* 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>
		
			
				
	
	
		
			259 lines
		
	
	
		
			8.4 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			259 lines
		
	
	
		
			8.4 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
/*
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 * Copyright (c) 2012 Stefano Sabatini
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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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 * audio to video transmedia filter
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 */
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#include "libavutil/audioconvert.h"
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#include "libavutil/opt.h"
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#include "libavutil/parseutils.h"
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#include "avfilter.h"
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#include "formats.h"
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#include "audio.h"
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#include "video.h"
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#include "internal.h"
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typedef struct {
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    const AVClass *class;
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    int w, h;
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    char *rate_str;
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    AVRational rate;
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    int buf_idx;
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    AVFilterBufferRef *outpicref;
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    int req_fullfilled;
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    int n;
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    int sample_count_mod;
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} ShowWavesContext;
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#define OFFSET(x) offsetof(ShowWavesContext, x)
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static const AVOption showwaves_options[] = {
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    { "rate", "set video rate", OFFSET(rate_str), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0 },
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    { "r",    "set video rate", OFFSET(rate_str), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0 },
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    { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "600x240"}, 0, 0 },
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    { "s",    "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "600x240"}, 0, 0 },
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    { "n",    "set how many samples to show in the same point", OFFSET(n), AV_OPT_TYPE_INT, {.dbl = 0}, 0, INT_MAX},
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    { NULL },
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};
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AVFILTER_DEFINE_CLASS(showwaves);
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static av_cold int init(AVFilterContext *ctx, const char *args)
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{
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    ShowWavesContext *showwaves = ctx->priv;
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    int err;
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    showwaves->class = &showwaves_class;
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    av_opt_set_defaults(showwaves);
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    showwaves->buf_idx = 0;
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    if ((err = av_set_options_string(showwaves, args, "=", ":")) < 0) {
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        av_log(ctx, AV_LOG_ERROR, "Error parsing options string: '%s'\n", args);
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        return err;
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    }
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    return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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    ShowWavesContext *showwaves = ctx->priv;
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    av_freep(&showwaves->rate_str);
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    avfilter_unref_bufferp(&showwaves->outpicref);
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}
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static int query_formats(AVFilterContext *ctx)
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{
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    AVFilterFormats *formats = NULL;
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    AVFilterChannelLayouts *layouts = NULL;
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    AVFilterLink *inlink = ctx->inputs[0];
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    AVFilterLink *outlink = ctx->outputs[0];
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    static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
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    static const enum PixelFormat pix_fmts[] = { PIX_FMT_GRAY8, -1 };
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    /* set input audio formats */
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    formats = ff_make_format_list(sample_fmts);
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    if (!formats)
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        return AVERROR(ENOMEM);
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    ff_formats_ref(formats, &inlink->out_formats);
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    layouts = ff_all_channel_layouts();
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    if (!layouts)
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        return AVERROR(ENOMEM);
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    ff_channel_layouts_ref(layouts, &inlink->out_channel_layouts);
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    formats = ff_all_samplerates();
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    if (!formats)
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        return AVERROR(ENOMEM);
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    ff_formats_ref(formats, &inlink->out_samplerates);
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    /* set output video format */
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    formats = ff_make_format_list(pix_fmts);
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    if (!formats)
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        return AVERROR(ENOMEM);
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    ff_formats_ref(formats, &outlink->in_formats);
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    return 0;
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}
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static int config_output(AVFilterLink *outlink)
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{
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    AVFilterContext *ctx = outlink->src;
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    AVFilterLink *inlink = ctx->inputs[0];
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    ShowWavesContext *showwaves = ctx->priv;
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    int err;
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    if (showwaves->n && showwaves->rate_str) {
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        av_log(ctx, AV_LOG_ERROR, "Options 'n' and 'rate' cannot be set at the same time\n");
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        return AVERROR(EINVAL);
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    }
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    if (!showwaves->n) {
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        if (!showwaves->rate_str)
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            showwaves->rate = (AVRational){25,1}; /* set default value */
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        else if ((err = av_parse_video_rate(&showwaves->rate, showwaves->rate_str)) < 0) {
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            av_log(ctx, AV_LOG_ERROR, "Invalid frame rate: '%s'\n", showwaves->rate_str);
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            return err;
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        }
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        showwaves->n = FFMAX(1, ((double)inlink->sample_rate / (showwaves->w * av_q2d(showwaves->rate))) + 0.5);
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    }
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    outlink->w = showwaves->w;
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    outlink->h = showwaves->h;
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    outlink->sample_aspect_ratio = (AVRational){1,1};
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    outlink->frame_rate = av_div_q((AVRational){inlink->sample_rate,showwaves->n},
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                                   (AVRational){showwaves->w,1});
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    av_log(ctx, AV_LOG_INFO, "s:%dx%d r:%f n:%d\n",
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           showwaves->w, showwaves->h, av_q2d(outlink->frame_rate), showwaves->n);
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    return 0;
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}
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inline static void push_frame(AVFilterLink *outlink)
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{
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    ShowWavesContext *showwaves = outlink->src->priv;
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    ff_start_frame(outlink, showwaves->outpicref);
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    ff_draw_slice(outlink, 0, outlink->h, 1);
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    ff_end_frame(outlink);
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    showwaves->req_fullfilled = 1;
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    showwaves->outpicref = NULL;
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    showwaves->buf_idx = 0;
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}
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static int request_frame(AVFilterLink *outlink)
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{
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    ShowWavesContext *showwaves = outlink->src->priv;
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    AVFilterLink *inlink = outlink->src->inputs[0];
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    int ret;
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    showwaves->req_fullfilled = 0;
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    do {
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        ret = ff_request_frame(inlink);
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    } while (!showwaves->req_fullfilled && ret >= 0);
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    if (ret == AVERROR_EOF && showwaves->outpicref)
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        push_frame(outlink);
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    return ret;
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}
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#define MAX_INT16 ((1<<15) -1)
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static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
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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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    ShowWavesContext *showwaves = ctx->priv;
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    const int nb_samples = insamples->audio->nb_samples;
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    AVFilterBufferRef *outpicref = showwaves->outpicref;
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    int linesize = outpicref ? outpicref->linesize[0] : 0;
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    int16_t *p = (int16_t *)insamples->data[0];
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    int nb_channels = av_get_channel_layout_nb_channels(insamples->audio->channel_layout);
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    int i, j, h;
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    const int n = showwaves->n;
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    const int x = 255 / (nb_channels * n); /* multiplication factor, pre-computed to avoid in-loop divisions */
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    /* draw data in the buffer */
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    for (i = 0; i < nb_samples; i++) {
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        if (showwaves->buf_idx == 0 && showwaves->sample_count_mod == 0) {
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            showwaves->outpicref = outpicref =
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                ff_get_video_buffer(outlink, AV_PERM_WRITE|AV_PERM_ALIGN,
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                                    outlink->w, outlink->h);
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            outpicref->video->w = outlink->w;
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            outpicref->video->h = outlink->h;
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            outpicref->pts = insamples->pts +
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                             av_rescale_q((p - (int16_t *)insamples->data[0]) / nb_channels,
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                                          (AVRational){ 1, inlink->sample_rate },
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                                          outlink->time_base);
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            outlink->out_buf = outpicref;
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            linesize = outpicref->linesize[0];
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            memset(outpicref->data[0], 0, showwaves->h*linesize);
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        }
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        for (j = 0; j < nb_channels; j++) {
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            h = showwaves->h/2 - av_rescale(*p++, showwaves->h/2, MAX_INT16);
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            if (h >= 0 && h < outlink->h)
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                *(outpicref->data[0] + showwaves->buf_idx + h * linesize) += x;
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        }
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        showwaves->sample_count_mod++;
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        if (showwaves->sample_count_mod == n) {
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            showwaves->sample_count_mod = 0;
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            showwaves->buf_idx++;
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        }
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        if (showwaves->buf_idx == showwaves->w)
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            push_frame(outlink);
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    }
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    avfilter_unref_buffer(insamples);
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    return 0;
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}
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AVFilter avfilter_avf_showwaves = {
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    .name           = "showwaves",
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    .description    = NULL_IF_CONFIG_SMALL("Convert input audio to a video output."),
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    .init           = init,
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    .uninit         = uninit,
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    .query_formats  = query_formats,
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    .priv_size      = sizeof(ShowWavesContext),
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    .inputs  = (const AVFilterPad[]) {
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        {
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            .name           = "default",
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            .type           = AVMEDIA_TYPE_AUDIO,
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            .filter_samples = filter_samples,
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            .min_perms      = AV_PERM_READ,
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        },
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        { .name = NULL }
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    },
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    .outputs = (const AVFilterPad[]) {
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        {
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            .name           = "default",
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            .type           = AVMEDIA_TYPE_VIDEO,
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            .config_props   = config_output,
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            .request_frame  = request_frame,
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        },
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        { .name = NULL }
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    },
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};
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