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