@@ -2789,6 +2789,47 @@ 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 HRTFs are provided via additional streams.
+
+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.
+
+@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 lfegain
+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[bl],amovie=azi_135_ele_0_DFC.wav[br],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][br][bl][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 +1,803 @@
+/*
+ * 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_coeffs;
+ int eof_coeffs;
+ 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(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);
+ 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->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(&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_coeffs = 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_coeffs && s->eof_coeffs) {
+ ret = convert_coeffs(ctx, inlink);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (s->have_coeffs) {
+ 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;
+ }
+
+ 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("ir%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_coeffs && 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_coeffs = 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,
+};
@@ -105,6 +105,7 @@ static void register_all(void)
REGISTER_FILTER(FIREQUALIZER, firequalizer, af);
REGISTER_FILTER(FLANGER, flanger, af);
REGISTER_FILTER(HDCD, hdcd, af);
+ REGISTER_FILTER(HEADPHONE, headphone, af);
REGISTER_FILTER(HIGHPASS, highpass, af);
REGISTER_FILTER(JOIN, join, af);
REGISTER_FILTER(LADSPA, ladspa, af);
Signed-off-by: Paul B Mahol <onemda@gmail.com> --- doc/filters.texi | 41 +++ libavfilter/Makefile | 1 + libavfilter/af_headphone.c | 803 +++++++++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 4 files changed, 846 insertions(+) create mode 100644 libavfilter/af_headphone.c