ch_layout = layouts[k];
in_channels = av_get_channel_layout_nb_channels(in_ch_layout);
for (m = 0; m < num_rates; m++) {
in_rate = rates[m];
ret = av_samples_fill_arrays(in_data, &in_linesize, in_buf,
in_channels, in_rate * 6,
in_fmt, 0);
if (ret < 0) {
av_log(s, AV_LOG_ERROR, "failed in_data fill arrays\n");
goto end;
}
audiogen(&rnd, (void **)in_data, in_fmt, in_channels, in_rate, in_rate * 6);
for (j = 0; j < num_formats; j++) {
out_fmt = formats[j];
for (l = 0; l < num_layouts; l++) {
out_ch_layout = layouts[l];
out_channels = av_get_channel_layout_nb_channels(out_ch_layout);
for (n = 0; n < num_rates; n++) {
out_rate = rates[n];
av_log(NULL, AV_LOG_INFO, "%s to %s, %d to %d channels, %d Hz to %d Hz\n",
av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt),
in_channels, out_channels, in_rate, out_rate);
ret = av_samples_fill_arrays(out_data, &out_linesize,
out_buf, out_channels,
out_rate * 6, out_fmt, 0);
if (ret < 0) {
av_log(s, AV_LOG_ERROR, "failed out_data fill arrays\n");
goto end;
}
av_opt_set_int(s, "in_channel_layout", in_ch_layout, 0);
av_opt_set_int(s, "in_sample_fmt", in_fmt, 0);
av_opt_set_int(s, "in_sample_rate", in_rate, 0);
av_opt_set_int(s, "out_channel_layout", out_ch_layout, 0);
av_opt_set_int(s, "out_sample_fmt", out_fmt, 0);
av_opt_set_int(s, "out_sample_rate", out_rate, 0);
av_opt_set_int(s, "internal_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
ret = avresample_open(s);
if (ret < 0) {
av_log(s, AV_LOG_ERROR, "Error opening context\n");
goto end;
}
ret = avresample_convert(s, (void **)out_data, out_linesize, out_rate * 6,
(void **) in_data, in_linesize, in_rate * 6);
if (ret < 0) {
char errbuf[256];
av_strerror(ret, errbuf, sizeof(errbuf));
av_log(NULL, AV_LOG_ERROR, "%s\n", errbuf);
goto end;
}
av_log(NULL, AV_LOG_INFO, "Converted %d samples to %d samples\n",
in_rate * 6, ret);
if (avresample_get_delay(s) > 0)
av_log(NULL, AV_LOG_INFO, "%d delay samples not converted\n",
avresample_get_delay(s));
if (avresample_available(s) > 0)
av_log(NULL, AV_LOG_INFO, "%d samples available for output\n",
avresample_available(s));
av_log(NULL, AV_LOG_INFO, "\n");
avresample_close(s);
}
}
}
}
}
}
ret = 0;
end:
av_freep(&in_buf);
av_freep(&out_buf);
avresample_free(&s);
return ret;
}
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