/* * 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 #include "libavutil/avstring.h" #include "libavutil/channel_layout.h" #include "libavutil/float_dsp.h" #include "libavutil/intmath.h" #include "libavutil/opt.h" #include "libavutil/tx.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]; AVComplexFloat *out_fft[2]; AVComplexFloat *in_fft[2]; AVComplexFloat *temp_afft[2]; AVTXContext *fft[2], *ifft[2]; av_tx_fn tx_fn[2], itx_fn[2]; AVComplexFloat *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; AVComplexFloat **out_fft; AVComplexFloat **in_fft; AVComplexFloat **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]; AVComplexFloat *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; AVComplexFloat *fft_out = s->out_fft[jobnr]; AVComplexFloat *fft_in = s->in_fft[jobnr]; AVComplexFloat *fft_acc = s->temp_afft[jobnr]; AVTXContext *ifft = s->ifft[jobnr]; AVTXContext *fft = s->fft[jobnr]; av_tx_fn tx_fn = s->tx_fn[jobnr]; av_tx_fn itx_fn = s->itx_fn[jobnr]; const int n_fft = s->n_fft; const float fft_scale = 1.0f / s->n_fft; AVComplexFloat *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(AVComplexFloat) * 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(AVComplexFloat) * n_fft); for (j = 0; j < in->nb_samples; j++) { fft_in[j].re = src[j * in_channels + i]; } tx_fn(fft, fft_out, fft_in, sizeof(float)); for (j = 0; j < n_fft; j++) { const AVComplexFloat *hcomplex = hrtf_offset + j; const float re = fft_out[j].re; const float im = fft_out[j].im; fft_acc[j].re += re * hcomplex->re - im * hcomplex->im; fft_acc[j].im += re * hcomplex->im + im * hcomplex->re; } } itx_fn(ifft, fft_out, fft_acc, sizeof(float)); for (j = 0; j < in->nb_samples; j++) { dst[2 * j] += fft_out[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_out[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.out_fft = s->out_fft; td.in_fft = s->in_fft; td.temp_afft = s->temp_afft; if (s->type == TIME_DOMAIN) { ff_filter_execute(ctx, headphone_convolute, &td, NULL, 2); } else { ff_filter_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) { float scale; ret = av_tx_init(&s->fft[0], &s->tx_fn[0], AV_TX_FLOAT_FFT, 0, s->n_fft, &scale, 0); if (ret < 0) goto fail; ret = av_tx_init(&s->fft[1], &s->tx_fn[1], AV_TX_FLOAT_FFT, 0, s->n_fft, &scale, 0); if (ret < 0) goto fail; ret = av_tx_init(&s->ifft[0], &s->itx_fn[0], AV_TX_FLOAT_FFT, 1, s->n_fft, &scale, 0); if (ret < 0) goto fail; ret = av_tx_init(&s->ifft[1], &s->itx_fn[1], AV_TX_FLOAT_FFT, 1, s->n_fft, &scale, 0); if (ret < 0) goto fail; 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->out_fft[0] = av_calloc(s->n_fft, sizeof(AVComplexFloat)); s->out_fft[1] = av_calloc(s->n_fft, sizeof(AVComplexFloat)); s->in_fft[0] = av_calloc(s->n_fft, sizeof(AVComplexFloat)); s->in_fft[1] = av_calloc(s->n_fft, sizeof(AVComplexFloat)); s->temp_afft[0] = av_calloc(s->n_fft, sizeof(AVComplexFloat)); s->temp_afft[1] = av_calloc(s->n_fft, sizeof(AVComplexFloat)); if (!s->in_fft[0] || !s->in_fft[1] || !s->out_fft[0] || !s->out_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 { AVComplexFloat *fft_out_l = s->data_hrtf[0] + idx * n_fft; AVComplexFloat *fft_out_r = s->data_hrtf[1] + idx * n_fft; AVComplexFloat *fft_in_l = s->in_fft[0]; AVComplexFloat *fft_in_r = s->in_fft[1]; 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; } s->tx_fn[0](s->fft[0], fft_out_l, fft_in_l, sizeof(float)); s->tx_fn[0](s->fft[0], fft_out_r, fft_in_r, sizeof(float)); } } 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 { AVComplexFloat *fft_out_l = s->data_hrtf[0] + idx * n_fft; AVComplexFloat *fft_out_r = s->data_hrtf[1] + idx * n_fft; AVComplexFloat *fft_in_l = s->in_fft[0]; AVComplexFloat *fft_in_r = s->in_fft[1]; 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; } s->tx_fn[0](s->fft[0], fft_out_l, fft_in_l, sizeof(float)); s->tx_fn[0](s->fft[0], fft_out_r, fft_in_r, sizeof(float)); } } } } 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; } } return ff_set_common_all_samplerates(ctx); } 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_tx_uninit(&s->ifft[0]); av_tx_uninit(&s->ifft[1]); av_tx_uninit(&s->fft[0]); av_tx_uninit(&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->out_fft[0]); av_freep(&s->out_fft[1]); av_freep(&s->in_fft[0]); av_freep(&s->in_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 } }; const 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, };