mirror of https://github.com/FFmpeg/FFmpeg.git
Signed-off-by: Paul B Mahol <onemda@gmail.com>pull/262/head
Paul B Mahol
8 years ago
parent
aad1b6786e
commit
d4d1fc823f
6 changed files with 858 additions and 1 deletions
@ -0,0 +1,811 @@ |
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/*
|
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* Copyright (C) 2017 Paul B Mahol |
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* Copyright (C) 2013-2015 Andreas Fuchs, Wolfgang Hrauda |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#include <math.h> |
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#include "libavutil/audio_fifo.h" |
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#include "libavutil/avstring.h" |
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#include "libavutil/channel_layout.h" |
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#include "libavutil/float_dsp.h" |
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#include "libavutil/intmath.h" |
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#include "libavutil/opt.h" |
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#include "libavcodec/avfft.h" |
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#include "avfilter.h" |
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#include "internal.h" |
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#include "audio.h" |
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#define TIME_DOMAIN 0 |
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#define FREQUENCY_DOMAIN 1 |
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typedef struct HeadphoneContext { |
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const AVClass *class; |
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char *map; |
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int type; |
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int lfe_channel; |
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int have_hrirs; |
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int eof_hrirs; |
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int64_t pts; |
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int ir_len; |
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int mapping[64]; |
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int nb_inputs; |
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int nb_irs; |
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float gain; |
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float lfe_gain, gain_lfe; |
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float *ringbuffer[2]; |
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int write[2]; |
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int buffer_length; |
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int n_fft; |
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int size; |
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int *delay[2]; |
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float *data_ir[2]; |
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float *temp_src[2]; |
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FFTComplex *temp_fft[2]; |
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FFTContext *fft[2], *ifft[2]; |
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FFTComplex *data_hrtf[2]; |
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AVFloatDSPContext *fdsp; |
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struct headphone_inputs { |
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AVAudioFifo *fifo; |
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AVFrame *frame; |
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int ir_len; |
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int delay_l; |
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int delay_r; |
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int eof; |
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} *in; |
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} HeadphoneContext; |
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static int parse_channel_name(HeadphoneContext *s, int x, char **arg, int *rchannel, char *buf) |
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{ |
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int len, i, channel_id = 0; |
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int64_t layout, layout0; |
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if (sscanf(*arg, "%7[A-Z]%n", buf, &len)) { |
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layout0 = layout = av_get_channel_layout(buf); |
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if (layout == AV_CH_LOW_FREQUENCY) |
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s->lfe_channel = x; |
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for (i = 32; i > 0; i >>= 1) { |
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if (layout >= 1LL << i) { |
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channel_id += i; |
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layout >>= i; |
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} |
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} |
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if (channel_id >= 64 || layout0 != 1LL << channel_id) |
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return AVERROR(EINVAL); |
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*rchannel = channel_id; |
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*arg += len; |
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return 0; |
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} |
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return AVERROR(EINVAL); |
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} |
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static void parse_map(AVFilterContext *ctx) |
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{ |
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HeadphoneContext *s = ctx->priv; |
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char *arg, *tokenizer, *p, *args = av_strdup(s->map); |
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int i; |
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if (!args) |
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return; |
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p = args; |
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s->lfe_channel = -1; |
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s->nb_inputs = 1; |
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for (i = 0; i < 64; i++) { |
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s->mapping[i] = -1; |
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} |
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while ((arg = av_strtok(p, "|", &tokenizer))) { |
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int out_ch_id; |
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char buf[8]; |
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p = NULL; |
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if (parse_channel_name(s, s->nb_inputs - 1, &arg, &out_ch_id, buf)) { |
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av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%s\' as channel name.\n", buf); |
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continue; |
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} |
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s->mapping[s->nb_inputs - 1] = out_ch_id; |
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s->nb_inputs++; |
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} |
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s->nb_irs = s->nb_inputs - 1; |
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av_free(args); |
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} |
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typedef struct ThreadData { |
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AVFrame *in, *out; |
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int *write; |
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int **delay; |
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float **ir; |
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int *n_clippings; |
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float **ringbuffer; |
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float **temp_src; |
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FFTComplex **temp_fft; |
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} ThreadData; |
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static int headphone_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) |
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{ |
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HeadphoneContext *s = ctx->priv; |
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ThreadData *td = arg; |
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AVFrame *in = td->in, *out = td->out; |
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int offset = jobnr; |
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int *write = &td->write[jobnr]; |
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const int *const delay = td->delay[jobnr]; |
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const float *const ir = td->ir[jobnr]; |
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int *n_clippings = &td->n_clippings[jobnr]; |
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float *ringbuffer = td->ringbuffer[jobnr]; |
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float *temp_src = td->temp_src[jobnr]; |
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const int ir_len = s->ir_len; |
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const float *src = (const float *)in->data[0]; |
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float *dst = (float *)out->data[0]; |
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const int in_channels = in->channels; |
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const int buffer_length = s->buffer_length; |
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const uint32_t modulo = (uint32_t)buffer_length - 1; |
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float *buffer[16]; |
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int wr = *write; |
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int read; |
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int i, l; |
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dst += offset; |
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for (l = 0; l < in_channels; l++) { |
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buffer[l] = ringbuffer + l * buffer_length; |
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} |
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for (i = 0; i < in->nb_samples; i++) { |
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const float *temp_ir = ir; |
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*dst = 0; |
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for (l = 0; l < in_channels; l++) { |
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*(buffer[l] + wr) = src[l]; |
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} |
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for (l = 0; l < in_channels; l++) { |
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const float *const bptr = buffer[l]; |
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if (l == s->lfe_channel) { |
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*dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe; |
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temp_ir += FFALIGN(ir_len, 16); |
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continue; |
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} |
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read = (wr - *(delay + l) - (ir_len - 1) + buffer_length) & modulo; |
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if (read + ir_len < buffer_length) { |
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memcpy(temp_src, bptr + read, ir_len * sizeof(*temp_src)); |
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} else { |
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int len = FFMIN(ir_len - (read % ir_len), buffer_length - read); |
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memcpy(temp_src, bptr + read, len * sizeof(*temp_src)); |
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memcpy(temp_src + len, bptr, (ir_len - len) * sizeof(*temp_src)); |
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} |
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dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, ir_len); |
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temp_ir += FFALIGN(ir_len, 16); |
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} |
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if (fabs(*dst) > 1) |
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*n_clippings += 1; |
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dst += 2; |
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src += in_channels; |
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wr = (wr + 1) & modulo; |
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} |
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*write = wr; |
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return 0; |
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} |
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static int headphone_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) |
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{ |
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HeadphoneContext *s = ctx->priv; |
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ThreadData *td = arg; |
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AVFrame *in = td->in, *out = td->out; |
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int offset = jobnr; |
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int *write = &td->write[jobnr]; |
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FFTComplex *hrtf = s->data_hrtf[jobnr]; |
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int *n_clippings = &td->n_clippings[jobnr]; |
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float *ringbuffer = td->ringbuffer[jobnr]; |
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const int ir_len = s->ir_len; |
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const float *src = (const float *)in->data[0]; |
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float *dst = (float *)out->data[0]; |
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const int in_channels = in->channels; |
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const int buffer_length = s->buffer_length; |
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const uint32_t modulo = (uint32_t)buffer_length - 1; |
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FFTComplex *fft_in = s->temp_fft[jobnr]; |
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FFTContext *ifft = s->ifft[jobnr]; |
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FFTContext *fft = s->fft[jobnr]; |
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const int n_fft = s->n_fft; |
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const float fft_scale = 1.0f / s->n_fft; |
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FFTComplex *hrtf_offset; |
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int wr = *write; |
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int n_read; |
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int i, j; |
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dst += offset; |
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n_read = FFMIN(s->ir_len, in->nb_samples); |
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for (j = 0; j < n_read; j++) { |
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dst[2 * j] = ringbuffer[wr]; |
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ringbuffer[wr] = 0.0; |
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wr = (wr + 1) & modulo; |
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} |
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for (j = n_read; j < in->nb_samples; j++) { |
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dst[2 * j] = 0; |
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} |
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for (i = 0; i < in_channels; i++) { |
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if (i == s->lfe_channel) { |
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for (j = 0; j < in->nb_samples; j++) { |
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dst[2 * j] += src[i + j * in_channels] * s->gain_lfe; |
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} |
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continue; |
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} |
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offset = i * n_fft; |
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hrtf_offset = hrtf + offset; |
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memset(fft_in, 0, sizeof(FFTComplex) * n_fft); |
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for (j = 0; j < in->nb_samples; j++) { |
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fft_in[j].re = src[j * in_channels + i]; |
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} |
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av_fft_permute(fft, fft_in); |
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av_fft_calc(fft, fft_in); |
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for (j = 0; j < n_fft; j++) { |
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const FFTComplex *hcomplex = hrtf_offset + j; |
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const float re = fft_in[j].re; |
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const float im = fft_in[j].im; |
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fft_in[j].re = re * hcomplex->re - im * hcomplex->im; |
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fft_in[j].im = re * hcomplex->im + im * hcomplex->re; |
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} |
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av_fft_permute(ifft, fft_in); |
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av_fft_calc(ifft, fft_in); |
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for (j = 0; j < in->nb_samples; j++) { |
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dst[2 * j] += fft_in[j].re * fft_scale; |
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} |
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for (j = 0; j < ir_len - 1; j++) { |
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int write_pos = (wr + j) & modulo; |
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*(ringbuffer + write_pos) += fft_in[in->nb_samples + j].re * fft_scale; |
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} |
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} |
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for (i = 0; i < out->nb_samples; i++) { |
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if (fabs(*dst) > 1) { |
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n_clippings[0]++; |
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} |
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dst += 2; |
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} |
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*write = wr; |
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return 0; |
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} |
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static int read_ir(AVFilterLink *inlink, AVFrame *frame) |
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{ |
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AVFilterContext *ctx = inlink->dst; |
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HeadphoneContext *s = ctx->priv; |
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int ir_len, max_ir_len, input_number; |
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for (input_number = 0; input_number < s->nb_inputs; input_number++) |
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if (inlink == ctx->inputs[input_number]) |
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break; |
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av_audio_fifo_write(s->in[input_number].fifo, (void **)frame->extended_data, |
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frame->nb_samples); |
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av_frame_free(&frame); |
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ir_len = av_audio_fifo_size(s->in[input_number].fifo); |
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max_ir_len = 4096; |
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if (ir_len > max_ir_len) { |
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av_log(ctx, AV_LOG_ERROR, "Too big length of IRs: %d > %d.\n", ir_len, max_ir_len); |
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return AVERROR(EINVAL); |
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} |
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s->in[input_number].ir_len = ir_len; |
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s->ir_len = FFMAX(ir_len, s->ir_len); |
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return 0; |
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} |
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static int headphone_frame(HeadphoneContext *s, AVFilterLink *outlink) |
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{ |
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AVFilterContext *ctx = outlink->src; |
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AVFrame *in = s->in[0].frame; |
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int n_clippings[2] = { 0 }; |
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ThreadData td; |
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AVFrame *out; |
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av_audio_fifo_read(s->in[0].fifo, (void **)in->extended_data, s->size); |
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out = ff_get_audio_buffer(outlink, in->nb_samples); |
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if (!out) { |
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av_frame_free(&in); |
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return AVERROR(ENOMEM); |
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} |
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out->pts = s->pts; |
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if (s->pts != AV_NOPTS_VALUE) |
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s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); |
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td.in = in; td.out = out; td.write = s->write; |
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td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings; |
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td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src; |
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td.temp_fft = s->temp_fft; |
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if (s->type == TIME_DOMAIN) { |
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ctx->internal->execute(ctx, headphone_convolute, &td, NULL, 2); |
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} else { |
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ctx->internal->execute(ctx, headphone_fast_convolute, &td, NULL, 2); |
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} |
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emms_c(); |
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if (n_clippings[0] + n_clippings[1] > 0) { |
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av_log(ctx, AV_LOG_WARNING, "%d of %d samples clipped. Please reduce gain.\n", |
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n_clippings[0] + n_clippings[1], out->nb_samples * 2); |
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} |
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return ff_filter_frame(outlink, out); |
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} |
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static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink) |
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{ |
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struct HeadphoneContext *s = ctx->priv; |
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const int ir_len = s->ir_len; |
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int nb_irs = s->nb_irs; |
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int nb_input_channels = ctx->inputs[0]->channels; |
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float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10); |
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FFTComplex *data_hrtf_l = NULL; |
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FFTComplex *data_hrtf_r = NULL; |
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FFTComplex *fft_in_l = NULL; |
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FFTComplex *fft_in_r = NULL; |
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float *data_ir_l = NULL; |
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float *data_ir_r = NULL; |
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int offset = 0; |
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int n_fft; |
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int i, j; |
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s->buffer_length = 1 << (32 - ff_clz(s->ir_len)); |
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s->n_fft = n_fft = 1 << (32 - ff_clz(s->ir_len + inlink->sample_rate)); |
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if (s->type == FREQUENCY_DOMAIN) { |
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fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l)); |
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fft_in_r = av_calloc(n_fft, sizeof(*fft_in_r)); |
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if (!fft_in_l || !fft_in_r) { |
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return AVERROR(ENOMEM); |
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} |
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av_fft_end(s->fft[0]); |
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av_fft_end(s->fft[1]); |
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s->fft[0] = av_fft_init(log2(s->n_fft), 0); |
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s->fft[1] = av_fft_init(log2(s->n_fft), 0); |
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av_fft_end(s->ifft[0]); |
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av_fft_end(s->ifft[1]); |
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s->ifft[0] = av_fft_init(log2(s->n_fft), 1); |
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s->ifft[1] = av_fft_init(log2(s->n_fft), 1); |
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if (!s->fft[0] || !s->fft[1] || !s->ifft[0] || !s->ifft[1]) { |
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av_log(ctx, AV_LOG_ERROR, "Unable to create FFT contexts of size %d.\n", s->n_fft); |
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return AVERROR(ENOMEM); |
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} |
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} |
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s->data_ir[0] = av_calloc(FFALIGN(s->ir_len, 16), sizeof(float) * s->nb_irs); |
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s->data_ir[1] = av_calloc(FFALIGN(s->ir_len, 16), sizeof(float) * s->nb_irs); |
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s->delay[0] = av_malloc_array(s->nb_irs, sizeof(float)); |
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s->delay[1] = av_malloc_array(s->nb_irs, sizeof(float)); |
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if (s->type == TIME_DOMAIN) { |
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s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels); |
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s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels); |
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} else { |
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s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float)); |
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s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float)); |
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s->temp_fft[0] = av_malloc_array(s->n_fft, sizeof(FFTComplex)); |
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s->temp_fft[1] = av_malloc_array(s->n_fft, sizeof(FFTComplex)); |
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if (!s->temp_fft[0] || !s->temp_fft[1]) |
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return AVERROR(ENOMEM); |
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} |
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if (!s->data_ir[0] || !s->data_ir[1] || |
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!s->ringbuffer[0] || !s->ringbuffer[1]) |
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return AVERROR(ENOMEM); |
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s->in[0].frame = ff_get_audio_buffer(ctx->inputs[0], s->size); |
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if (!s->in[0].frame) |
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return AVERROR(ENOMEM); |
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for (i = 0; i < s->nb_irs; i++) { |
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s->in[i + 1].frame = ff_get_audio_buffer(ctx->inputs[i + 1], s->ir_len); |
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if (!s->in[i + 1].frame) |
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return AVERROR(ENOMEM); |
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} |
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if (s->type == TIME_DOMAIN) { |
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s->temp_src[0] = av_calloc(FFALIGN(ir_len, 16), sizeof(float)); |
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s->temp_src[1] = av_calloc(FFALIGN(ir_len, 16), sizeof(float)); |
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data_ir_l = av_calloc(nb_irs * FFALIGN(ir_len, 16), sizeof(*data_ir_l)); |
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data_ir_r = av_calloc(nb_irs * FFALIGN(ir_len, 16), sizeof(*data_ir_r)); |
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if (!data_ir_r || !data_ir_l || !s->temp_src[0] || !s->temp_src[1]) { |
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av_free(data_ir_l); |
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av_free(data_ir_r); |
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return AVERROR(ENOMEM); |
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} |
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} else { |
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data_hrtf_l = av_malloc_array(n_fft, sizeof(*data_hrtf_l) * nb_irs); |
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data_hrtf_r = av_malloc_array(n_fft, sizeof(*data_hrtf_r) * nb_irs); |
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if (!data_hrtf_r || !data_hrtf_l) { |
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av_free(data_hrtf_l); |
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av_free(data_hrtf_r); |
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return AVERROR(ENOMEM); |
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} |
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} |
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|
||||
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, |
||||
}; |
||||
Loading…
Reference in new issue