libavfilter/af_apad.c
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 /*
  * Copyright (c) 2012 Michael Niedermayer
  *
  * 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
  * audio pad filter.
  *
  * Based on af_aresample.c
  */
 
 #include "libavutil/avstring.h"
 #include "libavutil/channel_layout.h"
 #include "libavutil/opt.h"
 #include "libavutil/samplefmt.h"
 #include "libavutil/avassert.h"
 #include "avfilter.h"
 #include "audio.h"
 #include "internal.h"
 
 typedef struct {
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     const AVClass *class;
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     int64_t next_pts;
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     int packet_size;
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     int64_t pad_len;
     int64_t whole_len;
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 } APadContext;
 
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 #define OFFSET(x) offsetof(APadContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
 
 static const AVOption apad_options[] = {
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     { "packet_size", "set silence packet size", OFFSET(packet_size), AV_OPT_TYPE_INT, { .i64 = 4096 }, 0, INT_MAX, A },
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     { "pad_len",     "number of samples of silence to add",          OFFSET(pad_len),   AV_OPT_TYPE_INT64, { .i64 = 0 }, 0, INT64_MAX, A },
     { "whole_len",   "target number of samples in the audio stream", OFFSET(whole_len), AV_OPT_TYPE_INT64, { .i64 = 0 }, 0, INT64_MAX, A },
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     { NULL },
 };
 
 AVFILTER_DEFINE_CLASS(apad);
 
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 static av_cold int init(AVFilterContext *ctx, const char *args)
 {
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     int ret;
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     APadContext *apad = ctx->priv;
 
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     apad->class = &apad_class;
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     apad->next_pts = AV_NOPTS_VALUE;
 
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     av_opt_set_defaults(apad);
 
     if ((ret = av_opt_set_from_string(apad, args, NULL, "=", ":")) < 0)
         return ret;
 
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     if (apad->whole_len && apad->pad_len) {
         av_log(ctx, AV_LOG_ERROR, "Both whole and pad length are set, this is not possible\n");
         return AVERROR(EINVAL);
     }
 
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     return 0;
 }
 
 static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *frame)
 {
     AVFilterContext *ctx = inlink->dst;
     APadContext *apad = ctx->priv;
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     if (apad->whole_len)
         apad->whole_len -= frame->audio->nb_samples;
 
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     apad->next_pts = frame->pts + av_rescale_q(frame->audio->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
     return ff_filter_frame(ctx->outputs[0], frame);
 }
 
 static int request_frame(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     APadContext *apad = ctx->priv;
     int ret;
 
     ret = ff_request_frame(ctx->inputs[0]);
 
     if (ret == AVERROR_EOF) {
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         int n_out = apad->packet_size;
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         AVFilterBufferRef *outsamplesref;
 
         if (apad->whole_len > 0) {
             apad->pad_len = apad->whole_len;
             apad->whole_len = 0;
         }
         if (apad->pad_len > 0) {
             n_out = FFMIN(n_out, apad->pad_len);
             apad->pad_len -= n_out;
         }
 
         if(!n_out)
             return AVERROR_EOF;
 
         outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
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         if (!outsamplesref)
             return AVERROR(ENOMEM);
 
         av_assert0(outsamplesref->audio->sample_rate == outlink->sample_rate);
         av_assert0(outsamplesref->audio->nb_samples  == n_out);
 
         av_samples_set_silence(outsamplesref->extended_data, 0,
                                n_out,
                                outsamplesref->audio->channels,
                                outsamplesref->format);
 
         outsamplesref->pts = apad->next_pts;
         if (apad->next_pts != AV_NOPTS_VALUE)
             apad->next_pts += av_rescale_q(n_out, (AVRational){1, outlink->sample_rate}, outlink->time_base);
 
         return ff_filter_frame(outlink, outsamplesref);
     }
     return ret;
 }
 
 static const AVFilterPad apad_inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .filter_frame = filter_frame,
         .min_perms    = AV_PERM_READ,
     },
     { NULL },
 };
 
 static const AVFilterPad apad_outputs[] = {
     {
         .name          = "default",
         .request_frame = request_frame,
         .type          = AVMEDIA_TYPE_AUDIO,
     },
     { NULL },
 };
 
 AVFilter avfilter_af_apad = {
     .name          = "apad",
     .description   = NULL_IF_CONFIG_SMALL("Pad audio with silence."),
     .init          = init,
     .priv_size     = sizeof(APadContext),
     .inputs        = apad_inputs,
     .outputs       = apad_outputs,
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     .priv_class    = &apad_class,
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 };