mirror of https://git.ffmpeg.org/ffmpeg.git
127 lines
4.4 KiB
C
127 lines
4.4 KiB
C
/*
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* Copyright (c) 2011 Stefano Sabatini
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* Copyright (c) 2011 Mina Nagy Zaki
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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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* resampling audio filter
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*/
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#include "libswresample/swresample.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct {
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int out_rate;
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double ratio;
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struct SwrContext *swr;
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} AResampleContext;
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static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
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{
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AResampleContext *aresample = ctx->priv;
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int ret;
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if (args) {
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if ((ret = ff_parse_sample_rate(&aresample->out_rate, args, ctx)) < 0)
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return ret;
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} else {
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aresample->out_rate = -1;
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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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AResampleContext *aresample = ctx->priv;
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swr_free(&aresample->swr);
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}
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static int config_output(AVFilterLink *outlink)
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{
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int ret;
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AVFilterContext *ctx = outlink->src;
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AVFilterLink *inlink = ctx->inputs[0];
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AResampleContext *aresample = ctx->priv;
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if (aresample->out_rate == -1)
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aresample->out_rate = outlink->sample_rate;
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else
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outlink->sample_rate = aresample->out_rate;
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outlink->time_base = (AVRational) {1, aresample->out_rate};
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//TODO: make the resampling parameters (filter size, phrase shift, linear, cutoff) configurable
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aresample->swr = swr_alloc_set_opts(aresample->swr,
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inlink->channel_layout, inlink->format, aresample->out_rate,
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inlink->channel_layout, inlink->format, inlink->sample_rate,
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0, ctx);
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if (!aresample->swr)
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return AVERROR(ENOMEM);
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ret = swr_init(aresample->swr);
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if (ret < 0)
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return ret;
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aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
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av_log(ctx, AV_LOG_INFO, "r:%"PRId64"Hz -> r:%"PRId64"Hz\n",
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inlink->sample_rate, outlink->sample_rate);
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return 0;
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}
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static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
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{
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AResampleContext *aresample = inlink->dst->priv;
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const int n_in = insamplesref->audio->nb_samples;
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int n_out = n_in * aresample->ratio;
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AVFilterLink *const outlink = inlink->dst->outputs[0];
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AVFilterBufferRef *outsamplesref = avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
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n_out = swr_convert(aresample->swr, outsamplesref->data, n_out,
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(void *)insamplesref->data, n_in);
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avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
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outsamplesref->audio->sample_rate = outlink->sample_rate;
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outsamplesref->audio->nb_samples = n_out;
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outsamplesref->pts = av_rescale(outlink->sample_rate, insamplesref->pts,
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inlink ->sample_rate);
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avfilter_filter_samples(outlink, outsamplesref);
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avfilter_unref_buffer(insamplesref);
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}
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AVFilter avfilter_af_aresample = {
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.name = "aresample",
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.description = NULL_IF_CONFIG_SMALL("Resample audio data."),
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.init = init,
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.uninit = uninit,
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.priv_size = sizeof(AResampleContext),
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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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.min_perms = AV_PERM_READ, },
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{ .name = NULL}},
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.outputs = (const AVFilterPad[]) {{ .name = "default",
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.config_props = config_output,
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.type = AVMEDIA_TYPE_AUDIO, },
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{ .name = NULL}},
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};
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