libavfilter/af_headphone.c - manifest_repos/ffmpeg - Git at Google

 /*
  * 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/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 "filters.h"
 #include "internal.h"
 #include "audio.h"

 #define TIME_DOMAIN      0
 #define FREQUENCY_DOMAIN 1

 #define HRIR_STEREO 0
 #define HRIR_MULTI  1

 typedef struct HeadphoneContext {
     const AVClass *class;

     char *map;
     int type;

     int lfe_channel;

     int have_hrirs;
     int eof_hrirs;

     int ir_len;
     int air_len;

     int nb_hrir_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 hrir_fmt;

     float *data_ir[2];
     float *temp_src[2];
     FFTComplex *temp_fft[2];
     FFTComplex *temp_afft[2];

     FFTContext *fft[2], *ifft[2];
     FFTComplex *data_hrtf[2];

     float (*scalarproduct_float)(const float *v1, const float *v2, int len);
     struct hrir_inputs {
         int          ir_len;
         int          eof;
     } hrir_in[64];
     uint64_t mapping[64];
 } HeadphoneContext;

 static int parse_channel_name(const char *arg, uint64_t *rchannel)
 {
     uint64_t layout = av_get_channel_layout(arg);

     if (av_get_channel_layout_nb_channels(layout) != 1)
         return AVERROR(EINVAL);
     *rchannel = layout;
     return 0;
 }

 static void parse_map(AVFilterContext *ctx)
 {
     HeadphoneContext *s = ctx->priv;
     char *arg, *tokenizer, *p;
     uint64_t used_channels = 0;

     p = s->map;
     while ((arg = av_strtok(p, "|", &tokenizer))) {
         uint64_t out_channel;

         p = NULL;
         if (parse_channel_name(arg, &out_channel)) {
             av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%s\' as channel name.\n", arg);
             continue;
         }
         if (used_channels & out_channel) {
             av_log(ctx, AV_LOG_WARNING, "Ignoring duplicate channel '%s'.\n", arg);
             continue;
         }
         used_channels        |= out_channel;
         s->mapping[s->nb_irs] = out_channel;
         s->nb_irs++;
     }

     if (s->hrir_fmt == HRIR_MULTI)
         s->nb_hrir_inputs = 1;
     else
         s->nb_hrir_inputs = s->nb_irs;
 }

 typedef struct ThreadData {
     AVFrame *in, *out;
     int *write;
     float **ir;
     int *n_clippings;
     float **ringbuffer;
     float **temp_src;
     FFTComplex **temp_fft;
     FFTComplex **temp_afft;
 } 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 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 int air_len = s->air_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[64];
     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 *cur_ir = ir;

         *dst = 0;
         for (l = 0; l < in_channels; l++) {
             *(buffer[l] + wr) = src[l];
         }

         for (l = 0; l < in_channels; cur_ir += air_len, l++) {
             const float *const bptr = buffer[l];

             if (l == s->lfe_channel) {
                 *dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;
                 continue;
             }

             read = (wr - (ir_len - 1)) & modulo;

             if (read + ir_len < buffer_length) {
                 memcpy(temp_src, bptr + read, ir_len * sizeof(*temp_src));
             } else {
                 int len = FFMIN(air_len - (read % ir_len), buffer_length - read);

                 memcpy(temp_src, bptr + read, len * sizeof(*temp_src));
                 memcpy(temp_src + len, bptr, (air_len - len) * sizeof(*temp_src));
             }

             dst[0] += s->scalarproduct_float(cur_ir, temp_src, FFALIGN(ir_len, 32));
         }

         if (fabsf(dst[0]) > 1)
             n_clippings[0]++;

         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];
     FFTComplex *fft_acc = s->temp_afft[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(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;
     }

     memset(fft_acc, 0, sizeof(FFTComplex) * n_fft);

     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_acc[j].re += re * hcomplex->re - im * hcomplex->im;
             fft_acc[j].im += re * hcomplex->im + im * hcomplex->re;
         }
     }

     av_fft_permute(ifft, fft_acc);
     av_fft_calc(ifft, fft_acc);

     for (j = 0; j < in->nb_samples; j++) {
         dst[2 * j] += fft_acc[j].re * fft_scale;
         if (fabsf(dst[2 * j]) > 1)
             n_clippings[0]++;
     }

     for (j = 0; j < ir_len - 1; j++) {
         int write_pos = (wr + j) & modulo;

         *(ringbuffer + write_pos) += fft_acc[in->nb_samples + j].re * fft_scale;
     }

     *write = wr;

     return 0;
 }

 static int check_ir(AVFilterLink *inlink, int input_number)
 {
     AVFilterContext *ctx = inlink->dst;
     HeadphoneContext *s = ctx->priv;
     int ir_len, max_ir_len;

     ir_len = ff_inlink_queued_samples(inlink);
     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->hrir_in[input_number].ir_len = ir_len;
     s->ir_len = FFMAX(ir_len, s->ir_len);

     return 0;
 }

 static int headphone_frame(HeadphoneContext *s, AVFrame *in, AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     int n_clippings[2] = { 0 };
     ThreadData td;
     AVFrame *out;

     out = ff_get_audio_buffer(outlink, in->nb_samples);
     if (!out) {
         av_frame_free(&in);
         return AVERROR(ENOMEM);
     }
     out->pts = in->pts;

     td.in = in; td.out = out; td.write = s->write;
     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;
     td.temp_afft = s->temp_afft;

     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);
     }

     av_frame_free(&in);
     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_input_channels = ctx->inputs[0]->channels;
     float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10);
     AVFrame *frame;
     int ret = 0;
     int n_fft;
     int i, j, k;

     s->air_len = 1 << (32 - ff_clz(ir_len));
     if (s->type == TIME_DOMAIN) {
         s->air_len = FFALIGN(s->air_len, 32);
     }
     s->buffer_length = 1 << (32 - ff_clz(s->air_len));
     s->n_fft = n_fft = 1 << (32 - ff_clz(ir_len + s->size));

     if (s->type == FREQUENCY_DOMAIN) {
         s->fft[0] = av_fft_init(av_log2(s->n_fft), 0);
         s->fft[1] = av_fft_init(av_log2(s->n_fft), 0);
         s->ifft[0] = av_fft_init(av_log2(s->n_fft), 1);
         s->ifft[1] = av_fft_init(av_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;
         }
     }

     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_calloc(s->n_fft, sizeof(FFTComplex));
         s->temp_fft[1] = av_calloc(s->n_fft, sizeof(FFTComplex));
         s->temp_afft[0] = av_calloc(s->n_fft, sizeof(FFTComplex));
         s->temp_afft[1] = av_calloc(s->n_fft, sizeof(FFTComplex));
         if (!s->temp_fft[0] || !s->temp_fft[1] ||
             !s->temp_afft[0] || !s->temp_afft[1]) {
             ret = AVERROR(ENOMEM);
             goto fail;
         }
     }

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

     if (s->type == TIME_DOMAIN) {
         s->temp_src[0] = av_calloc(s->air_len, sizeof(float));
         s->temp_src[1] = av_calloc(s->air_len, sizeof(float));

         s->data_ir[0] = av_calloc(nb_input_channels * s->air_len, sizeof(*s->data_ir[0]));
         s->data_ir[1] = av_calloc(nb_input_channels * s->air_len, sizeof(*s->data_ir[1]));
         if (!s->data_ir[0] || !s->data_ir[1] || !s->temp_src[0] || !s->temp_src[1]) {
             ret = AVERROR(ENOMEM);
             goto fail;
         }
     } else {
         s->data_hrtf[0] = av_calloc(n_fft, sizeof(*s->data_hrtf[0]) * nb_input_channels);
         s->data_hrtf[1] = av_calloc(n_fft, sizeof(*s->data_hrtf[1]) * nb_input_channels);
         if (!s->data_hrtf[0] || !s->data_hrtf[1]) {
             ret = AVERROR(ENOMEM);
             goto fail;
         }
     }

     for (i = 0; i < s->nb_hrir_inputs; av_frame_free(&frame), i++) {
         int len = s->hrir_in[i].ir_len;
         float *ptr;

         ret = ff_inlink_consume_samples(ctx->inputs[i + 1], len, len, &frame);
         if (ret < 0)
             goto fail;
         ptr = (float *)frame->extended_data[0];

         if (s->hrir_fmt == HRIR_STEREO) {
             int idx = av_get_channel_layout_channel_index(inlink->channel_layout,
                                                           s->mapping[i]);
             if (idx < 0)
                 continue;
             if (s->type == TIME_DOMAIN) {
                 float *data_ir_l = s->data_ir[0] + idx * s->air_len;
                 float *data_ir_r = s->data_ir[1] + idx * s->air_len;

                 for (j = 0; j < len; j++) {
                     data_ir_l[j] = ptr[len * 2 - j * 2 - 2] * gain_lin;
                     data_ir_r[j] = ptr[len * 2 - j * 2 - 1] * gain_lin;
                 }
             } else {
                 FFTComplex *fft_in_l = s->data_hrtf[0] + idx * n_fft;
                 FFTComplex *fft_in_r = s->data_hrtf[1] + idx * n_fft;

                 for (j = 0; j < len; j++) {
                     fft_in_l[j].re = ptr[j * 2    ] * gain_lin;
                     fft_in_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);
                 av_fft_permute(s->fft[0], fft_in_r);
                 av_fft_calc(s->fft[0], fft_in_r);
             }
         } else {
             int I, N = ctx->inputs[1]->channels;

             for (k = 0; k < N / 2; k++) {
                 int idx = av_get_channel_layout_channel_index(inlink->channel_layout,
                                                               s->mapping[k]);
                 if (idx < 0)
                     continue;

                 I = k * 2;
                 if (s->type == TIME_DOMAIN) {
                     float *data_ir_l = s->data_ir[0] + idx * s->air_len;
                     float *data_ir_r = s->data_ir[1] + idx * s->air_len;

                     for (j = 0; j < len; j++) {
                         data_ir_l[j] = ptr[len * N - j * N - N + I    ] * gain_lin;
                         data_ir_r[j] = ptr[len * N - j * N - N + I + 1] * gain_lin;
                     }
                 } else {
                     FFTComplex *fft_in_l = s->data_hrtf[0] + idx * n_fft;
                     FFTComplex *fft_in_r = s->data_hrtf[1] + idx * n_fft;

                     for (j = 0; j < len; j++) {
                         fft_in_l[j].re = ptr[j * N + I    ] * gain_lin;
                         fft_in_r[j].re = ptr[j * N + I + 1] * gain_lin;
                     }

                     av_fft_permute(s->fft[0], fft_in_l);
                     av_fft_calc(s->fft[0], fft_in_l);
                     av_fft_permute(s->fft[0], fft_in_r);
                     av_fft_calc(s->fft[0], fft_in_r);
                 }
             }
         }
     }

     s->have_hrirs = 1;

 fail:
     return ret;
 }

 static int activate(AVFilterContext *ctx)
 {
     HeadphoneContext *s = ctx->priv;
     AVFilterLink *inlink = ctx->inputs[0];
     AVFilterLink *outlink = ctx->outputs[0];
     AVFrame *in = NULL;
     int i, ret;

     FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
     if (!s->eof_hrirs) {
         int eof = 1;
         for (i = 0; i < s->nb_hrir_inputs; i++) {
             AVFilterLink *input = ctx->inputs[i + 1];

             if (s->hrir_in[i].eof)
                 continue;

             if ((ret = check_ir(input, i)) < 0)
                 return ret;

             if (ff_outlink_get_status(input) == AVERROR_EOF) {
                 if (!ff_inlink_queued_samples(input)) {
                     av_log(ctx, AV_LOG_ERROR, "No samples provided for "
                            "HRIR stream %d.\n", i);
                     return AVERROR_INVALIDDATA;
                 }
                 s->hrir_in[i].eof = 1;
             } else {
                 if (ff_outlink_frame_wanted(ctx->outputs[0]))
                     ff_inlink_request_frame(input);
                 eof = 0;
             }
         }
         if (!eof)
             return 0;
         s->eof_hrirs = 1;

         ret = convert_coeffs(ctx, inlink);
         if (ret < 0)
             return ret;
     } else if (!s->have_hrirs)
         return AVERROR_EOF;

     if ((ret = ff_inlink_consume_samples(ctx->inputs[0], s->size, s->size, &in)) > 0) {
         ret = headphone_frame(s, in, outlink);
         if (ret < 0)
             return ret;
     }

     if (ret < 0)
         return ret;

     FF_FILTER_FORWARD_STATUS(ctx->inputs[0], ctx->outputs[0]);
     if (ff_outlink_frame_wanted(ctx->outputs[0]))
         ff_inlink_request_frame(ctx->inputs[0]);

     return 0;
 }

 static int query_formats(AVFilterContext *ctx)
 {
     struct HeadphoneContext *s = ctx->priv;
     AVFilterFormats *formats = NULL;
     AVFilterChannelLayouts *layouts = NULL;
     AVFilterChannelLayouts *stereo_layout = NULL;
     AVFilterChannelLayouts *hrir_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]->outcfg.channel_layouts);
     if (ret)
         return ret;

     ret = ff_add_channel_layout(&stereo_layout, AV_CH_LAYOUT_STEREO);
     if (ret)
         return ret;
     ret = ff_channel_layouts_ref(stereo_layout, &ctx->outputs[0]->incfg.channel_layouts);
     if (ret)
         return ret;

     if (s->hrir_fmt == HRIR_MULTI) {
         hrir_layouts = ff_all_channel_counts();
         if (!hrir_layouts)
             return AVERROR(ENOMEM);
         ret = ff_channel_layouts_ref(hrir_layouts, &ctx->inputs[1]->outcfg.channel_layouts);
         if (ret)
             return ret;
     } else {
         for (i = 1; i <= s->nb_hrir_inputs; i++) {
             ret = ff_channel_layouts_ref(stereo_layout, &ctx->inputs[i]->outcfg.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->nb_irs < inlink->channels) {
         av_log(ctx, AV_LOG_ERROR, "Number of HRIRs must be >= %d.\n", inlink->channels);
         return AVERROR(EINVAL);
     }

     s->lfe_channel = av_get_channel_layout_channel_index(inlink->channel_layout,
                                                          AV_CH_LOW_FREQUENCY);
     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,
     };
     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);

     for (i = 0; i < s->nb_hrir_inputs; i++) {
         char *name = av_asprintf("hrir%d", i);
         AVFilterPad pad = {
             .name         = name,
             .type         = AVMEDIA_TYPE_AUDIO,
         };
         if (!name)
             return AVERROR(ENOMEM);
         if ((ret = ff_insert_inpad(ctx, i + 1, &pad)) < 0) {
             av_freep(&pad.name);
             return ret;
         }
     }

     if (s->type == TIME_DOMAIN) {
         AVFloatDSPContext *fdsp = avpriv_float_dsp_alloc(0);
         if (!fdsp)
             return AVERROR(ENOMEM);
         s->scalarproduct_float = fdsp->scalarproduct_float;
         av_free(fdsp);
     }

     return 0;
 }

 static int config_output(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     HeadphoneContext *s = ctx->priv;
     AVFilterLink *inlink = ctx->inputs[0];

     if (s->hrir_fmt == HRIR_MULTI) {
         AVFilterLink *hrir_link = ctx->inputs[1];

         if (hrir_link->channels < inlink->channels * 2) {
             av_log(ctx, AV_LOG_ERROR, "Number of channels in HRIR stream must be >= %d.\n", inlink->channels * 2);
             return AVERROR(EINVAL);
         }
     }

     s->gain_lfe = expf((s->gain - 3 * inlink->channels + s->lfe_gain) / 20 * M_LN10);

     return 0;
 }

 static av_cold void uninit(AVFilterContext *ctx)
 {
     HeadphoneContext *s = ctx->priv;

     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->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->temp_afft[0]);
     av_freep(&s->temp_afft[1]);
     av_freep(&s->data_hrtf[0]);
     av_freep(&s->data_hrtf[1]);

     for (unsigned i = 1; i < ctx->nb_inputs; i++)
         av_freep(&ctx->input_pads[i].name);
 }

 #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" },
     { "size",      "set frame size",                     OFFSET(size),     AV_OPT_TYPE_INT,    {.i64=1024},1024,96000, .flags = FLAGS },
     { "hrir",      "set hrir format",                    OFFSET(hrir_fmt), AV_OPT_TYPE_INT,    {.i64=HRIR_STEREO}, 0, 1, .flags = FLAGS, "hrir" },
     { "stereo",    "hrir files have exactly 2 channels", 0,                AV_OPT_TYPE_CONST,  {.i64=HRIR_STEREO}, 0, 0, .flags = FLAGS, "hrir" },
     { "multich",   "single multichannel hrir file",      0,                AV_OPT_TYPE_CONST,  {.i64=HRIR_MULTI},  0, 0, .flags = FLAGS, "hrir" },
     { NULL }
 };

 AVFILTER_DEFINE_CLASS(headphone);

 static const AVFilterPad outputs[] = {
     {
         .name          = "default",
         .type          = AVMEDIA_TYPE_AUDIO,
         .config_props  = config_output,
     },
     { 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,
     .activate      = activate,
     .inputs        = NULL,
     .outputs       = outputs,
     .flags         = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_DYNAMIC_INPUTS,
 };
	/*
	* 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/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 "filters.h"
	#include "internal.h"
	#include "audio.h"

	#define TIME_DOMAIN 0
	#define FREQUENCY_DOMAIN 1

	#define HRIR_STEREO 0
	#define HRIR_MULTI 1

	typedef struct HeadphoneContext {
	const AVClass *class;

	char *map;
	int type;

	int lfe_channel;

	int have_hrirs;
	int eof_hrirs;

	int ir_len;
	int air_len;

	int nb_hrir_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 hrir_fmt;

	float *data_ir[2];
	float *temp_src[2];
	FFTComplex *temp_fft[2];
	FFTComplex *temp_afft[2];

	FFTContext fft[2], ifft[2];
	FFTComplex *data_hrtf[2];

	float (scalarproduct_float)(const float v1, const float *v2, int len);
	struct hrir_inputs {
	int ir_len;
	int eof;
	} hrir_in[64];
	uint64_t mapping[64];
	} HeadphoneContext;

	static int parse_channel_name(const char arg, uint64_t rchannel)
	{
	uint64_t layout = av_get_channel_layout(arg);

	if (av_get_channel_layout_nb_channels(layout) != 1)
	return AVERROR(EINVAL);
	*rchannel = layout;
	return 0;
	}

	static void parse_map(AVFilterContext *ctx)
	{
	HeadphoneContext *s = ctx->priv;
	char arg, tokenizer, *p;
	uint64_t used_channels = 0;

	p = s->map;
	while ((arg = av_strtok(p, "\|", &tokenizer))) {
	uint64_t out_channel;

	p = NULL;
	if (parse_channel_name(arg, &out_channel)) {
	av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%s\' as channel name.\n", arg);
	continue;
	}
	if (used_channels & out_channel) {
	av_log(ctx, AV_LOG_WARNING, "Ignoring duplicate channel '%s'.\n", arg);
	continue;
	}
	used_channels \|= out_channel;
	s->mapping[s->nb_irs] = out_channel;
	s->nb_irs++;
	}

	if (s->hrir_fmt == HRIR_MULTI)
	s->nb_hrir_inputs = 1;
	else
	s->nb_hrir_inputs = s->nb_irs;
	}

	typedef struct ThreadData {
	AVFrame in, out;
	int *write;
	float **ir;
	int *n_clippings;
	float **ringbuffer;
	float **temp_src;
	FFTComplex **temp_fft;
	FFTComplex **temp_afft;
	} 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 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 int air_len = s->air_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[64];
	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 *cur_ir = ir;

	*dst = 0;
	for (l = 0; l < in_channels; l++) {
	*(buffer[l] + wr) = src[l];
	}

	for (l = 0; l < in_channels; cur_ir += air_len, l++) {
	const float *const bptr = buffer[l];

	if (l == s->lfe_channel) {
	dst += (buffer[s->lfe_channel] + wr) * s->gain_lfe;
	continue;
	}

	read = (wr - (ir_len - 1)) & modulo;

	if (read + ir_len < buffer_length) {
	memcpy(temp_src, bptr + read, ir_len * sizeof(*temp_src));
	} else {
	int len = FFMIN(air_len - (read % ir_len), buffer_length - read);

	memcpy(temp_src, bptr + read, len * sizeof(*temp_src));
	memcpy(temp_src + len, bptr, (air_len - len) * sizeof(*temp_src));
	}

	dst[0] += s->scalarproduct_float(cur_ir, temp_src, FFALIGN(ir_len, 32));
	}

	if (fabsf(dst[0]) > 1)
	n_clippings[0]++;

	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];
	FFTComplex *fft_acc = s->temp_afft[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(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;
	}

	memset(fft_acc, 0, sizeof(FFTComplex) * n_fft);

	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_acc[j].re += re * hcomplex->re - im * hcomplex->im;
	fft_acc[j].im += re * hcomplex->im + im * hcomplex->re;
	}
	}

	av_fft_permute(ifft, fft_acc);
	av_fft_calc(ifft, fft_acc);

	for (j = 0; j < in->nb_samples; j++) {
	dst[2 * j] += fft_acc[j].re * fft_scale;
	if (fabsf(dst[2 * j]) > 1)
	n_clippings[0]++;
	}

	for (j = 0; j < ir_len - 1; j++) {
	int write_pos = (wr + j) & modulo;

	(ringbuffer + write_pos) += fft_acc[in->nb_samples + j].re fft_scale;
	}

	*write = wr;

	return 0;
	}

	static int check_ir(AVFilterLink *inlink, int input_number)
	{
	AVFilterContext *ctx = inlink->dst;
	HeadphoneContext *s = ctx->priv;
	int ir_len, max_ir_len;

	ir_len = ff_inlink_queued_samples(inlink);
	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->hrir_in[input_number].ir_len = ir_len;
	s->ir_len = FFMAX(ir_len, s->ir_len);

	return 0;
	}

	static int headphone_frame(HeadphoneContext s, AVFrame in, AVFilterLink *outlink)
	{
	AVFilterContext *ctx = outlink->src;
	int n_clippings[2] = { 0 };
	ThreadData td;
	AVFrame *out;

	out = ff_get_audio_buffer(outlink, in->nb_samples);
	if (!out) {
	av_frame_free(&in);
	return AVERROR(ENOMEM);
	}
	out->pts = in->pts;

	td.in = in; td.out = out; td.write = s->write;
	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;
	td.temp_afft = s->temp_afft;

	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);
	}

	av_frame_free(&in);
	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_input_channels = ctx->inputs[0]->channels;
	float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10);
	AVFrame *frame;
	int ret = 0;
	int n_fft;
	int i, j, k;

	s->air_len = 1 << (32 - ff_clz(ir_len));
	if (s->type == TIME_DOMAIN) {
	s->air_len = FFALIGN(s->air_len, 32);
	}
	s->buffer_length = 1 << (32 - ff_clz(s->air_len));
	s->n_fft = n_fft = 1 << (32 - ff_clz(ir_len + s->size));

	if (s->type == FREQUENCY_DOMAIN) {
	s->fft[0] = av_fft_init(av_log2(s->n_fft), 0);
	s->fft[1] = av_fft_init(av_log2(s->n_fft), 0);
	s->ifft[0] = av_fft_init(av_log2(s->n_fft), 1);
	s->ifft[1] = av_fft_init(av_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;
	}
	}

	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_calloc(s->n_fft, sizeof(FFTComplex));
	s->temp_fft[1] = av_calloc(s->n_fft, sizeof(FFTComplex));
	s->temp_afft[0] = av_calloc(s->n_fft, sizeof(FFTComplex));
	s->temp_afft[1] = av_calloc(s->n_fft, sizeof(FFTComplex));
	if (!s->temp_fft[0] \|\| !s->temp_fft[1] \|\|
	!s->temp_afft[0] \|\| !s->temp_afft[1]) {
	ret = AVERROR(ENOMEM);
	goto fail;
	}
	}

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

	if (s->type == TIME_DOMAIN) {
	s->temp_src[0] = av_calloc(s->air_len, sizeof(float));
	s->temp_src[1] = av_calloc(s->air_len, sizeof(float));

	s->data_ir[0] = av_calloc(nb_input_channels * s->air_len, sizeof(*s->data_ir[0]));
	s->data_ir[1] = av_calloc(nb_input_channels * s->air_len, sizeof(*s->data_ir[1]));
	if (!s->data_ir[0] \|\| !s->data_ir[1] \|\| !s->temp_src[0] \|\| !s->temp_src[1]) {
	ret = AVERROR(ENOMEM);
	goto fail;
	}
	} else {
	s->data_hrtf[0] = av_calloc(n_fft, sizeof(s->data_hrtf[0]) nb_input_channels);
	s->data_hrtf[1] = av_calloc(n_fft, sizeof(s->data_hrtf[1]) nb_input_channels);
	if (!s->data_hrtf[0] \|\| !s->data_hrtf[1]) {
	ret = AVERROR(ENOMEM);
	goto fail;
	}
	}

	for (i = 0; i < s->nb_hrir_inputs; av_frame_free(&frame), i++) {
	int len = s->hrir_in[i].ir_len;
	float *ptr;

	ret = ff_inlink_consume_samples(ctx->inputs[i + 1], len, len, &frame);
	if (ret < 0)
	goto fail;
	ptr = (float *)frame->extended_data[0];

	if (s->hrir_fmt == HRIR_STEREO) {
	int idx = av_get_channel_layout_channel_index(inlink->channel_layout,
	s->mapping[i]);
	if (idx < 0)
	continue;
	if (s->type == TIME_DOMAIN) {
	float data_ir_l = s->data_ir[0] + idx s->air_len;
	float data_ir_r = s->data_ir[1] + idx s->air_len;

	for (j = 0; j < len; j++) {
	data_ir_l[j] = ptr[len * 2 - j * 2 - 2] * gain_lin;
	data_ir_r[j] = ptr[len * 2 - j * 2 - 1] * gain_lin;
	}
	} else {
	FFTComplex fft_in_l = s->data_hrtf[0] + idx n_fft;
	FFTComplex fft_in_r = s->data_hrtf[1] + idx n_fft;

	for (j = 0; j < len; j++) {
	fft_in_l[j].re = ptr[j * 2 ] * gain_lin;
	fft_in_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);
	av_fft_permute(s->fft[0], fft_in_r);
	av_fft_calc(s->fft[0], fft_in_r);
	}
	} else {
	int I, N = ctx->inputs[1]->channels;

	for (k = 0; k < N / 2; k++) {
	int idx = av_get_channel_layout_channel_index(inlink->channel_layout,
	s->mapping[k]);
	if (idx < 0)
	continue;

	I = k * 2;
	if (s->type == TIME_DOMAIN) {
	float data_ir_l = s->data_ir[0] + idx s->air_len;
	float data_ir_r = s->data_ir[1] + idx s->air_len;

	for (j = 0; j < len; j++) {
	data_ir_l[j] = ptr[len * N - j * N - N + I ] * gain_lin;
	data_ir_r[j] = ptr[len * N - j * N - N + I + 1] * gain_lin;
	}
	} else {
	FFTComplex fft_in_l = s->data_hrtf[0] + idx n_fft;
	FFTComplex fft_in_r = s->data_hrtf[1] + idx n_fft;

	for (j = 0; j < len; j++) {
	fft_in_l[j].re = ptr[j * N + I ] * gain_lin;
	fft_in_r[j].re = ptr[j * N + I + 1] * gain_lin;
	}

	av_fft_permute(s->fft[0], fft_in_l);
	av_fft_calc(s->fft[0], fft_in_l);
	av_fft_permute(s->fft[0], fft_in_r);
	av_fft_calc(s->fft[0], fft_in_r);
	}
	}
	}
	}

	s->have_hrirs = 1;

	fail:
	return ret;
	}

	static int activate(AVFilterContext *ctx)
	{
	HeadphoneContext *s = ctx->priv;
	AVFilterLink *inlink = ctx->inputs[0];
	AVFilterLink *outlink = ctx->outputs[0];
	AVFrame *in = NULL;
	int i, ret;

	FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
	if (!s->eof_hrirs) {
	int eof = 1;
	for (i = 0; i < s->nb_hrir_inputs; i++) {
	AVFilterLink *input = ctx->inputs[i + 1];

	if (s->hrir_in[i].eof)
	continue;

	if ((ret = check_ir(input, i)) < 0)
	return ret;

	if (ff_outlink_get_status(input) == AVERROR_EOF) {
	if (!ff_inlink_queued_samples(input)) {
	av_log(ctx, AV_LOG_ERROR, "No samples provided for "
	"HRIR stream %d.\n", i);
	return AVERROR_INVALIDDATA;
	}
	s->hrir_in[i].eof = 1;
	} else {
	if (ff_outlink_frame_wanted(ctx->outputs[0]))
	ff_inlink_request_frame(input);
	eof = 0;
	}
	}
	if (!eof)
	return 0;
	s->eof_hrirs = 1;

	ret = convert_coeffs(ctx, inlink);
	if (ret < 0)
	return ret;
	} else if (!s->have_hrirs)
	return AVERROR_EOF;

	if ((ret = ff_inlink_consume_samples(ctx->inputs[0], s->size, s->size, &in)) > 0) {
	ret = headphone_frame(s, in, outlink);
	if (ret < 0)
	return ret;
	}

	if (ret < 0)
	return ret;

	FF_FILTER_FORWARD_STATUS(ctx->inputs[0], ctx->outputs[0]);
	if (ff_outlink_frame_wanted(ctx->outputs[0]))
	ff_inlink_request_frame(ctx->inputs[0]);

	return 0;
	}

	static int query_formats(AVFilterContext *ctx)
	{
	struct HeadphoneContext *s = ctx->priv;
	AVFilterFormats *formats = NULL;
	AVFilterChannelLayouts *layouts = NULL;
	AVFilterChannelLayouts *stereo_layout = NULL;
	AVFilterChannelLayouts *hrir_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]->outcfg.channel_layouts);
	if (ret)
	return ret;

	ret = ff_add_channel_layout(&stereo_layout, AV_CH_LAYOUT_STEREO);
	if (ret)
	return ret;
	ret = ff_channel_layouts_ref(stereo_layout, &ctx->outputs[0]->incfg.channel_layouts);
	if (ret)
	return ret;

	if (s->hrir_fmt == HRIR_MULTI) {
	hrir_layouts = ff_all_channel_counts();
	if (!hrir_layouts)
	return AVERROR(ENOMEM);
	ret = ff_channel_layouts_ref(hrir_layouts, &ctx->inputs[1]->outcfg.channel_layouts);
	if (ret)
	return ret;
	} else {
	for (i = 1; i <= s->nb_hrir_inputs; i++) {
	ret = ff_channel_layouts_ref(stereo_layout, &ctx->inputs[i]->outcfg.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->nb_irs < inlink->channels) {
	av_log(ctx, AV_LOG_ERROR, "Number of HRIRs must be >= %d.\n", inlink->channels);
	return AVERROR(EINVAL);
	}

	s->lfe_channel = av_get_channel_layout_channel_index(inlink->channel_layout,
	AV_CH_LOW_FREQUENCY);
	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,
	};
	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);

	for (i = 0; i < s->nb_hrir_inputs; i++) {
	char *name = av_asprintf("hrir%d", i);
	AVFilterPad pad = {
	.name = name,
	.type = AVMEDIA_TYPE_AUDIO,
	};
	if (!name)
	return AVERROR(ENOMEM);
	if ((ret = ff_insert_inpad(ctx, i + 1, &pad)) < 0) {
	av_freep(&pad.name);
	return ret;
	}
	}

	if (s->type == TIME_DOMAIN) {
	AVFloatDSPContext *fdsp = avpriv_float_dsp_alloc(0);
	if (!fdsp)
	return AVERROR(ENOMEM);
	s->scalarproduct_float = fdsp->scalarproduct_float;
	av_free(fdsp);
	}

	return 0;
	}

	static int config_output(AVFilterLink *outlink)
	{
	AVFilterContext *ctx = outlink->src;
	HeadphoneContext *s = ctx->priv;
	AVFilterLink *inlink = ctx->inputs[0];

	if (s->hrir_fmt == HRIR_MULTI) {
	AVFilterLink *hrir_link = ctx->inputs[1];

	if (hrir_link->channels < inlink->channels * 2) {
	av_log(ctx, AV_LOG_ERROR, "Number of channels in HRIR stream must be >= %d.\n", inlink->channels * 2);
	return AVERROR(EINVAL);
	}
	}

	s->gain_lfe = expf((s->gain - 3 * inlink->channels + s->lfe_gain) / 20 * M_LN10);

	return 0;
	}

	static av_cold void uninit(AVFilterContext *ctx)
	{
	HeadphoneContext *s = ctx->priv;

	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->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->temp_afft[0]);
	av_freep(&s->temp_afft[1]);
	av_freep(&s->data_hrtf[0]);
	av_freep(&s->data_hrtf[1]);

	for (unsigned i = 1; i < ctx->nb_inputs; i++)
	av_freep(&ctx->input_pads[i].name);
	}

	#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" },
	{ "size", "set frame size", OFFSET(size), AV_OPT_TYPE_INT, {.i64=1024},1024,96000, .flags = FLAGS },
	{ "hrir", "set hrir format", OFFSET(hrir_fmt), AV_OPT_TYPE_INT, {.i64=HRIR_STEREO}, 0, 1, .flags = FLAGS, "hrir" },
	{ "stereo", "hrir files have exactly 2 channels", 0, AV_OPT_TYPE_CONST, {.i64=HRIR_STEREO}, 0, 0, .flags = FLAGS, "hrir" },
	{ "multich", "single multichannel hrir file", 0, AV_OPT_TYPE_CONST, {.i64=HRIR_MULTI}, 0, 0, .flags = FLAGS, "hrir" },
	{ NULL }
	};

	AVFILTER_DEFINE_CLASS(headphone);

	static const AVFilterPad outputs[] = {
	{
	.name = "default",
	.type = AVMEDIA_TYPE_AUDIO,
	.config_props = config_output,
	},
	{ 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,
	.activate = activate,
	.inputs = NULL,
	.outputs = outputs,
	.flags = AVFILTER_FLAG_SLICE_THREADS \| AVFILTER_FLAG_DYNAMIC_INPUTS,
	};