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FFmpeg/libavfilter/af_earwax.c
Ganesh Ajjanagadde 6aaac24d72 avfilter/all: propagate errors of functions from avfilter/formats
Many of the functions from avfilter/formats can return errors, usually AVERROR(ENOMEM).
This propagates the return values.

All of these were found by using av_warn_unused_result, demonstrating its utility.

Tested with FATE. I am least sure of the changes to avfilter/filtergraph,
since I don't know what/how reduce_format is intended to behave and how it should
react to errors.

Fixes: CID 1325680, 1325679, 1325678.

Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Previous version Reviewed-by: Nicolas George <george@nsup.org>
Previous version Reviewed-by: Clément Bœsch <u@pkh.me>
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
2015-10-14 10:04:01 -04:00

175 lines
6.3 KiB
C

/*
* Copyright (c) 2011 Mina Nagy Zaki
* Copyright (c) 2000 Edward Beingessner And Sundry Contributors.
* This source code is freely redistributable and may be used for any purpose.
* This copyright notice must be maintained. Edward Beingessner And Sundry
* Contributors are not responsible for the consequences of using this
* software.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Stereo Widening Effect. Adds audio cues to move stereo image in
* front of the listener. Adapted from the libsox earwax effect.
*/
#include "libavutil/channel_layout.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"
#define NUMTAPS 64
static const int8_t filt[NUMTAPS] = {
/* 30° 330° */
4, -6, /* 32 tap stereo FIR filter. */
4, -11, /* One side filters as if the */
-1, -5, /* signal was from 30 degrees */
3, 3, /* from the ear, the other as */
-2, 5, /* if 330 degrees. */
-5, 0,
9, 1,
6, 3, /* Input */
-4, -1, /* Left Right */
-5, -3, /* __________ __________ */
-2, -5, /* | | | | */
-7, 1, /* .---| Hh,0(f) | | Hh,0(f) |---. */
6, -7, /* / |__________| |__________| \ */
30, -29, /* / \ / \ */
12, -3, /* / X \ */
-11, 4, /* / / \ \ */
-3, 7, /* ____V_____ __________V V__________ _____V____ */
-20, 23, /* | | | | | | | | */
2, 0, /* | Hh,30(f) | | Hh,330(f)| | Hh,330(f)| | Hh,30(f) | */
1, -6, /* |__________| |__________| |__________| |__________| */
-14, -5, /* \ ___ / \ ___ / */
15, -18, /* \ / \ / _____ \ / \ / */
6, 7, /* `->| + |<--' / \ `-->| + |<-' */
15, -10, /* \___/ _/ \_ \___/ */
-14, 22, /* \ / \ / \ / */
-7, -2, /* `--->| | | |<---' */
-4, 9, /* \_/ \_/ */
6, -12, /* */
6, -6, /* Headphones */
0, -11,
0, -5,
4, 0};
typedef struct {
int16_t taps[NUMTAPS * 2];
} EarwaxContext;
static int query_formats(AVFilterContext *ctx)
{
static const int sample_rates[] = { 44100, -1 };
int ret;
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layout = NULL;
if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16 )) < 0 ||
(ret = ff_set_common_formats (ctx , formats )) < 0 ||
(ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO )) < 0 ||
(ret = ff_set_common_channel_layouts (ctx , layout )) < 0 ||
(ret = ff_set_common_samplerates (ctx , ff_make_format_list(sample_rates) )) < 0)
return ret;
return 0;
}
//FIXME: replace with DSPContext.scalarproduct_int16
static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, int16_t *out)
{
int32_t sample;
int16_t j;
while (in < endin) {
sample = 0;
for (j = 0; j < NUMTAPS; j++)
sample += in[j] * filt[j];
*out = av_clip_int16(sample >> 6);
out++;
in++;
}
return out;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
{
AVFilterLink *outlink = inlink->dst->outputs[0];
int16_t *taps, *endin, *in, *out;
AVFrame *outsamples = ff_get_audio_buffer(inlink, insamples->nb_samples);
int len;
if (!outsamples) {
av_frame_free(&insamples);
return AVERROR(ENOMEM);
}
av_frame_copy_props(outsamples, insamples);
taps = ((EarwaxContext *)inlink->dst->priv)->taps;
out = (int16_t *)outsamples->data[0];
in = (int16_t *)insamples ->data[0];
len = FFMIN(NUMTAPS, 2*insamples->nb_samples);
// copy part of new input and process with saved input
memcpy(taps+NUMTAPS, in, len * sizeof(*taps));
out = scalarproduct(taps, taps + len, out);
// process current input
if (2*insamples->nb_samples >= NUMTAPS ){
endin = in + insamples->nb_samples * 2 - NUMTAPS;
scalarproduct(in, endin, out);
// save part of input for next round
memcpy(taps, endin, NUMTAPS * sizeof(*taps));
} else
memmove(taps, taps + 2*insamples->nb_samples, NUMTAPS * sizeof(*taps));
av_frame_free(&insamples);
return ff_filter_frame(outlink, outsamples);
}
static const AVFilterPad earwax_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
{ NULL }
};
static const AVFilterPad earwax_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter ff_af_earwax = {
.name = "earwax",
.description = NULL_IF_CONFIG_SMALL("Widen the stereo image."),
.query_formats = query_formats,
.priv_size = sizeof(EarwaxContext),
.inputs = earwax_inputs,
.outputs = earwax_outputs,
};