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/* |
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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 64 |
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static const int8_t filt[NUMTAPS] = { |
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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 taps[NUMTAPS * 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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✓✗✓✗
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if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16 )) < 0 || |
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✓✗ |
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(ret = ff_set_common_formats (ctx , formats )) < 0 || |
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✓✗ |
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(ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO )) < 0 || |
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✗✓ |
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(ret = ff_set_common_channel_layouts (ctx , layout )) < 0 || |
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(ret = ff_set_common_samplerates (ctx , ff_make_format_list(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, 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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✓✓ |
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while (in < endin) { |
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sample = 0; |
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✓✓ |
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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 >> 6); |
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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 filter_frame(AVFilterLink *inlink, AVFrame *insamples) |
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{ |
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AVFilterLink *outlink = inlink->dst->outputs[0]; |
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int16_t *taps, *endin, *in, *out; |
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AVFrame *outsamples = ff_get_audio_buffer(outlink, insamples->nb_samples); |
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int len; |
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✗✓ |
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if (!outsamples) { |
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av_frame_free(&insamples); |
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return AVERROR(ENOMEM); |
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} |
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av_frame_copy_props(outsamples, insamples); |
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taps = ((EarwaxContext *)inlink->dst->priv)->taps; |
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out = (int16_t *)outsamples->data[0]; |
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in = (int16_t *)insamples ->data[0]; |
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len = FFMIN(NUMTAPS, 2*insamples->nb_samples); |
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// copy part of new input and process with saved input |
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memcpy(taps+NUMTAPS, in, len * sizeof(*taps)); |
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out = scalarproduct(taps, taps + len, out); |
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// process current input |
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✓✗ |
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if (2*insamples->nb_samples >= NUMTAPS ){ |
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endin = in + insamples->nb_samples * 2 - NUMTAPS; |
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scalarproduct(in, endin, out); |
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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 + 2*insamples->nb_samples, NUMTAPS * sizeof(*taps)); |
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av_frame_free(&insamples); |
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return ff_filter_frame(outlink, outsamples); |
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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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}, |
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{ NULL } |
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}; |
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static const AVFilterPad earwax_outputs[] = { |
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{ |
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.name = "default", |
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.type = AVMEDIA_TYPE_AUDIO, |
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}, |
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{ NULL } |
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}; |
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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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.query_formats = query_formats, |
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.priv_size = sizeof(EarwaxContext), |
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.inputs = earwax_inputs, |
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.outputs = earwax_outputs, |
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}; |