/*
  * Copyright (C) 2017 Paul B Mahol
  * Copyright (C) 2013-2015 Andreas Fuchs, Wolfgang Hrauda
  * 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 <math.h>
 
 #include "libavutil/audio_fifo.h"
 #include "libavutil/avstring.h"
 #include "libavutil/channel_layout.h"
 #include "libavutil/float_dsp.h"
 #include "libavutil/intmath.h"
 #include "libavutil/opt.h"
 #include "libavcodec/avfft.h"
 
 #include "avfilter.h"
 #include "internal.h"
 #include "audio.h"
 
 #define TIME_DOMAIN      0
 #define FREQUENCY_DOMAIN 1
 
 typedef struct HeadphoneContext {
     const AVClass *class;
 
     char *map;
     int type;
 
     int lfe_channel;
 
     int have_hrirs;
     int eof_hrirs;
     int64_t pts;
 
     int ir_len;
 
     int mapping[64];
 
     int nb_inputs;
 
     int nb_irs;
 
     float gain;
     float lfe_gain, gain_lfe;
 
     float *ringbuffer[2];
     int write[2];
 
     int buffer_length;
     int n_fft;
     int size;
 
     int *delay[2];
     float *data_ir[2];
     float *temp_src[2];
     FFTComplex *temp_fft[2];
 
     FFTContext *fft[2], *ifft[2];
     FFTComplex *data_hrtf[2];
 
     AVFloatDSPContext *fdsp;
     struct headphone_inputs {
         AVAudioFifo *fifo;
         AVFrame     *frame;
         int          ir_len;
         int          delay_l;
         int          delay_r;
         int          eof;
     } *in;
 } HeadphoneContext;
 
 static int parse_channel_name(HeadphoneContext *s, int x, char **arg, int *rchannel, char *buf)
 {
     int len, i, channel_id = 0;
     int64_t layout, layout0;
 
     if (sscanf(*arg, "%7[A-Z]%n", buf, &len)) {
         layout0 = layout = av_get_channel_layout(buf);
         if (layout == AV_CH_LOW_FREQUENCY)
             s->lfe_channel = x;
         for (i = 32; i > 0; i >>= 1) {
             if (layout >= 1LL << i) {
                 channel_id += i;
                 layout >>= i;
             }
         }
         if (channel_id >= 64 || layout0 != 1LL << channel_id)
             return AVERROR(EINVAL);
         *rchannel = channel_id;
         *arg += len;
         return 0;
     }
     return AVERROR(EINVAL);
 }
 
 static void parse_map(AVFilterContext *ctx)
 {
     HeadphoneContext *s = ctx->priv;
     char *arg, *tokenizer, *p, *args = av_strdup(s->map);
     int i;
 
     if (!args)
         return;
     p = args;
 
     s->lfe_channel = -1;
     s->nb_inputs = 1;
 
     for (i = 0; i < 64; i++) {
         s->mapping[i] = -1;
     }
 
     while ((arg = av_strtok(p, "|", &tokenizer))) {
         int out_ch_id;
         char buf[8];
 
         p = NULL;
         if (parse_channel_name(s, s->nb_inputs - 1, &arg, &out_ch_id, buf)) {
             av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%s\' as channel name.\n", buf);
             continue;
         }
         s->mapping[s->nb_inputs - 1] = out_ch_id;
         s->nb_inputs++;
     }
     s->nb_irs = s->nb_inputs - 1;
 
     av_free(args);
 }
 
 typedef struct ThreadData {
     AVFrame *in, *out;
     int *write;
     int **delay;
     float **ir;
     int *n_clippings;
     float **ringbuffer;
     float **temp_src;
     FFTComplex **temp_fft;
 } ThreadData;
 
 static int headphone_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
 {
     HeadphoneContext *s = ctx->priv;
     ThreadData *td = arg;
     AVFrame *in = td->in, *out = td->out;
     int offset = jobnr;
     int *write = &td->write[jobnr];
     const int *const delay = td->delay[jobnr];
     const float *const ir = td->ir[jobnr];
     int *n_clippings = &td->n_clippings[jobnr];
     float *ringbuffer = td->ringbuffer[jobnr];
     float *temp_src = td->temp_src[jobnr];
     const int ir_len = s->ir_len;
     const float *src = (const float *)in->data[0];
     float *dst = (float *)out->data[0];
     const int in_channels = in->channels;
     const int buffer_length = s->buffer_length;
     const uint32_t modulo = (uint32_t)buffer_length - 1;
     float *buffer[16];
     int wr = *write;
     int read;
     int i, l;
 
     dst += offset;
     for (l = 0; l < in_channels; l++) {
         buffer[l] = ringbuffer + l * buffer_length;
     }
 
     for (i = 0; i < in->nb_samples; i++) {
         const float *temp_ir = ir;
 
         *dst = 0;
         for (l = 0; l < in_channels; l++) {
             *(buffer[l] + wr) = src[l];
         }
 
         for (l = 0; l < in_channels; l++) {
             const float *const bptr = buffer[l];
 
             if (l == s->lfe_channel) {
                 *dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;
                 temp_ir += FFALIGN(ir_len, 16);
                 continue;
             }
 
             read = (wr - *(delay + l) - (ir_len - 1) + buffer_length) & modulo;
 
             if (read + ir_len < buffer_length) {
                 memcpy(temp_src, bptr + read, ir_len * sizeof(*temp_src));
             } else {
                 int len = FFMIN(ir_len - (read % ir_len), buffer_length - read);
 
                 memcpy(temp_src, bptr + read, len * sizeof(*temp_src));
                 memcpy(temp_src + len, bptr, (ir_len - len) * sizeof(*temp_src));
             }
 
             dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, ir_len);
             temp_ir += FFALIGN(ir_len, 16);
         }
 
         if (fabs(*dst) > 1)
             *n_clippings += 1;
 
         dst += 2;
         src += in_channels;
         wr   = (wr + 1) & modulo;
     }
 
     *write = wr;
 
     return 0;
 }
 
 static int headphone_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
 {
     HeadphoneContext *s = ctx->priv;
     ThreadData *td = arg;
     AVFrame *in = td->in, *out = td->out;
     int offset = jobnr;
     int *write = &td->write[jobnr];
     FFTComplex *hrtf = s->data_hrtf[jobnr];
     int *n_clippings = &td->n_clippings[jobnr];
     float *ringbuffer = td->ringbuffer[jobnr];
     const int ir_len = s->ir_len;
     const float *src = (const float *)in->data[0];
     float *dst = (float *)out->data[0];
     const int in_channels = in->channels;
     const int buffer_length = s->buffer_length;
     const uint32_t modulo = (uint32_t)buffer_length - 1;
     FFTComplex *fft_in = s->temp_fft[jobnr];
     FFTContext *ifft = s->ifft[jobnr];
     FFTContext *fft = s->fft[jobnr];
     const int n_fft = s->n_fft;
     const float fft_scale = 1.0f / s->n_fft;
     FFTComplex *hrtf_offset;
     int wr = *write;
     int n_read;
     int i, j;
 
     dst += offset;
 
     n_read = FFMIN(s->ir_len, in->nb_samples);
     for (j = 0; j < n_read; j++) {
         dst[2 * j]     = ringbuffer[wr];
         ringbuffer[wr] = 0.0;
         wr  = (wr + 1) & modulo;
     }
 
     for (j = n_read; j < in->nb_samples; j++) {
         dst[2 * j] = 0;
     }
 
     for (i = 0; i < in_channels; i++) {
         if (i == s->lfe_channel) {
             for (j = 0; j < in->nb_samples; j++) {
                 dst[2 * j] += src[i + j * in_channels] * s->gain_lfe;
             }
             continue;
         }
 
         offset = i * n_fft;
         hrtf_offset = hrtf + offset;
 
         memset(fft_in, 0, sizeof(FFTComplex) * n_fft);
 
         for (j = 0; j < in->nb_samples; j++) {
             fft_in[j].re = src[j * in_channels + i];
         }
 
         av_fft_permute(fft, fft_in);
         av_fft_calc(fft, fft_in);
         for (j = 0; j < n_fft; j++) {
             const FFTComplex *hcomplex = hrtf_offset + j;
             const float re = fft_in[j].re;
             const float im = fft_in[j].im;
 
             fft_in[j].re = re * hcomplex->re - im * hcomplex->im;
             fft_in[j].im = re * hcomplex->im + im * hcomplex->re;
         }
 
         av_fft_permute(ifft, fft_in);
         av_fft_calc(ifft, fft_in);
 
         for (j = 0; j < in->nb_samples; j++) {
             dst[2 * j] += fft_in[j].re * fft_scale;
         }
 
         for (j = 0; j < ir_len - 1; j++) {
             int write_pos = (wr + j) & modulo;
 
             *(ringbuffer + write_pos) += fft_in[in->nb_samples + j].re * fft_scale;
         }
     }
 
     for (i = 0; i < out->nb_samples; i++) {
         if (fabs(*dst) > 1) {
             n_clippings[0]++;
         }
 
         dst += 2;
     }
 
     *write = wr;
 
     return 0;
 }
 
 static int read_ir(AVFilterLink *inlink, AVFrame *frame)
 {
     AVFilterContext *ctx = inlink->dst;
     HeadphoneContext *s = ctx->priv;

     int ir_len, max_ir_len, input_number, ret;

 
     for (input_number = 0; input_number < s->nb_inputs; input_number++)
         if (inlink == ctx->inputs[input_number])
             break;
 

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

     av_frame_free(&frame);
 

     if (ret < 0)
         return ret;
 

     ir_len = av_audio_fifo_size(s->in[input_number].fifo);

     max_ir_len = 65536;

     if (ir_len > max_ir_len) {
         av_log(ctx, AV_LOG_ERROR, "Too big length of IRs: %d > %d.\n", ir_len, max_ir_len);
         return AVERROR(EINVAL);
     }
     s->in[input_number].ir_len = ir_len;
     s->ir_len = FFMAX(ir_len, s->ir_len);
 
     return 0;
 }
 
 static int headphone_frame(HeadphoneContext *s, AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     AVFrame *in = s->in[0].frame;
     int n_clippings[2] = { 0 };
     ThreadData td;
     AVFrame *out;
 
     av_audio_fifo_read(s->in[0].fifo, (void **)in->extended_data, s->size);
 
     out = ff_get_audio_buffer(outlink, in->nb_samples);

     if (!out)

         return AVERROR(ENOMEM);
     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);
 
     td.in = in; td.out = out; td.write = s->write;
     td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings;
     td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src;
     td.temp_fft = s->temp_fft;
 
     if (s->type == TIME_DOMAIN) {
         ctx->internal->execute(ctx, headphone_convolute, &td, NULL, 2);
     } else {
         ctx->internal->execute(ctx, headphone_fast_convolute, &td, NULL, 2);
     }
     emms_c();
 
     if (n_clippings[0] + n_clippings[1] > 0) {
         av_log(ctx, AV_LOG_WARNING, "%d of %d samples clipped. Please reduce gain.\n",
                n_clippings[0] + n_clippings[1], out->nb_samples * 2);
     }
 
     return ff_filter_frame(outlink, out);
 }
 
 static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink)
 {
     struct HeadphoneContext *s = ctx->priv;
     const int ir_len = s->ir_len;
     int nb_irs = s->nb_irs;
     int nb_input_channels = ctx->inputs[0]->channels;
     float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10);
     FFTComplex *data_hrtf_l = NULL;
     FFTComplex *data_hrtf_r = NULL;
     FFTComplex *fft_in_l = NULL;
     FFTComplex *fft_in_r = NULL;
     float *data_ir_l = NULL;
     float *data_ir_r = NULL;

     int offset = 0, ret = 0;

     int n_fft;
     int i, j;
 
     s->buffer_length = 1 << (32 - ff_clz(s->ir_len));
     s->n_fft = n_fft = 1 << (32 - ff_clz(s->ir_len + inlink->sample_rate));
 
     if (s->type == FREQUENCY_DOMAIN) {
         fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l));
         fft_in_r = av_calloc(n_fft, sizeof(*fft_in_r));
         if (!fft_in_l || !fft_in_r) {

             ret = AVERROR(ENOMEM);
             goto fail;

         }
 
         av_fft_end(s->fft[0]);
         av_fft_end(s->fft[1]);
         s->fft[0] = av_fft_init(log2(s->n_fft), 0);
         s->fft[1] = av_fft_init(log2(s->n_fft), 0);
         av_fft_end(s->ifft[0]);
         av_fft_end(s->ifft[1]);
         s->ifft[0] = av_fft_init(log2(s->n_fft), 1);
         s->ifft[1] = av_fft_init(log2(s->n_fft), 1);
 
         if (!s->fft[0] || !s->fft[1] || !s->ifft[0] || !s->ifft[1]) {
             av_log(ctx, AV_LOG_ERROR, "Unable to create FFT contexts of size %d.\n", s->n_fft);

             ret = AVERROR(ENOMEM);
             goto fail;

         }
     }
 
     s->data_ir[0] = av_calloc(FFALIGN(s->ir_len, 16), sizeof(float) * s->nb_irs);
     s->data_ir[1] = av_calloc(FFALIGN(s->ir_len, 16), sizeof(float) * s->nb_irs);
     s->delay[0] = av_malloc_array(s->nb_irs, sizeof(float));
     s->delay[1] = av_malloc_array(s->nb_irs, sizeof(float));
 
     if (s->type == TIME_DOMAIN) {
         s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
         s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
     } else {
         s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float));
         s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float));
         s->temp_fft[0] = av_malloc_array(s->n_fft, sizeof(FFTComplex));
         s->temp_fft[1] = av_malloc_array(s->n_fft, sizeof(FFTComplex));

         if (!s->temp_fft[0] || !s->temp_fft[1]) {
             ret = AVERROR(ENOMEM);
             goto fail;
         }

     }
 
     if (!s->data_ir[0] || !s->data_ir[1] ||

         !s->ringbuffer[0] || !s->ringbuffer[1]) {
         ret = AVERROR(ENOMEM);
         goto fail;
     }

 
     s->in[0].frame = ff_get_audio_buffer(ctx->inputs[0], s->size);

     if (!s->in[0].frame) {
         ret = AVERROR(ENOMEM);
         goto fail;
     }

     for (i = 0; i < s->nb_irs; i++) {
         s->in[i + 1].frame = ff_get_audio_buffer(ctx->inputs[i + 1], s->ir_len);

         if (!s->in[i + 1].frame) {
             ret = AVERROR(ENOMEM);
             goto fail;
         }

     }
 
     if (s->type == TIME_DOMAIN) {
         s->temp_src[0] = av_calloc(FFALIGN(ir_len, 16), sizeof(float));
         s->temp_src[1] = av_calloc(FFALIGN(ir_len, 16), sizeof(float));
 
         data_ir_l = av_calloc(nb_irs * FFALIGN(ir_len, 16), sizeof(*data_ir_l));
         data_ir_r = av_calloc(nb_irs * FFALIGN(ir_len, 16), sizeof(*data_ir_r));
         if (!data_ir_r || !data_ir_l || !s->temp_src[0] || !s->temp_src[1]) {

             ret = AVERROR(ENOMEM);
             goto fail;

         }
     } else {
         data_hrtf_l = av_malloc_array(n_fft, sizeof(*data_hrtf_l) * nb_irs);
         data_hrtf_r = av_malloc_array(n_fft, sizeof(*data_hrtf_r) * nb_irs);
         if (!data_hrtf_r || !data_hrtf_l) {

             ret = AVERROR(ENOMEM);
             goto fail;

         }
     }
 
     for (i = 0; i < s->nb_irs; i++) {
         int len = s->in[i + 1].ir_len;
         int delay_l = s->in[i + 1].delay_l;
         int delay_r = s->in[i + 1].delay_r;
         int idx = -1;
         float *ptr;
 
         for (j = 0; j < inlink->channels; j++) {
             if (s->mapping[i] < 0) {
                 continue;
             }
 
             if ((av_channel_layout_extract_channel(inlink->channel_layout, j)) == (1LL << s->mapping[i])) {
                 idx = j;
                 break;
             }
         }
         if (idx == -1)
             continue;
 
         av_audio_fifo_read(s->in[i + 1].fifo, (void **)s->in[i + 1].frame->extended_data, len);
         ptr = (float *)s->in[i + 1].frame->extended_data[0];
 
         if (s->type == TIME_DOMAIN) {
             offset = idx * FFALIGN(len, 16);
             for (j = 0; j < len; j++) {
                 data_ir_l[offset + j] = ptr[len * 2 - j * 2 - 2] * gain_lin;
                 data_ir_r[offset + j] = ptr[len * 2 - j * 2 - 1] * gain_lin;
             }
         } else {
             memset(fft_in_l, 0, n_fft * sizeof(*fft_in_l));
             memset(fft_in_r, 0, n_fft * sizeof(*fft_in_r));
 
             offset = idx * n_fft;
             for (j = 0; j < len; j++) {
                 fft_in_l[delay_l + j].re = ptr[j * 2    ] * gain_lin;
                 fft_in_r[delay_r + j].re = ptr[j * 2 + 1] * gain_lin;
             }
 
             av_fft_permute(s->fft[0], fft_in_l);
             av_fft_calc(s->fft[0], fft_in_l);
             memcpy(data_hrtf_l + offset, fft_in_l, n_fft * sizeof(*fft_in_l));
             av_fft_permute(s->fft[0], fft_in_r);
             av_fft_calc(s->fft[0], fft_in_r);
             memcpy(data_hrtf_r + offset, fft_in_r, n_fft * sizeof(*fft_in_r));
         }
     }
 
     if (s->type == TIME_DOMAIN) {
         memcpy(s->data_ir[0], data_ir_l, sizeof(float) * nb_irs * FFALIGN(ir_len, 16));
         memcpy(s->data_ir[1], data_ir_r, sizeof(float) * nb_irs * FFALIGN(ir_len, 16));
     } else {
         s->data_hrtf[0] = av_malloc_array(n_fft * s->nb_irs, sizeof(FFTComplex));
         s->data_hrtf[1] = av_malloc_array(n_fft * s->nb_irs, sizeof(FFTComplex));
         if (!s->data_hrtf[0] || !s->data_hrtf[1]) {

             ret = AVERROR(ENOMEM);
             goto fail;

         }
 
         memcpy(s->data_hrtf[0], data_hrtf_l,
             sizeof(FFTComplex) * nb_irs * n_fft);
         memcpy(s->data_hrtf[1], data_hrtf_r,
             sizeof(FFTComplex) * nb_irs * n_fft);
     }
 
     s->have_hrirs = 1;
 

 fail:
 
     av_freep(&data_ir_l);
     av_freep(&data_ir_r);
 
     av_freep(&data_hrtf_l);
     av_freep(&data_hrtf_r);
 
     av_freep(&fft_in_l);
     av_freep(&fft_in_r);
 
     return ret;

 }
 
 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext *ctx = inlink->dst;
     HeadphoneContext *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
     int ret = 0;
 

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

     if (s->pts == AV_NOPTS_VALUE)
         s->pts = in->pts;
 
     av_frame_free(&in);
 

     if (ret < 0)
         return ret;
 

     if (!s->have_hrirs && s->eof_hrirs) {
         ret = convert_coeffs(ctx, inlink);
         if (ret < 0)
             return ret;
     }
 
     if (s->have_hrirs) {
         while (av_audio_fifo_size(s->in[0].fifo) >= s->size) {
             ret = headphone_frame(s, outlink);
             if (ret < 0)

                 return ret;

         }
     }

 
     return 0;

 }
 
 static int query_formats(AVFilterContext *ctx)
 {
     struct HeadphoneContext *s = ctx->priv;
     AVFilterFormats *formats = NULL;
     AVFilterChannelLayouts *layouts = NULL;
     int ret, i;
 
     ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT);
     if (ret)
         return ret;
     ret = ff_set_common_formats(ctx, formats);
     if (ret)
         return ret;
 
     layouts = ff_all_channel_layouts();
     if (!layouts)
         return AVERROR(ENOMEM);
 
     ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts);
     if (ret)
         return ret;
 
     layouts = NULL;
     ret = ff_add_channel_layout(&layouts, AV_CH_LAYOUT_STEREO);
     if (ret)
         return ret;
 
     for (i = 1; i < s->nb_inputs; i++) {
         ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts);
         if (ret)
             return ret;
     }
 
     ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts);
     if (ret)
         return ret;
 
     formats = ff_all_samplerates();
     if (!formats)
         return AVERROR(ENOMEM);
     return ff_set_common_samplerates(ctx, formats);
 }
 
 static int config_input(AVFilterLink *inlink)
 {
     AVFilterContext *ctx = inlink->dst;
     HeadphoneContext *s = ctx->priv;
 
     if (s->type == FREQUENCY_DOMAIN) {
         inlink->partial_buf_size =
         inlink->min_samples =
         inlink->max_samples = inlink->sample_rate;
     }
 
     if (s->nb_irs < inlink->channels) {
         av_log(ctx, AV_LOG_ERROR, "Number of inputs must be >= %d.\n", inlink->channels + 1);
         return AVERROR(EINVAL);
     }
 
     return 0;
 }
 
 static av_cold int init(AVFilterContext *ctx)
 {
     HeadphoneContext *s = ctx->priv;

     int i, ret;

 
     AVFilterPad pad = {
         .name         = "in0",
         .type         = AVMEDIA_TYPE_AUDIO,
         .config_props = config_input,
         .filter_frame = filter_frame,
     };

     if ((ret = ff_insert_inpad(ctx, 0, &pad)) < 0)
         return ret;

 
     if (!s->map) {
         av_log(ctx, AV_LOG_ERROR, "Valid mapping must be set.\n");
         return AVERROR(EINVAL);
     }
 
     parse_map(ctx);
 
     s->in = av_calloc(s->nb_inputs, sizeof(*s->in));
     if (!s->in)
         return AVERROR(ENOMEM);
 
     for (i = 1; i < s->nb_inputs; i++) {
         char *name = av_asprintf("hrir%d", i - 1);
         AVFilterPad pad = {
             .name         = name,
             .type         = AVMEDIA_TYPE_AUDIO,
             .filter_frame = read_ir,
         };
         if (!name)
             return AVERROR(ENOMEM);

         if ((ret = ff_insert_inpad(ctx, i, &pad)) < 0) {
             av_freep(&pad.name);
             return ret;
         }

     }
 
     s->fdsp = avpriv_float_dsp_alloc(0);
     if (!s->fdsp)
         return AVERROR(ENOMEM);
     s->pts = AV_NOPTS_VALUE;
 
     return 0;
 }
 
 static int config_output(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     HeadphoneContext *s = ctx->priv;
     AVFilterLink *inlink = ctx->inputs[0];
     int i;
 
     if (s->type == TIME_DOMAIN)
         s->size = 1024;
     else
         s->size = inlink->sample_rate;
 
     for (i = 0; i < s->nb_inputs; i++) {
         s->in[i].fifo = av_audio_fifo_alloc(ctx->inputs[i]->format, ctx->inputs[i]->channels, 1024);
         if (!s->in[i].fifo)
             return AVERROR(ENOMEM);
     }
     s->gain_lfe = expf((s->gain - 3 * inlink->channels - 6 + s->lfe_gain) / 20 * M_LN10);
 
     return 0;
 }
 
 static int request_frame(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     HeadphoneContext *s = ctx->priv;
     int i, ret;
 
     for (i = 1; !s->eof_hrirs && i < s->nb_inputs; i++) {
         if (!s->in[i].eof) {
             ret = ff_request_frame(ctx->inputs[i]);
             if (ret == AVERROR_EOF) {
                 s->in[i].eof = 1;
                 ret = 0;
             }
             return ret;
         } else {
             if (i == s->nb_inputs - 1)
                 s->eof_hrirs = 1;
         }
     }
     return ff_request_frame(ctx->inputs[0]);
 }
 
 static av_cold void uninit(AVFilterContext *ctx)
 {
     HeadphoneContext *s = ctx->priv;
     int i;
 
     av_fft_end(s->ifft[0]);
     av_fft_end(s->ifft[1]);
     av_fft_end(s->fft[0]);
     av_fft_end(s->fft[1]);
     av_freep(&s->delay[0]);
     av_freep(&s->delay[1]);
     av_freep(&s->data_ir[0]);
     av_freep(&s->data_ir[1]);
     av_freep(&s->ringbuffer[0]);
     av_freep(&s->ringbuffer[1]);
     av_freep(&s->temp_src[0]);
     av_freep(&s->temp_src[1]);
     av_freep(&s->temp_fft[0]);
     av_freep(&s->temp_fft[1]);
     av_freep(&s->data_hrtf[0]);
     av_freep(&s->data_hrtf[1]);
     av_freep(&s->fdsp);
 
     for (i = 0; i < s->nb_inputs; i++) {
         av_frame_free(&s->in[i].frame);
         av_audio_fifo_free(s->in[i].fifo);
         if (ctx->input_pads && i)
             av_freep(&ctx->input_pads[i].name);
     }
     av_freep(&s->in);
 }
 
 #define OFFSET(x) offsetof(HeadphoneContext, x)
 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
 
 static const AVOption headphone_options[] = {
     { "map",       "set channels convolution mappings",  OFFSET(map),      AV_OPT_TYPE_STRING, {.str=NULL},            .flags = FLAGS },
     { "gain",      "set gain in dB",                     OFFSET(gain),     AV_OPT_TYPE_FLOAT,  {.dbl=0},     -20,  40, .flags = FLAGS },
     { "lfe",       "set lfe gain in dB",                 OFFSET(lfe_gain), AV_OPT_TYPE_FLOAT,  {.dbl=0},     -20,  40, .flags = FLAGS },
     { "type",      "set processing",                     OFFSET(type),     AV_OPT_TYPE_INT,    {.i64=1},       0,   1, .flags = FLAGS, "type" },
     { "time",      "time domain",                        0,                AV_OPT_TYPE_CONST,  {.i64=0},       0,   0, .flags = FLAGS, "type" },
     { "freq",      "frequency domain",                   0,                AV_OPT_TYPE_CONST,  {.i64=1},       0,   0, .flags = FLAGS, "type" },
     { NULL }
 };
 
 AVFILTER_DEFINE_CLASS(headphone);
 
 static const AVFilterPad outputs[] = {
     {
         .name          = "default",
         .type          = AVMEDIA_TYPE_AUDIO,
         .config_props  = config_output,
         .request_frame = request_frame,
     },
     { NULL }
 };
 
 AVFilter ff_af_headphone = {
     .name          = "headphone",
     .description   = NULL_IF_CONFIG_SMALL("Apply headphone binaural spatialization with HRTFs in additional streams."),
     .priv_size     = sizeof(HeadphoneContext),
     .priv_class    = &headphone_class,
     .init          = init,
     .uninit        = uninit,
     .query_formats = query_formats,
     .inputs        = NULL,
     .outputs       = outputs,
     .flags         = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_DYNAMIC_INPUTS,
 };