/*
  * Copyright (c) 2017 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
  */
 
 /**
  * @file
  * An arbitrary audio FIR filter
  */
 
 #include "libavutil/audio_fifo.h"
 #include "libavutil/common.h"
 #include "libavutil/float_dsp.h"
 #include "libavutil/opt.h"
 #include "libavcodec/avfft.h"
 
 #include "audio.h"
 #include "avfilter.h"
 #include "formats.h"
 #include "internal.h"
 #include "af_afir.h"
 
 static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
 {
     int n;
 
     for (n = 0; n < len; n++) {
         const float cre = c[2 * n    ];
         const float cim = c[2 * n + 1];
         const float tre = t[2 * n    ];
         const float tim = t[2 * n + 1];
 
         sum[2 * n    ] += tre * cre - tim * cim;
         sum[2 * n + 1] += tre * cim + tim * cre;
     }
 
     sum[2 * n] += t[2 * n] * c[2 * n];
 }
 
 static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
 {
     AudioFIRContext *s = ctx->priv;
     const float *src = (const float *)s->in[0]->extended_data[ch];
     int index1 = (s->index + 1) % 3;
     int index2 = (s->index + 2) % 3;
     float *sum = s->sum[ch];
     AVFrame *out = arg;
     float *block;
     float *dst;
     int n, i, j;
 
     memset(sum, 0, sizeof(*sum) * s->fft_length);
     block = s->block[ch] + s->part_index * s->block_size;
     memset(block, 0, sizeof(*block) * s->fft_length);
 

     s->fdsp->vector_fmul_scalar(block + s->part_size, src, s->dry_gain, FFALIGN(s->nb_samples, 4));

     emms_c();
 
     av_rdft_calc(s->rdft[ch], block);
     block[2 * s->part_size] = block[1];
     block[1] = 0;
 
     j = s->part_index;
 
     for (i = 0; i < s->nb_partitions; i++) {
         const int coffset = i * s->coeff_size;
         const FFTComplex *coeff = s->coeff[ch * !s->one2many] + coffset;
 
         block = s->block[ch] + j * s->block_size;
         s->fcmul_add(sum, block, (const float *)coeff, s->part_size);
 
         if (j == 0)
             j = s->nb_partitions;
         j--;
     }
 
     sum[1] = sum[2 * s->part_size];
     av_rdft_calc(s->irdft[ch], sum);
 
     dst = (float *)s->buffer->extended_data[ch] + index1 * s->part_size;
     for (n = 0; n < s->part_size; n++) {
         dst[n] += sum[n];
     }
 
     dst = (float *)s->buffer->extended_data[ch] + index2 * s->part_size;
 
     memcpy(dst, sum + s->part_size, s->part_size * sizeof(*dst));
 
     dst = (float *)s->buffer->extended_data[ch] + s->index * s->part_size;
 
     if (out) {
         float *ptr = (float *)out->extended_data[ch];

         s->fdsp->vector_fmul_scalar(ptr, dst, s->gain * s->wet_gain, FFALIGN(out->nb_samples, 4));

         emms_c();
     }
 
     return 0;
 }
 
 static int fir_frame(AudioFIRContext *s, AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     AVFrame *out = NULL;
     int ret;
 
     s->nb_samples = FFMIN(s->part_size, av_audio_fifo_size(s->fifo[0]));
 
     if (!s->want_skip) {
         out = ff_get_audio_buffer(outlink, s->nb_samples);
         if (!out)
             return AVERROR(ENOMEM);
     }
 
     s->in[0] = ff_get_audio_buffer(ctx->inputs[0], s->nb_samples);
     if (!s->in[0]) {
         av_frame_free(&out);
         return AVERROR(ENOMEM);
     }
 
     av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data, s->nb_samples);
 
     ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
 
     s->part_index = (s->part_index + 1) % s->nb_partitions;
 
     av_audio_fifo_drain(s->fifo[0], s->nb_samples);
 
     if (!s->want_skip) {
         out->pts = s->pts;
         if (s->pts != AV_NOPTS_VALUE)
             s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
     }
 
     s->index++;
     if (s->index == 3)
         s->index = 0;
 
     av_frame_free(&s->in[0]);
 
     if (s->want_skip == 1) {
         s->want_skip = 0;
         ret = 0;
     } else {
         ret = ff_filter_frame(outlink, out);
     }
 
     return ret;
 }
 
 static int convert_coeffs(AVFilterContext *ctx)
 {
     AudioFIRContext *s = ctx->priv;
     int i, ch, n, N;
     float power = 0;
 
     s->nb_taps = av_audio_fifo_size(s->fifo[1]);
     if (s->nb_taps <= 0)
         return AVERROR(EINVAL);
 
     for (n = 4; (1 << n) < s->nb_taps; n++);
     N = FFMIN(n, 16);
     s->ir_length = 1 << n;
     s->fft_length = (1 << (N + 1)) + 1;
     s->part_size = 1 << (N - 1);
     s->block_size = FFALIGN(s->fft_length, 32);
     s->coeff_size = FFALIGN(s->part_size + 1, 32);
     s->nb_partitions = (s->nb_taps + s->part_size - 1) / s->part_size;
     s->nb_coeffs = s->ir_length + s->nb_partitions;
 
     for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
         s->sum[ch] = av_calloc(s->fft_length, sizeof(**s->sum));
         if (!s->sum[ch])
             return AVERROR(ENOMEM);
     }
 
     for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
         s->coeff[ch] = av_calloc(s->nb_partitions * s->coeff_size, sizeof(**s->coeff));
         if (!s->coeff[ch])
             return AVERROR(ENOMEM);
     }
 
     for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
         s->block[ch] = av_calloc(s->nb_partitions * s->block_size, sizeof(**s->block));
         if (!s->block[ch])
             return AVERROR(ENOMEM);
     }
 
     for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
         s->rdft[ch]  = av_rdft_init(N, DFT_R2C);
         s->irdft[ch] = av_rdft_init(N, IDFT_C2R);
         if (!s->rdft[ch] || !s->irdft[ch])
             return AVERROR(ENOMEM);
     }
 
     s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps);
     if (!s->in[1])
         return AVERROR(ENOMEM);
 
     s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size * 3);
     if (!s->buffer)
         return AVERROR(ENOMEM);
 
     av_audio_fifo_read(s->fifo[1], (void **)s->in[1]->extended_data, s->nb_taps);
 
     for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
         float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
         float *block = s->block[ch];
         FFTComplex *coeff = s->coeff[ch];
 
         power += s->fdsp->scalarproduct_float(time, time, s->nb_taps);
 
         for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
             time[i] = 0;
 
         for (i = 0; i < s->nb_partitions; i++) {
             const float scale = 1.f / s->part_size;
             const int toffset = i * s->part_size;
             const int coffset = i * s->coeff_size;
             const int boffset = s->part_size;
             const int remaining = s->nb_taps - (i * s->part_size);
             const int size = remaining >= s->part_size ? s->part_size : remaining;
 
             memset(block, 0, sizeof(*block) * s->fft_length);
             memcpy(block + boffset, time + toffset, size * sizeof(*block));
 
             av_rdft_calc(s->rdft[0], block);
 
             coeff[coffset].re = block[0] * scale;
             coeff[coffset].im = 0;
             for (n = 1; n < s->part_size; n++) {
                 coeff[coffset + n].re = block[2 * n] * scale;
                 coeff[coffset + n].im = block[2 * n + 1] * scale;
             }
             coeff[coffset + s->part_size].re = block[1] * scale;
             coeff[coffset + s->part_size].im = 0;
         }
     }
 
     av_frame_free(&s->in[1]);
     s->gain = s->again ? 1.f / sqrtf(power / ctx->inputs[1]->channels) : 1.f;
     av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
     av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions);
     av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->part_size);
     av_log(ctx, AV_LOG_DEBUG, "ir_length: %d\n", s->ir_length);
 
     s->have_coeffs = 1;
 
     return 0;
 }
 
 static int read_ir(AVFilterLink *link, AVFrame *frame)
 {
     AVFilterContext *ctx = link->dst;
     AudioFIRContext *s = ctx->priv;

     int nb_taps, max_nb_taps, ret;

     ret = av_audio_fifo_write(s->fifo[1], (void **)frame->extended_data,
                              frame->nb_samples);

     av_frame_free(&frame);

     if (ret < 0)
         return ret;

 
     nb_taps = av_audio_fifo_size(s->fifo[1]);
     max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate;
     if (nb_taps > max_nb_taps) {
         av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
         return AVERROR(EINVAL);
     }
 
     return 0;
 }
 
 static int filter_frame(AVFilterLink *link, AVFrame *frame)
 {
     AVFilterContext *ctx = link->dst;
     AudioFIRContext *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];

     int ret;

     ret = av_audio_fifo_write(s->fifo[0], (void **)frame->extended_data,
                               frame->nb_samples);
     if (ret > 0 && s->pts == AV_NOPTS_VALUE)

         s->pts = frame->pts;
 
     av_frame_free(&frame);
 

     if (ret < 0)
         return ret;
 

     if (!s->have_coeffs && s->eof_coeffs) {
         ret = convert_coeffs(ctx);
         if (ret < 0)
             return ret;
     }
 
     if (s->have_coeffs) {
         while (av_audio_fifo_size(s->fifo[0]) >= s->part_size) {
             ret = fir_frame(s, outlink);
             if (ret < 0)

                 return ret;

         }
     }

     return 0;

 }
 
 static int request_frame(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     AudioFIRContext *s = ctx->priv;
     int ret;
 
     if (!s->eof_coeffs) {
         ret = ff_request_frame(ctx->inputs[1]);
         if (ret == AVERROR_EOF) {
             s->eof_coeffs = 1;
             ret = 0;
         }
         return ret;
     }
     ret = ff_request_frame(ctx->inputs[0]);
     if (ret == AVERROR_EOF && s->have_coeffs) {
         if (s->need_padding) {
             AVFrame *silence = ff_get_audio_buffer(outlink, s->part_size);
 
             if (!silence)
                 return AVERROR(ENOMEM);

             ret = av_audio_fifo_write(s->fifo[0], (void **)silence->extended_data,
                                       silence->nb_samples);

             av_frame_free(&silence);

             if (ret < 0)
                 return ret;

             s->need_padding = 0;
         }
 
         while (av_audio_fifo_size(s->fifo[0]) > 0) {
             ret = fir_frame(s, outlink);
             if (ret < 0)
                 return ret;
         }
         ret = AVERROR_EOF;
     }
     return ret;
 }
 
 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterFormats *formats;
     AVFilterChannelLayouts *layouts;
     static const enum AVSampleFormat sample_fmts[] = {
         AV_SAMPLE_FMT_FLTP,
         AV_SAMPLE_FMT_NONE
     };
     int ret, i;
 
     layouts = ff_all_channel_counts();
     if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
         return ret;
 
     for (i = 0; i < 2; i++) {
         layouts = ff_all_channel_counts();
         if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
             return ret;
     }
 
     formats = ff_make_format_list(sample_fmts);
     if ((ret = ff_set_common_formats(ctx, formats)) < 0)
         return ret;
 
     formats = ff_all_samplerates();
     return ff_set_common_samplerates(ctx, formats);
 }
 
 static int config_output(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     AudioFIRContext *s = ctx->priv;
 
     if (ctx->inputs[0]->channels != ctx->inputs[1]->channels &&
         ctx->inputs[1]->channels != 1) {
         av_log(ctx, AV_LOG_ERROR,
                "Second input must have same number of channels as first input or "
                "exactly 1 channel.\n");
         return AVERROR(EINVAL);
     }
 
     s->one2many = ctx->inputs[1]->channels == 1;
     outlink->sample_rate = ctx->inputs[0]->sample_rate;
     outlink->time_base   = ctx->inputs[0]->time_base;
     outlink->channel_layout = ctx->inputs[0]->channel_layout;
     outlink->channels = ctx->inputs[0]->channels;
 
     s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
     s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
     if (!s->fifo[0] || !s->fifo[1])
         return AVERROR(ENOMEM);
 
     s->sum = av_calloc(outlink->channels, sizeof(*s->sum));
     s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff));
     s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block));
     s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft));
     s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft));
     if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft)
         return AVERROR(ENOMEM);
 
     s->nb_channels = outlink->channels;
     s->nb_coef_channels = ctx->inputs[1]->channels;
     s->want_skip = 1;
     s->need_padding = 1;
     s->pts = AV_NOPTS_VALUE;
 
     return 0;
 }
 
 static av_cold void uninit(AVFilterContext *ctx)
 {
     AudioFIRContext *s = ctx->priv;
     int ch;
 
     if (s->sum) {
         for (ch = 0; ch < s->nb_channels; ch++) {
             av_freep(&s->sum[ch]);
         }
     }
     av_freep(&s->sum);
 
     if (s->coeff) {
         for (ch = 0; ch < s->nb_coef_channels; ch++) {
             av_freep(&s->coeff[ch]);
         }
     }
     av_freep(&s->coeff);
 
     if (s->block) {
         for (ch = 0; ch < s->nb_channels; ch++) {
             av_freep(&s->block[ch]);
         }
     }
     av_freep(&s->block);
 
     if (s->rdft) {
         for (ch = 0; ch < s->nb_channels; ch++) {
             av_rdft_end(s->rdft[ch]);
         }
     }
     av_freep(&s->rdft);
 
     if (s->irdft) {
         for (ch = 0; ch < s->nb_channels; ch++) {
             av_rdft_end(s->irdft[ch]);
         }
     }
     av_freep(&s->irdft);
 
     av_frame_free(&s->in[0]);
     av_frame_free(&s->in[1]);
     av_frame_free(&s->buffer);
 
     av_audio_fifo_free(s->fifo[0]);
     av_audio_fifo_free(s->fifo[1]);
 
     av_freep(&s->fdsp);
 }
 
 static av_cold int init(AVFilterContext *ctx)
 {
     AudioFIRContext *s = ctx->priv;
 
     s->fcmul_add = fcmul_add_c;
 
     s->fdsp = avpriv_float_dsp_alloc(0);
     if (!s->fdsp)
         return AVERROR(ENOMEM);
 
     if (ARCH_X86)
         ff_afir_init_x86(s);
 
     return 0;
 }
 
 static const AVFilterPad afir_inputs[] = {
     {
         .name           = "main",
         .type           = AVMEDIA_TYPE_AUDIO,
         .filter_frame   = filter_frame,
     },{
         .name           = "ir",
         .type           = AVMEDIA_TYPE_AUDIO,
         .filter_frame   = read_ir,
     },
     { NULL }
 };
 
 static const AVFilterPad afir_outputs[] = {
     {
         .name          = "default",
         .type          = AVMEDIA_TYPE_AUDIO,
         .config_props  = config_output,
         .request_frame = request_frame,
     },
     { NULL }
 };
 
 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
 #define OFFSET(x) offsetof(AudioFIRContext, x)
 
 static const AVOption afir_options[] = {
     { "dry",    "set dry gain",     OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
     { "wet",    "set wet gain",     OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
     { "length", "set IR length",    OFFSET(length),   AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
     { "again",  "enable auto gain", OFFSET(again),    AV_OPT_TYPE_BOOL,  {.i64=1}, 0, 1, AF },
     { NULL }
 };
 
 AVFILTER_DEFINE_CLASS(afir);
 
 AVFilter ff_af_afir = {
     .name          = "afir",
     .description   = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
     .priv_size     = sizeof(AudioFIRContext),
     .priv_class    = &afir_class,
     .query_formats = query_formats,
     .init          = init,
     .uninit        = uninit,
     .inputs        = afir_inputs,
     .outputs       = afir_outputs,
     .flags         = AVFILTER_FLAG_SLICE_THREADS,
 };