/*

 * 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 "error.h"

#include "macros.h"

#include "mem.h"

#include "samplefmt.h"

#include <limits.h>

#include <stdio.h>

#include <string.h>

typedef struct SampleFmtInfo {

    char name[8];

    int bits;

    int planar;

    enum AVSampleFormat altform; ///< planar<->packed alternative form

} SampleFmtInfo;

/** this table gives more information about formats */

static const SampleFmtInfo sample_fmt_info[AV_SAMPLE_FMT_NB] = {

    [AV_SAMPLE_FMT_U8]   = { .name =   "u8", .bits =  8, .planar = 0, .altform = AV_SAMPLE_FMT_U8P  },

    [AV_SAMPLE_FMT_S16]  = { .name =  "s16", .bits = 16, .planar = 0, .altform = AV_SAMPLE_FMT_S16P },

    [AV_SAMPLE_FMT_S32]  = { .name =  "s32", .bits = 32, .planar = 0, .altform = AV_SAMPLE_FMT_S32P },

    [AV_SAMPLE_FMT_S64]  = { .name =  "s64", .bits = 64, .planar = 0, .altform = AV_SAMPLE_FMT_S64P },

    [AV_SAMPLE_FMT_FLT]  = { .name =  "flt", .bits = 32, .planar = 0, .altform = AV_SAMPLE_FMT_FLTP },

    [AV_SAMPLE_FMT_DBL]  = { .name =  "dbl", .bits = 64, .planar = 0, .altform = AV_SAMPLE_FMT_DBLP },

    [AV_SAMPLE_FMT_U8P]  = { .name =  "u8p", .bits =  8, .planar = 1, .altform = AV_SAMPLE_FMT_U8   },

    [AV_SAMPLE_FMT_S16P] = { .name = "s16p", .bits = 16, .planar = 1, .altform = AV_SAMPLE_FMT_S16  },

    [AV_SAMPLE_FMT_S32P] = { .name = "s32p", .bits = 32, .planar = 1, .altform = AV_SAMPLE_FMT_S32  },

    [AV_SAMPLE_FMT_S64P] = { .name = "s64p", .bits = 64, .planar = 1, .altform = AV_SAMPLE_FMT_S64  },

    [AV_SAMPLE_FMT_FLTP] = { .name = "fltp", .bits = 32, .planar = 1, .altform = AV_SAMPLE_FMT_FLT  },

    [AV_SAMPLE_FMT_DBLP] = { .name = "dblp", .bits = 64, .planar = 1, .altform = AV_SAMPLE_FMT_DBL  },

    [AV_SAMPLE_FMT_DSD]  = { .name =  "dsd", .bits =  8, .planar = 0, .altform = AV_SAMPLE_FMT_DSD  },

};

const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)

{

    if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)

        return NULL;

    return sample_fmt_info[sample_fmt].name;

}

enum AVSampleFormat av_get_sample_fmt(const char *name)

{

    int i;

    for (i = 0; i < AV_SAMPLE_FMT_NB; i++)

        if (!strcmp(sample_fmt_info[i].name, name))

            return i;

    return AV_SAMPLE_FMT_NONE;

}

enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar)

{

    if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)

        return AV_SAMPLE_FMT_NONE;

    if (sample_fmt_info[sample_fmt].planar == planar)

        return sample_fmt;

    return sample_fmt_info[sample_fmt].altform;

}

enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)

{

    if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)

        return AV_SAMPLE_FMT_NONE;

    if (sample_fmt_info[sample_fmt].planar)

        return sample_fmt_info[sample_fmt].altform;

    return sample_fmt;

}

enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)

{

    if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)

        return AV_SAMPLE_FMT_NONE;

    if (sample_fmt_info[sample_fmt].planar)

        return sample_fmt;

    return sample_fmt_info[sample_fmt].altform;

}

char *av_get_sample_fmt_string (char *buf, int buf_size, enum AVSampleFormat sample_fmt)

{

    /* print header */

    if (sample_fmt < 0)

        snprintf(buf, buf_size, "name  " " depth");

    else if (sample_fmt < AV_SAMPLE_FMT_NB) {

        SampleFmtInfo info = sample_fmt_info[sample_fmt];

        snprintf (buf, buf_size, "%-6s" "   %2d ", info.name, info.bits);

    }

    return buf;

}

int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)

{

     return sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB ?

        0 : sample_fmt_info[sample_fmt].bits >> 3;

}

int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)

{

     if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)

         return 0;

     return sample_fmt_info[sample_fmt].planar;

}

int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,

                               enum AVSampleFormat sample_fmt, int align)

{

    int line_size;

    int sample_size = av_get_bytes_per_sample(sample_fmt);

    int planar      = av_sample_fmt_is_planar(sample_fmt);

    /* validate parameter ranges */

    if (!sample_size || nb_samples <= 0 || nb_channels <= 0)

        return AVERROR(EINVAL);

    /* auto-select alignment if not specified */

    if (!align) {

        if (nb_samples > INT_MAX - 31)

            return AVERROR(EINVAL);

        align = 1;

        nb_samples = FFALIGN(nb_samples, 32);

    }

    /* check for integer overflow */

    if (nb_channels > INT_MAX / align ||

        (int64_t)nb_channels * nb_samples > (INT_MAX - (align * nb_channels)) / sample_size)

        return AVERROR(EINVAL);

    line_size = planar ? FFALIGN(nb_samples * sample_size,               align) :

                         FFALIGN(nb_samples * sample_size * nb_channels, align);

    if (linesize)

        *linesize = line_size;

    return planar ? line_size * nb_channels : line_size;

}

int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,

                           const uint8_t *buf, int nb_channels, int nb_samples,

                           enum AVSampleFormat sample_fmt, int align)

{

    int ch, planar, buf_size, line_size;

    planar   = av_sample_fmt_is_planar(sample_fmt);

    buf_size = av_samples_get_buffer_size(&line_size, nb_channels, nb_samples,

                                          sample_fmt, align);

    if (buf_size < 0)

        return buf_size;

    if (linesize)

        *linesize = line_size;

    memset(audio_data, 0, planar

                          ? sizeof(*audio_data) * nb_channels

                          : sizeof(*audio_data));

    if (!buf)

        return buf_size;

    audio_data[0] = (uint8_t *)buf;

    for (ch = 1; planar && ch < nb_channels; ch++)

        audio_data[ch] = audio_data[ch-1] + line_size;

    return buf_size;

}

int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,

                     int nb_samples, enum AVSampleFormat sample_fmt, int align)

{

    uint8_t *buf;

    int size = av_samples_get_buffer_size(NULL, nb_channels, nb_samples,

                                          sample_fmt, align);

    if (size < 0)

        return size;

    buf = av_malloc(size);

    if (!buf)

        return AVERROR(ENOMEM);

    size = av_samples_fill_arrays(audio_data, linesize, buf, nb_channels,

                                  nb_samples, sample_fmt, align);

    if (size < 0) {

        av_free(buf);

        return size;

    }

    av_samples_set_silence(audio_data, 0, nb_samples, nb_channels, sample_fmt);

    return size;

}

int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels,

                                       int nb_samples, enum AVSampleFormat sample_fmt, int align)

{

    int ret, nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;

    *audio_data = av_calloc(nb_planes, sizeof(**audio_data));

    if (!*audio_data)

        return AVERROR(ENOMEM);

    ret = av_samples_alloc(*audio_data, linesize, nb_channels,

                           nb_samples, sample_fmt, align);

    if (ret < 0)

        av_freep(audio_data);

    return ret;

}

int av_samples_copy(uint8_t * const *dst, uint8_t * const *src, int dst_offset,

                    int src_offset, int nb_samples, int nb_channels,

                    enum AVSampleFormat sample_fmt)

{

    int planar      = av_sample_fmt_is_planar(sample_fmt);

    int planes      = planar ? nb_channels : 1;

    int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels);

    int data_size   = nb_samples * block_align;

    int i;

    dst_offset *= block_align;

    src_offset *= block_align;

    if((dst[0] < src[0] ? src[0] - dst[0] : dst[0] - src[0]) >= data_size) {

        for (i = 0; i < planes; i++)

            memcpy(dst[i] + dst_offset, src[i] + src_offset, data_size);

    } else {

        for (i = 0; i < planes; i++)

            memmove(dst[i] + dst_offset, src[i] + src_offset, data_size);

    }

    return 0;

}

int av_samples_set_silence(uint8_t * const *audio_data, int offset, int nb_samples,

                           int nb_channels, enum AVSampleFormat sample_fmt)

{

    int planar      = av_sample_fmt_is_planar(sample_fmt);

    int planes      = planar ? nb_channels : 1;

    int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels);

    int data_size   = nb_samples * block_align;

    int fill_char, i;

    if (sample_fmt == AV_SAMPLE_FMT_U8 || sample_fmt == AV_SAMPLE_FMT_U8P)

        fill_char = 0x80;

    else if (sample_fmt == AV_SAMPLE_FMT_DSD)

        fill_char = 0x69; // only ultrasonic tones, filtered out on playback

    else

        fill_char = 0x00;

    offset *= block_align;

    for (i = 0; i < planes; i++)

        memset(audio_data[i] + offset, fill_char, data_size);

    return 0;

}