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f8911b987d
* qatar/master: mss3: use standard zigzag table mss3: split DSP functions that are used in MTS2(MSS4) into separate file motion-test: do not use getopt() tcp: add initial timeout limit for incoming connections configure: Change the rdtsc check to a linker check avconv: propagate fatal errors from lavfi. lavfi: add error handling to filter_samples(). fate-run: make avconv() properly deal with multiple inputs. asplit: don't leak the input buffer. af_resample: fix request_frame() behavior. af_asyncts: fix request_frame() behavior. libx264: support aspect ratio switching matroskadec: honor error_recognition when encountering unknown elements. lavr: resampling: add support for s32p, fltp, and dblp internal sample formats lavr: resampling: add filter type and Kaiser window beta to AVOptions lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format lavr: mix: validate internal sample format in ff_audio_mix_init() Conflicts: ffmpeg.c ffplay.c libavcodec/libx264.c libavfilter/audio.c libavfilter/split.c libavformat/tcp.c tests/fate-run.sh Merged-by: Michael Niedermayer <michaelni@gmx.at>
170 lines
6.4 KiB
C
170 lines
6.4 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/audioconvert.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 {
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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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int sample_rates[] = { 44100, -1 };
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AVFilterFormats *formats = NULL;
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AVFilterChannelLayouts *layout = NULL;
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ff_add_format(&formats, AV_SAMPLE_FMT_S16);
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ff_set_common_formats(ctx, formats);
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ff_add_channel_layout(&layout, AV_CH_LAYOUT_STEREO);
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ff_set_common_channel_layouts(ctx, layout);
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ff_set_common_samplerates(ctx, ff_make_format_list(sample_rates));
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return 0;
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}
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static int config_input(AVFilterLink *inlink)
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{
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if (inlink->sample_rate != 44100) {
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av_log(inlink->dst, AV_LOG_ERROR,
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"The earwax filter only works for 44.1kHz audio. Insert "
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"a resample filter before this\n");
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return AVERROR(EINVAL);
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}
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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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while (in < endin) {
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sample = 32;
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for (j = 0; j < NUMTAPS; j++)
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sample += in[j] * filt[j];
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*out = 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_samples(AVFilterLink *inlink, AVFilterBufferRef *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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AVFilterBufferRef *outsamples =
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ff_get_audio_buffer(inlink, AV_PERM_WRITE,
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insamples->audio->nb_samples);
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int ret;
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avfilter_copy_buffer_ref_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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// copy part of new input and process with saved input
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memcpy(taps+NUMTAPS, in, NUMTAPS * sizeof(*taps));
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out = scalarproduct(taps, taps + NUMTAPS, out);
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// process current input
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endin = in + insamples->audio->nb_samples * 2 - NUMTAPS;
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out = 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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ret = ff_filter_samples(outlink, outsamples);
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avfilter_unref_buffer(insamples);
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return ret;
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}
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AVFilter avfilter_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 = (const AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples,
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.config_props = config_input,
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.min_perms = AV_PERM_READ, },
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{ .name = NULL}},
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.outputs = (const AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO, },
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{ .name = NULL}},
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};
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