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/*

 * Copyright (c) 2013 Paul B Mahol

 *

 * 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 "libavutil/avstring.h"

#include "libavutil/opt.h"

#include "libavutil/samplefmt.h"

#include "avfilter.h"

#include "audio.h"

#include "internal.h"

typedef struct ChanDelay {

    int delay;

    unsigned delay_index;

    unsigned index;

    uint8_t *samples;

} ChanDelay;

typedef struct AudioDelayContext {

    const AVClass *class;

    char *delays;

    ChanDelay *chandelay;

    int nb_delays;

    int block_align;

    unsigned max_delay;

    int64_t next_pts;

    void (*delay_channel)(ChanDelay *d, int nb_samples,

                          const uint8_t *src, uint8_t *dst);

} AudioDelayContext;

#define OFFSET(x) offsetof(AudioDelayContext, x)

#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

static const AVOption adelay_options[] = {

    { "delays", "set list of delays for each channel", OFFSET(delays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },

    { NULL }

};

AVFILTER_DEFINE_CLASS(adelay);

static int query_formats(AVFilterContext *ctx)

{

    AVFilterChannelLayouts *layouts;

    AVFilterFormats *formats;

    static const enum AVSampleFormat sample_fmts[] = {

        AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,

        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,

        AV_SAMPLE_FMT_NONE

    };

    int ret;

    layouts = ff_all_channel_layouts();

    if (!layouts)

        return AVERROR(ENOMEM);

    ret = ff_set_common_channel_layouts(ctx, layouts);

    if (ret < 0)

        return ret;

    formats = ff_make_format_list(sample_fmts);

    if (!formats)

        return AVERROR(ENOMEM);

    ret = ff_set_common_formats(ctx, formats);

    if (ret < 0)

        return ret;

    formats = ff_all_samplerates();

    if (!formats)

        return AVERROR(ENOMEM);

    return ff_set_common_samplerates(ctx, formats);

}

#define DELAY(name, type, fill)                                           \

static void delay_channel_## name ##p(ChanDelay *d, int nb_samples,       \

                                      const uint8_t *ssrc, uint8_t *ddst) \

{                                                                         \

    const type *src = (type *)ssrc;                                       \

    type *dst = (type *)ddst;                                             \

    type *samples = (type *)d->samples;                                   \

                                                                          \

    while (nb_samples) {                                                  \

        if (d->delay_index < d->delay) {                                  \

            const int len = FFMIN(nb_samples, d->delay - d->delay_index); \

                                                                          \

            memcpy(&samples[d->delay_index], src, len * sizeof(type));    \

            memset(dst, fill, len * sizeof(type));                        \

            d->delay_index += len;                                        \

            src += len;                                                   \

            dst += len;                                                   \

            nb_samples -= len;                                            \

        } else {                                                          \

            *dst = samples[d->index];                                     \

            samples[d->index] = *src;                                     \

            nb_samples--;                                                 \

            d->index++;                                                   \

            src++, dst++;                                                 \

            d->index = d->index >= d->delay ? 0 : d->index;               \

        }                                                                 \

    }                                                                     \

}

DELAY(u8,  uint8_t, 0x80)

DELAY(s16, int16_t, 0)

DELAY(s32, int32_t, 0)

DELAY(flt, float,   0)

DELAY(dbl, double,  0)

static int config_input(AVFilterLink *inlink)

{

    AVFilterContext *ctx = inlink->dst;

    AudioDelayContext *s = ctx->priv;

    char *p, *arg, *saveptr = NULL;

    int i;

    s->chandelay = av_calloc(inlink->channels, sizeof(*s->chandelay));

    if (!s->chandelay)

        return AVERROR(ENOMEM);

    s->nb_delays = inlink->channels;

    s->block_align = av_get_bytes_per_sample(inlink->format);

    p = s->delays;

    for (i = 0; i < s->nb_delays; i++) {

        ChanDelay *d = &s->chandelay[i];

        float delay;

        if (!(arg = av_strtok(p, "|", &saveptr)))

            break;

        p = NULL;

        sscanf(arg, "%f", &delay);

        d->delay = delay * inlink->sample_rate / 1000.0;

        if (d->delay < 0) {

            av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n");

            return AVERROR(EINVAL);

        }

    }

    for (i = 0; i < s->nb_delays; i++) {

        ChanDelay *d = &s->chandelay[i];

        if (!d->delay)

            continue;

        d->samples = av_malloc_array(d->delay, s->block_align);

        if (!d->samples)

            return AVERROR(ENOMEM);

        s->max_delay = FFMAX(s->max_delay, d->delay);

    }

    if (!s->max_delay) {

        av_log(ctx, AV_LOG_ERROR, "At least one delay >0 must be specified.\n");

        return AVERROR(EINVAL);

    }

    switch (inlink->format) {

    case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ; break;

    case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p; break;

    case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p; break;

    case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp; break;

    case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp; break;

    }

    return 0;

}

static int filter_frame(AVFilterLink *inlink, AVFrame *frame)

{

    AVFilterContext *ctx = inlink->dst;

    AudioDelayContext *s = ctx->priv;

    AVFrame *out_frame;

    int i;

    if (ctx->is_disabled || !s->delays)

        return ff_filter_frame(ctx->outputs[0], frame);

    out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);

    if (!out_frame)

        return AVERROR(ENOMEM);

    av_frame_copy_props(out_frame, frame);

    for (i = 0; i < s->nb_delays; i++) {

        ChanDelay *d = &s->chandelay[i];

        const uint8_t *src = frame->extended_data[i];

        uint8_t *dst = out_frame->extended_data[i];

        if (!d->delay)

            memcpy(dst, src, frame->nb_samples * s->block_align);

        else

            s->delay_channel(d, frame->nb_samples, src, dst);

    }

    s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);

    av_frame_free(&frame);

    return ff_filter_frame(ctx->outputs[0], out_frame);

}

static int request_frame(AVFilterLink *outlink)

{

    AVFilterContext *ctx = outlink->src;

    AudioDelayContext *s = ctx->priv;

    int ret;

    ret = ff_request_frame(ctx->inputs[0]);

    if (ret == AVERROR_EOF && !ctx->is_disabled && s->max_delay) {

        int nb_samples = FFMIN(s->max_delay, 2048);

        AVFrame *frame;

        frame = ff_get_audio_buffer(outlink, nb_samples);

        if (!frame)

            return AVERROR(ENOMEM);

        s->max_delay -= nb_samples;

        av_samples_set_silence(frame->extended_data, 0,

                               frame->nb_samples,

                               outlink->channels,

                               frame->format);

        frame->pts = s->next_pts;

        if (s->next_pts != AV_NOPTS_VALUE)

            s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);

        ret = filter_frame(ctx->inputs[0], frame);

    }

    return ret;

}

static av_cold void uninit(AVFilterContext *ctx)

{

    AudioDelayContext *s = ctx->priv;

    int i;

    for (i = 0; i < s->nb_delays; i++)

        av_freep(&s->chandelay[i].samples);

    av_freep(&s->chandelay);

}

static const AVFilterPad adelay_inputs[] = {

    {

        .name         = "default",

        .type         = AVMEDIA_TYPE_AUDIO,

        .config_props = config_input,

        .filter_frame = filter_frame,

    },

    { NULL }

};

static const AVFilterPad adelay_outputs[] = {

    {

        .name          = "default",

        .request_frame = request_frame,

        .type          = AVMEDIA_TYPE_AUDIO,

    },

    { NULL }

};

AVFilter ff_af_adelay = {

    .name          = "adelay",

    .description   = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),

    .query_formats = query_formats,

    .priv_size     = sizeof(AudioDelayContext),

    .priv_class    = &adelay_class,

    .uninit        = uninit,

    .inputs        = adelay_inputs,

    .outputs       = adelay_outputs,

    .flags         = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,

};