@@ -19,6 +19,7 @@ version <next>:

 - surround audio filter

 - sofalizer filter switched to libmysofa

 - Gremlin Digital Video demuxer and decoder

+- headphone audio filter

 version 3.3:

 - CrystalHD decoder moved to new decode API

@@ -2789,6 +2789,49 @@ Samples where the target gain does not match between channels

 @end table

 @end table

+@section headphone

+

+Apply head-related transfer functions (HRTFs) to create virtual

+loudspeakers around the user for binaural listening via headphones.

+The HRIRs are provided via additional streams, for each channel

+one stereo input stream is needed.

+

+The filter accepts the following options:

+

+@table @option

+@item map

+Set mapping of input streams for convolution.

+The argument is a '|'-separated list of channel names in order as they

+are given as additional stream inputs for filter.

+This also specify number of input streams. Number of input streams

+must be not less than number of channels in first stream plus one.

+

+@item gain

+Set gain applied to audio. Value is in dB. Default is 0.

+

+@item type

+Set processing type. Can be @var{time} or @var{freq}. @var{time} is

+processing audio in time domain which is slow.

+@var{freq} is processing audio in frequency domain which is fast.

+Default is @var{freq}.

+

+@item lfe

+Set custom gain for LFE channels. Value is in dB. Default is 0.

+@end table

+

+@subsection Examples

+

+@itemize

+@item

+Full example using wav files as coefficients with amovie filters for 7.1 downmix,

+each amovie filter use stereo file with IR coefficients as input.

+The files give coefficients for each position of virtual loudspeaker:

+@example

+ffmpeg -i input.wav -lavfi-complex "amovie=azi_270_ele_0_DFC.wav[sr],amovie=azi_90_ele_0_DFC.wav[sl],amovie=azi_225_ele_0_DFC.wav[br],amovie=azi_135_ele_0_DFC.wav[bl],amovie=azi_0_ele_0_DFC.wav,asplit[fc][lfe],amovie=azi_35_ele_0_DFC.wav[fl],amovie=azi_325_ele_0_DFC.wav[fr],[a:0][fl][fr][fc][lfe][bl][br][sl][sr]headphone=FL|FR|FC|LFE|BL|BR|SL|SR"

+output.wav

+@end example

+@end itemize

+

 @section highpass

 Apply a high-pass filter with 3dB point frequency.

@@ -92,6 +92,7 @@ OBJS-$(CONFIG_EXTRASTEREO_FILTER)            += af_extrastereo.o

 OBJS-$(CONFIG_FIREQUALIZER_FILTER)           += af_firequalizer.o

 OBJS-$(CONFIG_FLANGER_FILTER)                += af_flanger.o generate_wave_table.o

 OBJS-$(CONFIG_HDCD_FILTER)                   += af_hdcd.o

+OBJS-$(CONFIG_HEADPHONE_FILTER)              += af_headphone.o

 OBJS-$(CONFIG_HIGHPASS_FILTER)               += af_biquads.o

 OBJS-$(CONFIG_JOIN_FILTER)                   += af_join.o

 OBJS-$(CONFIG_LADSPA_FILTER)                 += af_ladspa.o

new file mode 100644

@@ -0,0 +1,811 @@

+/*

+ * 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;

+

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

+        if (inlink == ctx->inputs[input_number])

+            break;

+

+    av_audio_fifo_write(s->in[input_number].fifo, (void **)frame->extended_data,

+                        frame->nb_samples);

+    av_frame_free(&frame);

+

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

+    max_ir_len = 4096;

+    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) {

+        av_frame_free(&in);

+        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;

+    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) {

+            return AVERROR(ENOMEM);

+        }

+

+        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);

+            return AVERROR(ENOMEM);

+        }

+    }

+

+    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])

+            return AVERROR(ENOMEM);

+    }

+

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

+        !s->ringbuffer[0] || !s->ringbuffer[1])

+        return AVERROR(ENOMEM);

+

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

+    if (!s->in[0].frame)

+        return AVERROR(ENOMEM);

+    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)

+            return AVERROR(ENOMEM);

+    }

+

+    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]) {

+            av_free(data_ir_l);

+            av_free(data_ir_r);

+            return AVERROR(ENOMEM);

+        }

+    } 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) {

+            av_free(data_hrtf_l);

+            av_free(data_hrtf_r);

+            return AVERROR(ENOMEM);

+        }

+    }

+

+    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));

+

+        av_freep(&data_ir_l);

+        av_freep(&data_ir_r);

+    } 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]) {

+            av_freep(&data_hrtf_l);

+            av_freep(&data_hrtf_r);

+            av_freep(&fft_in_l);

+            av_freep(&fft_in_r);

+            return AVERROR(ENOMEM);

+        }

+

+        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);

+

+        av_freep(&data_hrtf_l);

+        av_freep(&data_hrtf_r);

+

+        av_freep(&fft_in_l);

+        av_freep(&fft_in_r);

+    }

+

+    s->have_hrirs = 1;

+

+    return 0;

+}

+

+static int filter_frame(AVFilterLink *inlink, AVFrame *in)

+{

+    AVFilterContext *ctx = inlink->dst;

+    HeadphoneContext *s = ctx->priv;

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

+    int ret = 0;

+

+    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 (!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)

+                break;

+        }

+    }

+    return ret;

+}

+

+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;

+

+    AVFilterPad pad = {

+        .name         = "in0",

+        .type         = AVMEDIA_TYPE_AUDIO,

+        .config_props = config_input,

+        .filter_frame = filter_frame,

+    };

+    ff_insert_inpad(ctx, 0, &pad);

+

+    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);

+        ff_insert_inpad(ctx, i, &pad);

+    }

+

+    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,