/* * 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 "libavutil/avstring.h" #include "libavutil/channel_layout.h" #include "libavutil/opt.h" #include "libavutil/samplefmt.h" #include "libavutil/avassert.h" #include "libswresample/swresample.h" #include "avfilter.h" #include "audio.h" #include "internal.h" typedef struct { const AVClass *class; int sample_rate_arg; double ratio; struct SwrContext *swr; int64_t next_pts; int req_fullfilled; } AResampleContext; static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts) { AResampleContext *aresample = ctx->priv; int ret = 0; aresample->next_pts = AV_NOPTS_VALUE; aresample->swr = swr_alloc(); if (!aresample->swr) { ret = AVERROR(ENOMEM); goto end; } if (opts) { AVDictionaryEntry *e = NULL; while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) { if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0) goto end; } av_dict_free(opts); } if (aresample->sample_rate_arg > 0) av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0); end: return ret; } static av_cold void uninit(AVFilterContext *ctx) { AResampleContext *aresample = ctx->priv; swr_free(&aresample->swr); } static int query_formats(AVFilterContext *ctx) { AResampleContext *aresample = ctx->priv; int out_rate = av_get_int(aresample->swr, "osr", NULL); uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL); enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL); AVFilterLink *inlink = ctx->inputs[0]; AVFilterLink *outlink = ctx->outputs[0]; AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO); AVFilterFormats *out_formats; AVFilterFormats *in_samplerates = ff_all_samplerates(); AVFilterFormats *out_samplerates; AVFilterChannelLayouts *in_layouts = ff_all_channel_counts(); AVFilterChannelLayouts *out_layouts; ff_formats_ref (in_formats, &inlink->out_formats); ff_formats_ref (in_samplerates, &inlink->out_samplerates); ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts); if(out_rate > 0) { out_samplerates = ff_make_format_list((int[]){ out_rate, -1 }); } else { out_samplerates = ff_all_samplerates(); } ff_formats_ref(out_samplerates, &outlink->in_samplerates); if(out_format != AV_SAMPLE_FMT_NONE) { out_formats = ff_make_format_list((int[]){ out_format, -1 }); } else out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO); ff_formats_ref(out_formats, &outlink->in_formats); if(out_layout) { out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 }); } else out_layouts = ff_all_channel_counts(); ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts); return 0; } static int config_output(AVFilterLink *outlink) { int ret; AVFilterContext *ctx = outlink->src; AVFilterLink *inlink = ctx->inputs[0]; AResampleContext *aresample = ctx->priv; int out_rate; uint64_t out_layout; enum AVSampleFormat out_format; char inchl_buf[128], outchl_buf[128]; aresample->swr = swr_alloc_set_opts(aresample->swr, outlink->channel_layout, outlink->format, outlink->sample_rate, inlink->channel_layout, inlink->format, inlink->sample_rate, 0, ctx); if (!aresample->swr) return AVERROR(ENOMEM); if (!inlink->channel_layout) av_opt_set_int(aresample->swr, "ich", inlink->channels, 0); if (!outlink->channel_layout) av_opt_set_int(aresample->swr, "och", outlink->channels, 0); ret = swr_init(aresample->swr); if (ret < 0) return ret; out_rate = av_get_int(aresample->swr, "osr", NULL); out_layout = av_get_int(aresample->swr, "ocl", NULL); out_format = av_get_int(aresample->swr, "osf", NULL); outlink->time_base = (AVRational) {1, out_rate}; av_assert0(outlink->sample_rate == out_rate); av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout); av_assert0(outlink->format == out_format); aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate; av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), inlink ->channels, inlink ->channel_layout); av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout); av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n", inlink ->channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate, outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate); return 0; } static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref) { AResampleContext *aresample = inlink->dst->priv; const int n_in = insamplesref->nb_samples; int n_out = n_in * aresample->ratio * 2 + 256; AVFilterLink *const outlink = inlink->dst->outputs[0]; AVFrame *outsamplesref = ff_get_audio_buffer(outlink, n_out); int ret; if(!outsamplesref) return AVERROR(ENOMEM); av_frame_copy_props(outsamplesref, insamplesref); outsamplesref->format = outlink->format; av_frame_set_channels(outsamplesref, outlink->channels); outsamplesref->channel_layout = outlink->channel_layout; outsamplesref->sample_rate = outlink->sample_rate; if(insamplesref->pts != AV_NOPTS_VALUE) { int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den); int64_t outpts= swr_next_pts(aresample->swr, inpts); aresample->next_pts = outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate); } else { outsamplesref->pts = AV_NOPTS_VALUE; } n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, (void *)insamplesref->extended_data, n_in); if (n_out <= 0) { av_frame_free(&outsamplesref); av_frame_free(&insamplesref); return 0; } outsamplesref->nb_samples = n_out; ret = ff_filter_frame(outlink, outsamplesref); aresample->req_fullfilled= 1; av_frame_free(&insamplesref); return ret; } static int request_frame(AVFilterLink *outlink) { AVFilterContext *ctx = outlink->src; AResampleContext *aresample = ctx->priv; AVFilterLink *const inlink = outlink->src->inputs[0]; int ret; aresample->req_fullfilled = 0; do{ ret = ff_request_frame(ctx->inputs[0]); }while(!aresample->req_fullfilled && ret>=0); if (ret == AVERROR_EOF) { AVFrame *outsamplesref; int n_out = 4096; outsamplesref = ff_get_audio_buffer(outlink, n_out); if (!outsamplesref) return AVERROR(ENOMEM); n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0); if (n_out <= 0) { av_frame_free(&outsamplesref); return (n_out == 0) ? AVERROR_EOF : n_out; } outsamplesref->sample_rate = outlink->sample_rate; outsamplesref->nb_samples = n_out; #if 0 outsamplesref->pts = aresample->next_pts; if(aresample->next_pts != AV_NOPTS_VALUE) aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base); #else outsamplesref->pts = swr_next_pts(aresample->swr, INT64_MIN); outsamplesref->pts = ROUNDED_DIV(outsamplesref->pts, inlink->sample_rate); #endif return ff_filter_frame(outlink, outsamplesref); } return ret; } static const AVClass *resample_child_class_next(const AVClass *prev) { return prev ? NULL : swr_get_class(); } static void *resample_child_next(void *obj, void *prev) { AResampleContext *s = obj; return prev ? NULL : s->swr; } #define OFFSET(x) offsetof(AResampleContext, x) #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM static const AVOption options[] = { {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS }, {NULL} }; static const AVClass aresample_class = { .class_name = "aresample", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, .child_class_next = resample_child_class_next, .child_next = resample_child_next, }; static const AVFilterPad aresample_inputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .filter_frame = filter_frame, }, { NULL } }; static const AVFilterPad aresample_outputs[] = { { .name = "default", .config_props = config_output, .request_frame = request_frame, .type = AVMEDIA_TYPE_AUDIO, }, { NULL } }; AVFilter avfilter_af_aresample = { .name = "aresample", .description = NULL_IF_CONFIG_SMALL("Resample audio data."), .init_dict = init_dict, .uninit = uninit, .query_formats = query_formats, .priv_size = sizeof(AResampleContext), .priv_class = &aresample_class, .inputs = aresample_inputs, .outputs = aresample_outputs, };