Up until now, an AVFilter's lists of input and output AVFilterPads were terminated by a sentinel and the only way to get the length of these lists was by using avfilter_pad_count(). This has two drawbacks: first, sizeof(AVFilterPad) is not negligible (i.e. 64B on 64bit systems); second, getting the size involves a function call instead of just reading the data. This commit therefore changes this. The sentinels are removed and new private fields nb_inputs and nb_outputs are added to AVFilter that contain the number of elements of the respective AVFilterPad array. Given that AVFilter.(in|out)puts are the only arrays of zero-terminated AVFilterPads an API user has access to (AVFilterContext.(in|out)put_pads are not zero-terminated and they already have a size field) the argument to avfilter_pad_count() is always one of these lists, so it just has to find the filter the list belongs to and read said number. This is slower than before, but a replacement function that just reads the internal numbers that users are expected to switch to will be added soon; and furthermore, avfilter_pad_count() is probably never called in hot loops anyway. This saves about 49KiB from the binary; notice that these sentinels are not in .bss despite being zeroed: they are in .data.rel.ro due to the non-sentinels. Reviewed-by: Nicolas George <george@nsup.org> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
		
			
				
	
	
		
			246 lines
		
	
	
		
			7.2 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			246 lines
		
	
	
		
			7.2 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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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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| #include "libavutil/channel_layout.h"
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| #include "libavutil/ffmath.h"
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| #include "libavutil/opt.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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| typedef struct ASubBoostContext {
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|     const AVClass *class;
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| 
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|     double dry_gain;
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|     double wet_gain;
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|     double feedback;
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|     double decay;
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|     double delay;
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|     double cutoff;
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|     double slope;
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| 
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|     double a0, a1, a2;
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|     double b0, b1, b2;
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| 
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|     int *write_pos;
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|     int buffer_samples;
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| 
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|     AVFrame *w;
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|     AVFrame *buffer;
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| } ASubBoostContext;
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| 
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| static int query_formats(AVFilterContext *ctx)
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| {
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|     static const enum AVSampleFormat sample_fmts[] = {
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|         AV_SAMPLE_FMT_DBLP,
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|         AV_SAMPLE_FMT_NONE
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|     };
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|     int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
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|     if (ret < 0)
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|         return ret;
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| 
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|     ret = ff_set_common_all_channel_counts(ctx);
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|     if (ret < 0)
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|         return ret;
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| 
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|     return ff_set_common_all_samplerates(ctx);
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| }
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| 
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| static int get_coeffs(AVFilterContext *ctx)
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| {
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|     ASubBoostContext *s = ctx->priv;
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|     AVFilterLink *inlink = ctx->inputs[0];
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|     double w0 = 2 * M_PI * s->cutoff / inlink->sample_rate;
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|     double alpha = sin(w0) / 2 * sqrt(2. * (1. / s->slope - 1.) + 2.);
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| 
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|     s->a0 =  1 + alpha;
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|     s->a1 = -2 * cos(w0);
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|     s->a2 =  1 - alpha;
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|     s->b0 = (1 - cos(w0)) / 2;
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|     s->b1 =  1 - cos(w0);
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|     s->b2 = (1 - cos(w0)) / 2;
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| 
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|     s->a1 /= s->a0;
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|     s->a2 /= s->a0;
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|     s->b0 /= s->a0;
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|     s->b1 /= s->a0;
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|     s->b2 /= s->a0;
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| 
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|     s->buffer_samples = inlink->sample_rate * s->delay / 1000;
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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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|     AVFilterContext *ctx = inlink->dst;
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|     ASubBoostContext *s = ctx->priv;
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| 
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|     s->buffer = ff_get_audio_buffer(inlink, inlink->sample_rate / 10);
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|     s->w = ff_get_audio_buffer(inlink, 2);
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|     s->write_pos = av_calloc(inlink->channels, sizeof(*s->write_pos));
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|     if (!s->buffer || !s->w || !s->write_pos)
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|         return AVERROR(ENOMEM);
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| 
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|     return get_coeffs(ctx);
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| }
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| 
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| typedef struct ThreadData {
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|     AVFrame *in, *out;
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| } ThreadData;
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| 
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| static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
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| {
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|     ASubBoostContext *s = ctx->priv;
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|     ThreadData *td = arg;
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|     AVFrame *out = td->out;
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|     AVFrame *in = td->in;
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|     const double mix = ctx->is_disabled ? 0. : 1.;
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|     const double wet = ctx->is_disabled ? 1. : s->wet_gain;
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|     const double dry = ctx->is_disabled ? 1. : s->dry_gain;
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|     const double feedback = s->feedback, decay = s->decay;
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|     const double b0 = s->b0;
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|     const double b1 = s->b1;
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|     const double b2 = s->b2;
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|     const double a1 = -s->a1;
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|     const double a2 = -s->a2;
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|     const int start = (in->channels * jobnr) / nb_jobs;
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|     const int end = (in->channels * (jobnr+1)) / nb_jobs;
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|     const int buffer_samples = s->buffer_samples;
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| 
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|     for (int ch = start; ch < end; ch++) {
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|         const double *src = (const double *)in->extended_data[ch];
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|         double *dst = (double *)out->extended_data[ch];
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|         double *buffer = (double *)s->buffer->extended_data[ch];
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|         double *w = (double *)s->w->extended_data[ch];
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|         int write_pos = s->write_pos[ch];
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| 
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|         for (int n = 0; n < in->nb_samples; n++) {
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|             double out_sample;
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| 
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|             out_sample = src[n] * b0 + w[0];
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|             w[0] = b1 * src[n] + w[1] + a1 * out_sample;
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|             w[1] = b2 * src[n] + a2 * out_sample;
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| 
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|             buffer[write_pos] = buffer[write_pos] * decay + out_sample * feedback;
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|             dst[n] = (src[n] * dry + buffer[write_pos] * mix) * wet;
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| 
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|             if (++write_pos >= buffer_samples)
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|                 write_pos = 0;
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|         }
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| 
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|         s->write_pos[ch] = write_pos;
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|     }
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| 
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|     return 0;
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| }
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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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|     AVFilterLink *outlink = ctx->outputs[0];
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|     ThreadData td;
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|     AVFrame *out;
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| 
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|     if (av_frame_is_writable(in)) {
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|         out = in;
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|     } else {
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|         out = ff_get_audio_buffer(outlink, in->nb_samples);
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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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|     }
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| 
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|     td.in = in; td.out = out;
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|     ff_filter_execute(ctx, filter_channels, &td, NULL,
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|                       FFMIN(inlink->channels, ff_filter_get_nb_threads(ctx)));
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| 
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|     if (out != in)
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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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| 
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| static av_cold void uninit(AVFilterContext *ctx)
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| {
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|     ASubBoostContext *s = ctx->priv;
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| 
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|     av_frame_free(&s->buffer);
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|     av_frame_free(&s->w);
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|     av_freep(&s->write_pos);
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| }
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| 
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| static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
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|                            char *res, int res_len, int flags)
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| {
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|     int ret;
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| 
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|     ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
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|     if (ret < 0)
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|         return ret;
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| 
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|     return get_coeffs(ctx);
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| }
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| 
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| #define OFFSET(x) offsetof(ASubBoostContext, x)
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| #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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| 
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| static const AVOption asubboost_options[] = {
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|     { "dry",      "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7},      0,   1, FLAGS },
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|     { "wet",      "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7},      0,   1, FLAGS },
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|     { "decay",    "set decay",    OFFSET(decay),    AV_OPT_TYPE_DOUBLE, {.dbl=0.7},      0,   1, FLAGS },
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|     { "feedback", "set feedback", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl=0.9},      0,   1, FLAGS },
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|     { "cutoff",   "set cutoff",   OFFSET(cutoff),   AV_OPT_TYPE_DOUBLE, {.dbl=100},     50, 900, FLAGS },
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|     { "slope",    "set slope",    OFFSET(slope),    AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0001,   1, FLAGS },
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|     { "delay",    "set delay",    OFFSET(delay),    AV_OPT_TYPE_DOUBLE, {.dbl=20},       1, 100, FLAGS },
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|     { NULL }
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| };
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| 
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| AVFILTER_DEFINE_CLASS(asubboost);
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| 
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| static const AVFilterPad 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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| 
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| static const AVFilterPad 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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| };
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| 
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| const AVFilter ff_af_asubboost = {
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|     .name           = "asubboost",
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|     .description    = NULL_IF_CONFIG_SMALL("Boost subwoofer frequencies."),
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|     .query_formats  = query_formats,
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|     .priv_size      = sizeof(ASubBoostContext),
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|     .priv_class     = &asubboost_class,
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|     .uninit         = uninit,
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|     FILTER_INPUTS(inputs),
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|     FILTER_OUTPUTS(outputs),
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|     .process_command = process_command,
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|     .flags           = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
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|                        AVFILTER_FLAG_SLICE_THREADS,
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| };
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