FFmpeg/libavresample/avresample-test.c

/*
 * Copyright (c) 2002 Fabrice Bellard
 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg 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.
 *
 * FFmpeg 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 FFmpeg; 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/common.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 sine 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 varying 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, out_data, out_linesize, out_rate * 6,
                                                         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;
}