2011-08-17 13:00:20 +02:00
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/*
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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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2012-05-17 13:10:14 +02:00
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#include "libavutil/avstring.h"
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#include "libavutil/opt.h"
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2012-05-17 16:12:54 +02:00
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#include "libavutil/samplefmt.h"
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#include "libavutil/avassert.h"
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2012-01-21 20:26:31 +01:00
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#include "libswresample/swresample.h"
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2011-08-17 13:00:20 +02:00
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#include "avfilter.h"
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2012-05-12 17:58:14 +02:00
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#include "audio.h"
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2011-08-17 13:00:20 +02:00
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#include "internal.h"
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typedef struct {
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double ratio;
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2012-01-21 20:26:31 +01:00
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struct SwrContext *swr;
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2011-08-17 13:00:20 +02:00
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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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2012-05-17 13:10:14 +02:00
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int ret = 0;
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char *argd = av_strdup(args);
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aresample->swr = swr_alloc();
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if (!aresample->swr)
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return AVERROR(ENOMEM);
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2011-08-17 13:00:20 +02:00
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if (args) {
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2012-05-17 13:10:14 +02:00
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char *ptr=argd, *token;
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while(token = av_strtok(ptr, ":", &ptr)) {
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char *value;
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av_strtok(token, "=", &value);
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if(value) {
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if((ret=av_opt_set(aresample->swr, token, value, 0)) < 0)
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goto end;
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2012-05-17 16:12:54 +02:00
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} else {
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int out_rate;
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if ((ret = ff_parse_sample_rate(&out_rate, token, ctx)) < 0)
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goto end;
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if((ret = av_opt_set_int(aresample->swr, "osr", out_rate, 0)) < 0)
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goto end;
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}
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2012-05-17 13:10:14 +02:00
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}
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2011-08-17 13:00:20 +02:00
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}
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2012-05-17 13:10:14 +02:00
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end:
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av_free(argd);
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return ret;
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2011-08-17 13:00:20 +02:00
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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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2012-01-21 20:26:31 +01:00
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swr_free(&aresample->swr);
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2011-08-17 13:00:20 +02:00
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}
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2012-05-16 02:27:31 +02:00
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static int query_formats(AVFilterContext *ctx)
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{
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AResampleContext *aresample = ctx->priv;
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2012-05-17 16:12:54 +02:00
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int out_rate = av_get_int(aresample->swr, "osr", NULL);
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uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL);
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enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);
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2012-05-16 02:27:31 +02:00
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AVFilterLink *inlink = ctx->inputs[0];
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AVFilterLink *outlink = ctx->outputs[0];
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AVFilterFormats *in_formats = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
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2012-05-17 16:12:54 +02:00
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AVFilterFormats *out_formats;
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2012-05-16 02:27:31 +02:00
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AVFilterFormats *in_samplerates = ff_all_samplerates();
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AVFilterFormats *out_samplerates;
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AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
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2012-05-17 16:12:54 +02:00
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AVFilterChannelLayouts *out_layouts;
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2012-05-16 02:27:31 +02:00
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2012-05-17 16:12:54 +02:00
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avfilter_formats_ref (in_formats, &inlink->out_formats);
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avfilter_formats_ref (in_samplerates, &inlink->out_samplerates);
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ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
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2012-05-16 02:27:31 +02:00
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2012-05-17 16:12:54 +02:00
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if(out_rate > 0) {
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out_samplerates = avfilter_make_format_list((int[]){ out_rate, -1 });
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2012-05-16 02:27:31 +02:00
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} else {
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out_samplerates = ff_all_samplerates();
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}
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2012-05-17 16:12:54 +02:00
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avfilter_formats_ref(out_samplerates, &outlink->in_samplerates);
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if(out_format != AV_SAMPLE_FMT_NONE) {
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out_formats = avfilter_make_format_list((int[]){ out_format, -1 });
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} else
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out_formats = avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO);
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avfilter_formats_ref(out_formats, &outlink->in_formats);
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if(out_layout) {
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out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 });
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} else
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out_layouts = ff_all_channel_layouts();
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ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
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2012-05-16 02:27:31 +02:00
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return 0;
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}
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2011-08-17 13:00:20 +02:00
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static int config_output(AVFilterLink *outlink)
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{
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2012-01-21 20:26:31 +01:00
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int ret;
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2011-08-17 13:00:20 +02:00
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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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2012-05-17 16:12:54 +02:00
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int out_rate;
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uint64_t out_layout;
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enum AVSampleFormat out_format;
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2011-08-17 13:00:20 +02:00
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2012-01-21 20:26:31 +01:00
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aresample->swr = swr_alloc_set_opts(aresample->swr,
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2012-05-16 03:53:29 +02:00
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outlink->channel_layout, outlink->format, outlink->sample_rate,
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2012-01-21 20:26:31 +01:00
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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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2012-05-17 16:12:54 +02:00
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2012-01-21 20:26:31 +01:00
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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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2011-08-17 13:00:20 +02:00
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2012-05-17 16:12:54 +02:00
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out_rate = av_get_int(aresample->swr, "osr", NULL);
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out_layout = av_get_int(aresample->swr, "ocl", NULL);
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out_format = av_get_int(aresample->swr, "osf", NULL);
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outlink->time_base = (AVRational) {1, out_rate};
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av_assert0(outlink->sample_rate == out_rate);
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av_assert0(outlink->channel_layout == out_layout);
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av_assert0(outlink->format == out_format);
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2011-08-17 13:00:20 +02:00
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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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2012-01-21 20:26:31 +01:00
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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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2012-05-10 22:41:29 +02:00
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AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
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2012-01-21 20:26:31 +01:00
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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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2012-05-17 22:38:34 +02:00
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if (n_out <= 0) {
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avfilter_unref_buffer(outsamplesref);
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avfilter_unref_buffer(insamplesref);
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return;
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}
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2012-01-21 20:26:31 +01:00
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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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2012-04-16 01:08:00 +02:00
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outsamplesref->pts = insamplesref->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
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2012-05-16 03:54:04 +02:00
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av_rescale_q(insamplesref->pts, inlink->time_base, outlink->time_base);
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2012-01-21 20:26:31 +01:00
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2012-05-10 22:41:29 +02:00
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ff_filter_samples(outlink, outsamplesref);
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2011-08-17 13:00:20 +02:00
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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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2012-05-16 02:27:31 +02:00
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.query_formats = query_formats,
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2011-08-17 13:00:20 +02:00
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.priv_size = sizeof(AResampleContext),
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2011-11-05 14:39:24 +01:00
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.inputs = (const AVFilterPad[]) {{ .name = "default",
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2011-08-17 13:00:20 +02:00
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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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2011-11-05 14:39:24 +01:00
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.outputs = (const AVFilterPad[]) {{ .name = "default",
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2011-08-17 13:00:20 +02:00
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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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