* 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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| /**
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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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| 
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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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| 
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| #define NUMTAPS 64
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| 
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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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| 
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| typedef struct {
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|     int16_t taps[NUMTAPS * 2];
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| } EarwaxContext;
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| 
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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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| 
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|     AVFilterFormats *formats = NULL;
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|     AVFilterChannelLayouts *layout = NULL;
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| 
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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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| 
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|     return 0;
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| }
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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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| 
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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 = 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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| 
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|     return out;
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| }
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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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| 
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|     avfilter_copy_buffer_ref_props(outsamples, insamples);
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| 
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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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| 
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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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| 
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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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| 
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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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| 
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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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| 
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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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| 
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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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