/*

 * Audio FIFO

 * 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

 */

/**

 * @file

 * Audio FIFO

 */

#include <limits.h>

#include <stddef.h>

#include "audio_fifo.h"

#include "error.h"

#include "fifo.h"

#include "macros.h"

#include "mem.h"

#include "samplefmt.h"

struct AVAudioFifo {

    AVFifo **buf;                   /**< single buffer for interleaved, per-channel buffers for planar */

    int nb_buffers;                 /**< number of buffers */

    int nb_samples;                 /**< number of samples currently in the FIFO */

    int allocated_samples;          /**< current allocated size, in samples */

    int channels;                   /**< number of channels */

    enum AVSampleFormat sample_fmt; /**< sample format */

    int sample_size;                /**< size, in bytes, of one sample in a buffer */

};

void av_audio_fifo_free(AVAudioFifo *af)

{

    if (af) {

        if (af->buf) {

            int i;

            for (i = 0; i < af->nb_buffers; i++) {

                av_fifo_freep2(&af->buf[i]);

            }

            av_freep(&af->buf);

        }

        av_free(af);

    }

}

AVAudioFifo *av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels,

                                 int nb_samples)

{

    AVAudioFifo *af;

    int buf_size, i;

    /* get channel buffer size (also validates parameters) */

    if (av_samples_get_buffer_size(&buf_size, channels, nb_samples, sample_fmt, 1) < 0)

        return NULL;

    af = av_mallocz(sizeof(*af));

    if (!af)

        return NULL;

    af->channels    = channels;

    af->sample_fmt  = sample_fmt;

    af->sample_size = buf_size / nb_samples;

    af->nb_buffers  = av_sample_fmt_is_planar(sample_fmt) ? channels : 1;

    af->buf = av_calloc(af->nb_buffers, sizeof(*af->buf));

    if (!af->buf)

        goto error;

    for (i = 0; i < af->nb_buffers; i++) {

        af->buf[i] = av_fifo_alloc2(buf_size, 1, 0);

        if (!af->buf[i])

            goto error;

    }

    af->allocated_samples = nb_samples;

    return af;

error:

    av_audio_fifo_free(af);

    return NULL;

}

int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)

{

    const size_t cur_size = av_fifo_can_read (af->buf[0]) +

                            av_fifo_can_write(af->buf[0]);

    int i, ret, buf_size;

    if ((ret = av_samples_get_buffer_size(&buf_size, af->channels, nb_samples,

                                          af->sample_fmt, 1)) < 0)

        return ret;

    if (buf_size > cur_size) {

        for (i = 0; i < af->nb_buffers; i++) {

            if ((ret = av_fifo_grow2(af->buf[i], buf_size - cur_size)) < 0)

                return ret;

        }

    }

    af->allocated_samples = nb_samples;

    return 0;

}

int av_audio_fifo_write(AVAudioFifo *af, void * const *data, int nb_samples)

{

    int i, ret, size;

    /* automatically reallocate buffers if needed */

    if (av_audio_fifo_space(af) < nb_samples) {

        int current_size = av_audio_fifo_size(af);

        /* check for integer overflow in new size calculation */

        if (INT_MAX / 2 - current_size < nb_samples)

            return AVERROR(EINVAL);

        /* reallocate buffers */

        if ((ret = av_audio_fifo_realloc(af, 2 * (current_size + nb_samples))) < 0)

            return ret;

    }

    size = nb_samples * af->sample_size;

    for (i = 0; i < af->nb_buffers; i++) {

        ret = av_fifo_write(af->buf[i], data[i], size);

        if (ret < 0)

            return AVERROR_BUG;

    }

    af->nb_samples += nb_samples;

    return nb_samples;

}

int av_audio_fifo_peek(const AVAudioFifo *af, void * const *data, int nb_samples)

{

    return av_audio_fifo_peek_at(af, data, nb_samples, 0);

}

int av_audio_fifo_peek_at(const AVAudioFifo *af, void * const *data,

                          int nb_samples, int offset)

{

    int i, ret, size;

    if (offset < 0 || offset >= af->nb_samples)

        return AVERROR(EINVAL);

    if (nb_samples < 0)

        return AVERROR(EINVAL);

    nb_samples = FFMIN(nb_samples, af->nb_samples);

    if (!nb_samples)

        return 0;

    if (offset > af->nb_samples - nb_samples)

        return AVERROR(EINVAL);

    offset *= af->sample_size;

    size = nb_samples * af->sample_size;

    for (i = 0; i < af->nb_buffers; i++) {

        if ((ret = av_fifo_peek(af->buf[i], data[i], size, offset)) < 0)

            return AVERROR_BUG;

    }

    return nb_samples;

}

int av_audio_fifo_read(AVAudioFifo *af, void * const *data, int nb_samples)

{

    int i, size;

    if (nb_samples < 0)

        return AVERROR(EINVAL);

    nb_samples = FFMIN(nb_samples, af->nb_samples);

    if (!nb_samples)

        return 0;

    size = nb_samples * af->sample_size;

    for (i = 0; i < af->nb_buffers; i++) {

        if (av_fifo_read(af->buf[i], data[i], size) < 0)

            return AVERROR_BUG;

    }

    af->nb_samples -= nb_samples;

    return nb_samples;

}

int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)

{

    int i, size;

    if (nb_samples < 0)

        return AVERROR(EINVAL);

    nb_samples = FFMIN(nb_samples, af->nb_samples);

    if (nb_samples) {

        size = nb_samples * af->sample_size;

        for (i = 0; i < af->nb_buffers; i++)

            av_fifo_drain2(af->buf[i], size);

        af->nb_samples -= nb_samples;

    }

    return 0;

}

void av_audio_fifo_reset(AVAudioFifo *af)

{

    int i;

    for (i = 0; i < af->nb_buffers; i++)

        av_fifo_reset2(af->buf[i]);

    af->nb_samples = 0;

}

int av_audio_fifo_size(AVAudioFifo *af)

{

    return af->nb_samples;

}

int av_audio_fifo_space(AVAudioFifo *af)

{

    return af->allocated_samples - af->nb_samples;

}