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/*

 * Audio Mix Filter

 * Copyright (c) 2012 Justin Ruggles 

 *

 * 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 Mix Filter

 *

 * Mixes audio from multiple sources into a single output. The channel layout,

 * sample rate, and sample format will be the same for all inputs and the

 * output.

 */

#include "libavutil/attributes.h"

#include "libavutil/audio_fifo.h"

#include "libavutil/avassert.h"

#include "libavutil/avstring.h"

#include "libavutil/channel_layout.h"

#include "libavutil/common.h"

#include "libavutil/float_dsp.h"

#include "libavutil/mathematics.h"

#include "libavutil/opt.h"

#include "libavutil/samplefmt.h"

#include "audio.h"

#include "avfilter.h"

#include "formats.h"

#include "internal.h"

#define INPUT_OFF      0    /**< input has reached EOF */

#define INPUT_ON       1    /**< input is active */

#define INPUT_INACTIVE 2    /**< input is on, but is currently inactive */

#define DURATION_LONGEST  0

#define DURATION_SHORTEST 1

#define DURATION_FIRST    2

typedef struct FrameInfo {

    int nb_samples;

    int64_t pts;

    struct FrameInfo *next;

} FrameInfo;

/**

 * Linked list used to store timestamps and frame sizes of all frames in the

 * FIFO for the first input.

 *

 * This is needed to keep timestamps synchronized for the case where multiple

 * input frames are pushed to the filter for processing before a frame is

 * requested by the output link.

 */

typedef struct FrameList {

    int nb_frames;

    int nb_samples;

    FrameInfo *list;

    FrameInfo *end;

} FrameList;

static void frame_list_clear(FrameList *frame_list)

{

    if (frame_list) {

        while (frame_list->list) {

            FrameInfo *info = frame_list->list;

            frame_list->list = info->next;

            av_free(info);

        }

        frame_list->nb_frames  = 0;

        frame_list->nb_samples = 0;

        frame_list->end        = NULL;

    }

}

static int frame_list_next_frame_size(FrameList *frame_list)

{

    if (!frame_list->list)

        return 0;

    return frame_list->list->nb_samples;

}

static int64_t frame_list_next_pts(FrameList *frame_list)

{

    if (!frame_list->list)

        return AV_NOPTS_VALUE;

    return frame_list->list->pts;

}

static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)

{

    if (nb_samples >= frame_list->nb_samples) {

        frame_list_clear(frame_list);

    } else {

        int samples = nb_samples;

        while (samples > 0) {

            FrameInfo *info = frame_list->list;

            av_assert0(info != NULL);

            if (info->nb_samples <= samples) {

                samples -= info->nb_samples;

                frame_list->list = info->next;

                if (!frame_list->list)

                    frame_list->end = NULL;

                frame_list->nb_frames--;

                frame_list->nb_samples -= info->nb_samples;

                av_free(info);

            } else {

                info->nb_samples       -= samples;

                info->pts              += samples;

                frame_list->nb_samples -= samples;

                samples = 0;

            }

        }

    }

}

static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)

{

    FrameInfo *info = av_malloc(sizeof(*info));

    if (!info)

        return AVERROR(ENOMEM);

    info->nb_samples = nb_samples;

    info->pts        = pts;

    info->next       = NULL;

    if (!frame_list->list) {

        frame_list->list = info;

        frame_list->end  = info;

    } else {

        av_assert0(frame_list->end != NULL);

        frame_list->end->next = info;

        frame_list->end       = info;

    }

    frame_list->nb_frames++;

    frame_list->nb_samples += nb_samples;

    return 0;

}

typedef struct MixContext {

    const AVClass *class;       /**< class for AVOptions */

    AVFloatDSPContext fdsp;

    int nb_inputs;              /**< number of inputs */

    int active_inputs;          /**< number of input currently active */

    int duration_mode;          /**< mode for determining duration */

    float dropout_transition;   /**< transition time when an input drops out */

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

    int sample_rate;            /**< sample rate */

    int planar;

    AVAudioFifo **fifos;        /**< audio fifo for each input */

    uint8_t *input_state;       /**< current state of each input */

    float *input_scale;         /**< mixing scale factor for each input */

    float scale_norm;           /**< normalization factor for all inputs */

    int64_t next_pts;           /**< calculated pts for next output frame */

    FrameList *frame_list;      /**< list of frame info for the first input */

} MixContext;

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

#define A AV_OPT_FLAG_AUDIO_PARAM

#define F AV_OPT_FLAG_FILTERING_PARAM

static const AVOption amix_options[] = {

    { "inputs", "Number of inputs.",

            OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, 32, A|F },

    { "duration", "How to determine the end-of-stream.",

            OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0,  2, A|F, "duration" },

        { "longest",  "Duration of longest input.",  0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST  }, INT_MIN, INT_MAX, A|F, "duration" },

        { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, INT_MIN, INT_MAX, A|F, "duration" },

        { "first",    "Duration of first input.",    0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST    }, INT_MIN, INT_MAX, A|F, "duration" },

    { "dropout_transition", "Transition time, in seconds, for volume "

                            "renormalization when an input stream ends.",

            OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A|F },

    { NULL }

};

AVFILTER_DEFINE_CLASS(amix);

/**

 * Update the scaling factors to apply to each input during mixing.

 *

 * This balances the full volume range between active inputs and handles

 * volume transitions when EOF is encountered on an input but mixing continues

 * with the remaining inputs.

 */

static void calculate_scales(MixContext *s, int nb_samples)

{

    int i;

    if (s->scale_norm > s->active_inputs) {

        s->scale_norm -= nb_samples / (s->dropout_transition * s->sample_rate);

        s->scale_norm = FFMAX(s->scale_norm, s->active_inputs);

    }

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

        if (s->input_state[i] == INPUT_ON)

            s->input_scale[i] = 1.0f / s->scale_norm;

        else

            s->input_scale[i] = 0.0f;

    }

}

static int config_output(AVFilterLink *outlink)

{

    AVFilterContext *ctx = outlink->src;

    MixContext *s      = ctx->priv;

    int i;

    char buf[64];

    s->planar          = av_sample_fmt_is_planar(outlink->format);

    s->sample_rate     = outlink->sample_rate;

    outlink->time_base = (AVRational){ 1, outlink->sample_rate };

    s->next_pts        = AV_NOPTS_VALUE;

    s->frame_list = av_mallocz(sizeof(*s->frame_list));

    if (!s->frame_list)

        return AVERROR(ENOMEM);

    s->fifos = av_mallocz(s->nb_inputs * sizeof(*s->fifos));

    if (!s->fifos)

        return AVERROR(ENOMEM);

    s->nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);

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

        s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);

        if (!s->fifos[i])

            return AVERROR(ENOMEM);

    }

    s->input_state = av_malloc(s->nb_inputs);

    if (!s->input_state)

        return AVERROR(ENOMEM);

    memset(s->input_state, INPUT_ON, s->nb_inputs);

    s->active_inputs = s->nb_inputs;

    s->input_scale = av_mallocz(s->nb_inputs * sizeof(*s->input_scale));

    if (!s->input_scale)

        return AVERROR(ENOMEM);

    s->scale_norm = s->active_inputs;

    calculate_scales(s, 0);

    av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout);

    av_log(ctx, AV_LOG_VERBOSE,

           "inputs:%d fmt:%s srate:%d cl:%s\n", s->nb_inputs,

           av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);

    return 0;

}

/**

 * Read samples from the input FIFOs, mix, and write to the output link.

 */

static int output_frame(AVFilterLink *outlink, int nb_samples)

{

    AVFilterContext *ctx = outlink->src;

    MixContext      *s = ctx->priv;

    AVFrame *out_buf, *in_buf;

    int i;

    calculate_scales(s, nb_samples);

    out_buf = ff_get_audio_buffer(outlink, nb_samples);

    if (!out_buf)

        return AVERROR(ENOMEM);

    in_buf = ff_get_audio_buffer(outlink, nb_samples);

    if (!in_buf) {

        av_frame_free(&out_buf);

        return AVERROR(ENOMEM);

    }

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

        if (s->input_state[i] == INPUT_ON) {

            int planes, plane_size, p;

            av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,

                               nb_samples);

            planes     = s->planar ? s->nb_channels : 1;

            plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);

            plane_size = FFALIGN(plane_size, 16);

            for (p = 0; p < planes; p++) {

                s->fdsp.vector_fmac_scalar((float *)out_buf->extended_data[p],

                                           (float *) in_buf->extended_data[p],

                                           s->input_scale[i], plane_size);

            }

        }

    }

    av_frame_free(&in_buf);

    out_buf->pts = s->next_pts;

    if (s->next_pts != AV_NOPTS_VALUE)

        s->next_pts += nb_samples;

    return ff_filter_frame(outlink, out_buf);

}

/**

 * Returns the smallest number of samples available in the input FIFOs other

 * than that of the first input.

 */

static int get_available_samples(MixContext *s)

{

    int i;

    int available_samples = INT_MAX;

    av_assert0(s->nb_inputs > 1);

    for (i = 1; i < s->nb_inputs; i++) {

        int nb_samples;

        if (s->input_state[i] == INPUT_OFF)

            continue;

        nb_samples = av_audio_fifo_size(s->fifos[i]);

        available_samples = FFMIN(available_samples, nb_samples);

    }

    if (available_samples == INT_MAX)

        return 0;

    return available_samples;

}

/**

 * Requests a frame, if needed, from each input link other than the first.

 */

static int request_samples(AVFilterContext *ctx, int min_samples)

{

    MixContext *s = ctx->priv;

    int i, ret;

    av_assert0(s->nb_inputs > 1);

    for (i = 1; i < s->nb_inputs; i++) {

        ret = 0;

        if (s->input_state[i] == INPUT_OFF)

            continue;

        while (!ret && av_audio_fifo_size(s->fifos[i]) < min_samples)

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

        if (ret == AVERROR_EOF) {

            if (av_audio_fifo_size(s->fifos[i]) == 0) {

                s->input_state[i] = INPUT_OFF;

                continue;

            }

        } else if (ret < 0)

            return ret;

    }

    return 0;

}

/**

 * Calculates the number of active inputs and determines EOF based on the

 * duration option.

 *

 * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.

 */

static int calc_active_inputs(MixContext *s)

{

    int i;

    int active_inputs = 0;

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

        active_inputs += !!(s->input_state[i] != INPUT_OFF);

    s->active_inputs = active_inputs;

    if (!active_inputs ||

        (s->duration_mode == DURATION_FIRST && s->input_state[0] == INPUT_OFF) ||

        (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))

        return AVERROR_EOF;

    return 0;

}

static int request_frame(AVFilterLink *outlink)

{

    AVFilterContext *ctx = outlink->src;

    MixContext      *s = ctx->priv;

    int ret;

    int wanted_samples, available_samples;

    ret = calc_active_inputs(s);

    if (ret < 0)

        return ret;

    if (s->input_state[0] == INPUT_OFF) {

        ret = request_samples(ctx, 1);

        if (ret < 0)

            return ret;

        ret = calc_active_inputs(s);

        if (ret < 0)

            return ret;

        available_samples = get_available_samples(s);

        if (!available_samples)

            return AVERROR(EAGAIN);

        return output_frame(outlink, available_samples);

    }

    if (s->frame_list->nb_frames == 0) {

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

        if (ret == AVERROR_EOF) {

            s->input_state[0] = INPUT_OFF;

            if (s->nb_inputs == 1)

                return AVERROR_EOF;

            else

                return AVERROR(EAGAIN);

        } else if (ret < 0)

            return ret;

    }

    av_assert0(s->frame_list->nb_frames > 0);

    wanted_samples = frame_list_next_frame_size(s->frame_list);

    if (s->active_inputs > 1) {

        ret = request_samples(ctx, wanted_samples);

        if (ret < 0)

            return ret;

        ret = calc_active_inputs(s);

        if (ret < 0)

            return ret;

    }

    if (s->active_inputs > 1) {

        available_samples = get_available_samples(s);

        if (!available_samples)

            return AVERROR(EAGAIN);

        available_samples = FFMIN(available_samples, wanted_samples);

    } else {

        available_samples = wanted_samples;

    }

    s->next_pts = frame_list_next_pts(s->frame_list);

    frame_list_remove_samples(s->frame_list, available_samples);

    return output_frame(outlink, available_samples);

}

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

{

    AVFilterContext  *ctx = inlink->dst;

    MixContext       *s = ctx->priv;

    AVFilterLink *outlink = ctx->outputs[0];

    int i, ret = 0;

    for (i = 0; i < ctx->nb_inputs; i++)

        if (ctx->inputs[i] == inlink)

            break;

    if (i >= ctx->nb_inputs) {

        av_log(ctx, AV_LOG_ERROR, "unknown input link\n");

        ret = AVERROR(EINVAL);

        goto fail;

    }

    if (i == 0) {

        int64_t pts = av_rescale_q(buf->pts, inlink->time_base,

                                   outlink->time_base);

        ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);

        if (ret < 0)

            goto fail;

    }

    ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,

                              buf->nb_samples);

fail:

    av_frame_free(&buf);

    return ret;

}

static av_cold int init(AVFilterContext *ctx)

{

    MixContext *s = ctx->priv;

    int i;

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

        char name[32];

        AVFilterPad pad = { 0 };

        snprintf(name, sizeof(name), "input%d", i);

        pad.type           = AVMEDIA_TYPE_AUDIO;

        pad.name           = av_strdup(name);

        pad.filter_frame   = filter_frame;

        ff_insert_inpad(ctx, i, &pad);

    }

    avpriv_float_dsp_init(&s->fdsp, 0);

    return 0;

}

static av_cold void uninit(AVFilterContext *ctx)

{

    int i;

    MixContext *s = ctx->priv;

    if (s->fifos) {

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

            av_audio_fifo_free(s->fifos[i]);

        av_freep(&s->fifos);

    }

    frame_list_clear(s->frame_list);

    av_freep(&s->frame_list);

    av_freep(&s->input_state);

    av_freep(&s->input_scale);

    for (i = 0; i < ctx->nb_inputs; i++)

        av_freep(&ctx->input_pads[i].name);

}

static int query_formats(AVFilterContext *ctx)

{

    AVFilterFormats *formats = NULL;

    ff_add_format(&formats, AV_SAMPLE_FMT_FLT);

    ff_add_format(&formats, AV_SAMPLE_FMT_FLTP);

    ff_set_common_formats(ctx, formats);

    ff_set_common_channel_layouts(ctx, ff_all_channel_layouts());

    ff_set_common_samplerates(ctx, ff_all_samplerates());

    return 0;

}

static const AVFilterPad avfilter_af_amix_outputs[] = {

    {

        .name          = "default",

        .type          = AVMEDIA_TYPE_AUDIO,

        .config_props  = config_output,

        .request_frame = request_frame

    },

    { NULL }

};

AVFilter avfilter_af_amix = {

    .name           = "amix",

    .description    = NULL_IF_CONFIG_SMALL("Audio mixing."),

    .priv_size      = sizeof(MixContext),

    .priv_class     = &amix_class,

    .init           = init,

    .uninit         = uninit,

    .query_formats  = query_formats,

    .inputs         = NULL,

    .outputs        = avfilter_af_amix_outputs,

    .flags          = AVFILTER_FLAG_DYNAMIC_INPUTS,

};