mirror of
https://git.videolan.org/git/ffmpeg.git
synced 2024-08-25 19:01:51 +02:00
352 lines
11 KiB
C
352 lines
11 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 "libavutil/avstring.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "libavutil/avassert.h"
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#include "libswresample/swresample.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "internal.h"
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typedef struct {
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const AVClass *class;
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int sample_rate_arg;
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double ratio;
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struct SwrContext *swr;
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int64_t next_pts;
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int more_data;
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} AResampleContext;
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static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)
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{
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AResampleContext *aresample = ctx->priv;
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int ret = 0;
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aresample->next_pts = AV_NOPTS_VALUE;
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aresample->swr = swr_alloc();
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if (!aresample->swr) {
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ret = AVERROR(ENOMEM);
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goto end;
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}
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if (opts) {
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AVDictionaryEntry *e = NULL;
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while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
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if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)
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goto end;
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}
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av_dict_free(opts);
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}
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if (aresample->sample_rate_arg > 0)
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av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
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end:
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return ret;
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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 query_formats(AVFilterContext *ctx)
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{
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AResampleContext *aresample = ctx->priv;
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enum AVSampleFormat out_format;
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int64_t out_rate, out_layout;
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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, *out_formats;
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AVFilterFormats *in_samplerates, *out_samplerates;
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AVFilterChannelLayouts *in_layouts, *out_layouts;
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int ret;
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av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
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av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
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av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);
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in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
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if ((ret = ff_formats_ref(in_formats, &inlink->out_formats)) < 0)
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return ret;
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in_samplerates = ff_all_samplerates();
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if ((ret = ff_formats_ref(in_samplerates, &inlink->out_samplerates)) < 0)
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return ret;
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in_layouts = ff_all_channel_counts();
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if ((ret = ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts)) < 0)
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return ret;
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if(out_rate > 0) {
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int ratelist[] = { out_rate, -1 };
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out_samplerates = ff_make_format_list(ratelist);
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} else {
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out_samplerates = ff_all_samplerates();
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}
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if ((ret = ff_formats_ref(out_samplerates, &outlink->in_samplerates)) < 0)
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return ret;
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if(out_format != AV_SAMPLE_FMT_NONE) {
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int formatlist[] = { out_format, -1 };
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out_formats = ff_make_format_list(formatlist);
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} else
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out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
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if ((ret = ff_formats_ref(out_formats, &outlink->in_formats)) < 0)
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return ret;
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if(out_layout) {
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int64_t layout_list[] = { out_layout, -1 };
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out_layouts = avfilter_make_format64_list(layout_list);
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} else
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out_layouts = ff_all_channel_counts();
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return ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
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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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int64_t out_rate, out_layout;
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enum AVSampleFormat out_format;
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char inchl_buf[128], outchl_buf[128];
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aresample->swr = swr_alloc_set_opts(aresample->swr,
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outlink->channel_layout, outlink->format, outlink->sample_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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if (!inlink->channel_layout)
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av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
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if (!outlink->channel_layout)
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av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
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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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av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
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av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);
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av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
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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 || !outlink->channel_layout);
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av_assert0(outlink->format == out_format);
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aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
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av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), inlink ->channels, inlink ->channel_layout);
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av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
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av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
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inlink ->channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
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outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
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{
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AResampleContext *aresample = inlink->dst->priv;
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const int n_in = insamplesref->nb_samples;
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int64_t delay;
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int n_out = n_in * aresample->ratio + 32;
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AVFilterLink *const outlink = inlink->dst->outputs[0];
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AVFrame *outsamplesref;
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int ret;
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delay = swr_get_delay(aresample->swr, outlink->sample_rate);
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if (delay > 0)
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n_out += FFMIN(delay, FFMAX(4096, n_out));
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outsamplesref = ff_get_audio_buffer(outlink, n_out);
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if(!outsamplesref)
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return AVERROR(ENOMEM);
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av_frame_copy_props(outsamplesref, insamplesref);
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outsamplesref->format = outlink->format;
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outsamplesref->channels = outlink->channels;
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outsamplesref->channel_layout = outlink->channel_layout;
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outsamplesref->sample_rate = outlink->sample_rate;
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if(insamplesref->pts != AV_NOPTS_VALUE) {
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int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
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int64_t outpts= swr_next_pts(aresample->swr, inpts);
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aresample->next_pts =
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outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
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} else {
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outsamplesref->pts = AV_NOPTS_VALUE;
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}
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n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
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(void *)insamplesref->extended_data, n_in);
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if (n_out <= 0) {
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av_frame_free(&outsamplesref);
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av_frame_free(&insamplesref);
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return 0;
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}
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aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers
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outsamplesref->nb_samples = n_out;
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ret = ff_filter_frame(outlink, outsamplesref);
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av_frame_free(&insamplesref);
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return ret;
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}
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static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret)
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{
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AVFilterContext *ctx = outlink->src;
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AResampleContext *aresample = ctx->priv;
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AVFilterLink *const inlink = outlink->src->inputs[0];
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AVFrame *outsamplesref;
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int n_out = 4096;
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int64_t pts;
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outsamplesref = ff_get_audio_buffer(outlink, n_out);
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*outsamplesref_ret = outsamplesref;
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if (!outsamplesref)
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return AVERROR(ENOMEM);
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pts = swr_next_pts(aresample->swr, INT64_MIN);
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pts = ROUNDED_DIV(pts, inlink->sample_rate);
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n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0);
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if (n_out <= 0) {
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av_frame_free(&outsamplesref);
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return (n_out == 0) ? AVERROR_EOF : n_out;
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}
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outsamplesref->sample_rate = outlink->sample_rate;
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outsamplesref->nb_samples = n_out;
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outsamplesref->pts = pts;
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return 0;
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}
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static int request_frame(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AResampleContext *aresample = ctx->priv;
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int ret;
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// First try to get data from the internal buffers
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if (aresample->more_data) {
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AVFrame *outsamplesref;
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if (flush_frame(outlink, 0, &outsamplesref) >= 0) {
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return ff_filter_frame(outlink, outsamplesref);
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}
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}
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aresample->more_data = 0;
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// Second request more data from the input
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ret = ff_request_frame(ctx->inputs[0]);
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// Third if we hit the end flush
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if (ret == AVERROR_EOF) {
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AVFrame *outsamplesref;
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if ((ret = flush_frame(outlink, 1, &outsamplesref)) < 0)
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return ret;
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return ff_filter_frame(outlink, outsamplesref);
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}
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return ret;
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}
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static const AVClass *resample_child_class_next(const AVClass *prev)
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{
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return prev ? NULL : swr_get_class();
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}
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static void *resample_child_next(void *obj, void *prev)
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{
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AResampleContext *s = obj;
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return prev ? NULL : s->swr;
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}
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#define OFFSET(x) offsetof(AResampleContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption options[] = {
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{"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
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{NULL}
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};
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static const AVClass aresample_class = {
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.class_name = "aresample",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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.child_class_next = resample_child_class_next,
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.child_next = resample_child_next,
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};
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static const AVFilterPad aresample_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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},
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{ NULL }
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};
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static const AVFilterPad aresample_outputs[] = {
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{
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.name = "default",
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.config_props = config_output,
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.request_frame = request_frame,
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL }
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};
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AVFilter ff_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_dict = init_dict,
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.uninit = uninit,
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.query_formats = query_formats,
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.priv_size = sizeof(AResampleContext),
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.priv_class = &aresample_class,
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.inputs = aresample_inputs,
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.outputs = aresample_outputs,
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};
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