234 lines
8.0 KiB
C
234 lines
8.0 KiB
C
/*
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* Copyright (c) 2011 Mina Nagy Zaki
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* Copyright (c) 2000 Edward Beingessner And Sundry Contributors.
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* This source code is freely redistributable and may be used for any purpose.
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* This copyright notice must be maintained. Edward Beingessner And Sundry
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* Contributors are not responsible for the consequences of using this
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* software.
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*
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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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/**
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* @file
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* Stereo Widening Effect. Adds audio cues to move stereo image in
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* front of the listener. Adapted from the libsox earwax effect.
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*/
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#include "libavutil/channel_layout.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "formats.h"
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#define NUMTAPS 32
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static const int8_t filt[NUMTAPS * 2] = {
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/* 30° 330° */
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4, -6, /* 32 tap stereo FIR filter. */
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4, -11, /* One side filters as if the */
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-1, -5, /* signal was from 30 degrees */
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3, 3, /* from the ear, the other as */
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-2, 5, /* if 330 degrees. */
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-5, 0,
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9, 1,
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6, 3, /* Input */
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-4, -1, /* Left Right */
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-5, -3, /* __________ __________ */
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-2, -5, /* | | | | */
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-7, 1, /* .---| Hh,0(f) | | Hh,0(f) |---. */
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6, -7, /* / |__________| |__________| \ */
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30, -29, /* / \ / \ */
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12, -3, /* / X \ */
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-11, 4, /* / / \ \ */
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-3, 7, /* ____V_____ __________V V__________ _____V____ */
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-20, 23, /* | | | | | | | | */
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2, 0, /* | Hh,30(f) | | Hh,330(f)| | Hh,330(f)| | Hh,30(f) | */
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1, -6, /* |__________| |__________| |__________| |__________| */
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-14, -5, /* \ ___ / \ ___ / */
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15, -18, /* \ / \ / _____ \ / \ / */
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6, 7, /* `->| + |<--' / \ `-->| + |<-' */
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15, -10, /* \___/ _/ \_ \___/ */
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-14, 22, /* \ / \ / \ / */
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-7, -2, /* `--->| | | |<---' */
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-4, 9, /* \_/ \_/ */
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6, -12, /* */
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6, -6, /* Headphones */
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0, -11,
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0, -5,
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4, 0};
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typedef struct EarwaxContext {
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int16_t filter[2][NUMTAPS];
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int16_t taps[4][NUMTAPS * 2];
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AVFrame *frame[2];
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} EarwaxContext;
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static int query_formats(AVFilterContext *ctx)
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{
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static const int sample_rates[] = { 44100, -1 };
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int ret;
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AVFilterFormats *formats = NULL;
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AVFilterChannelLayouts *layout = NULL;
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if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16P )) < 0 ||
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(ret = ff_set_common_formats (ctx , formats )) < 0 ||
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(ret = ff_add_channel_layout (&layout , &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO)) < 0 ||
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(ret = ff_set_common_channel_layouts (ctx , layout )) < 0 ||
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(ret = ff_set_common_samplerates_from_list(ctx, sample_rates)) < 0)
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return ret;
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return 0;
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}
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//FIXME: replace with DSPContext.scalarproduct_int16
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static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin,
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const int16_t *filt, int16_t *out)
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{
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int32_t sample;
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int16_t j;
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while (in < endin) {
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sample = 0;
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for (j = 0; j < NUMTAPS; j++)
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sample += in[j] * filt[j];
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*out = av_clip_int16(sample >> 7);
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out++;
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in++;
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}
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return out;
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}
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static int config_input(AVFilterLink *inlink)
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{
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EarwaxContext *s = inlink->dst->priv;
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for (int i = 0; i < NUMTAPS; i++) {
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s->filter[0][i] = filt[i * 2];
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s->filter[1][i] = filt[i * 2 + 1];
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}
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return 0;
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}
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static void convolve(AVFilterContext *ctx, AVFrame *in,
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int input_ch, int output_ch,
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int filter_ch, int tap_ch)
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{
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EarwaxContext *s = ctx->priv;
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int16_t *taps, *endin, *dst, *src;
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int len;
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taps = s->taps[tap_ch];
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dst = (int16_t *)s->frame[input_ch]->data[output_ch];
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src = (int16_t *)in->data[input_ch];
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len = FFMIN(NUMTAPS, in->nb_samples);
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// copy part of new input and process with saved input
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memcpy(taps+NUMTAPS, src, len * sizeof(*taps));
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dst = scalarproduct(taps, taps + len, s->filter[filter_ch], dst);
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// process current input
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if (in->nb_samples >= NUMTAPS) {
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endin = src + in->nb_samples - NUMTAPS;
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scalarproduct(src, endin, s->filter[filter_ch], dst);
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// save part of input for next round
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memcpy(taps, endin, NUMTAPS * sizeof(*taps));
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} else {
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memmove(taps, taps + in->nb_samples, NUMTAPS * sizeof(*taps));
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}
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}
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static void mix(AVFilterContext *ctx, AVFrame *out,
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int output_ch, int f0, int f1, int i0, int i1)
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{
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EarwaxContext *s = ctx->priv;
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const int16_t *srcl = (const int16_t *)s->frame[f0]->data[i0];
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const int16_t *srcr = (const int16_t *)s->frame[f1]->data[i1];
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int16_t *dst = (int16_t *)out->data[output_ch];
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for (int n = 0; n < out->nb_samples; n++)
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dst[n] = av_clip_int16(srcl[n] + srcr[n]);
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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EarwaxContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
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for (int ch = 0; ch < 2; ch++) {
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if (!s->frame[ch] || s->frame[ch]->nb_samples < in->nb_samples) {
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av_frame_free(&s->frame[ch]);
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s->frame[ch] = ff_get_audio_buffer(outlink, in->nb_samples);
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if (!s->frame[ch]) {
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av_frame_free(&in);
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av_frame_free(&out);
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return AVERROR(ENOMEM);
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}
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}
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}
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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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av_frame_copy_props(out, in);
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convolve(ctx, in, 0, 0, 0, 0);
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convolve(ctx, in, 0, 1, 1, 1);
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convolve(ctx, in, 1, 0, 0, 2);
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convolve(ctx, in, 1, 1, 1, 3);
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mix(ctx, out, 0, 0, 1, 1, 0);
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mix(ctx, out, 1, 0, 1, 0, 1);
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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EarwaxContext *s = ctx->priv;
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av_frame_free(&s->frame[0]);
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av_frame_free(&s->frame[1]);
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}
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static const AVFilterPad earwax_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.config_props = config_input,
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},
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};
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const AVFilter ff_af_earwax = {
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.name = "earwax",
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.description = NULL_IF_CONFIG_SMALL("Widen the stereo image."),
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.priv_size = sizeof(EarwaxContext),
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.uninit = uninit,
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FILTER_INPUTS(earwax_inputs),
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FILTER_OUTPUTS(ff_audio_default_filterpad),
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FILTER_QUERY_FUNC(query_formats),
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};
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