mirror of https://github.com/FFmpeg/FFmpeg.git
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/*
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* Copyright (c) 2021 Paul B Mahol |
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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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#include "libavutil/channel_layout.h" |
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#include "libavutil/common.h" |
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#include "libavutil/opt.h" |
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#include "audio.h" |
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#include "avfilter.h" |
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#include "formats.h" |
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#include "filters.h" |
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#include "internal.h" |
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typedef struct AudioSDRContext { |
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int channels; |
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int64_t pts; |
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double *sum_u; |
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double *sum_uv; |
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AVFrame *cache[2]; |
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} AudioSDRContext; |
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static void sdr(AVFilterContext *ctx, const AVFrame *u, const AVFrame *v) |
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{ |
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AudioSDRContext *s = ctx->priv; |
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for (int ch = 0; ch < u->channels; ch++) { |
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const double *const us = (double *)u->extended_data[ch]; |
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const double *const vs = (double *)v->extended_data[ch]; |
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double sum_uv = s->sum_uv[ch]; |
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double sum_u = s->sum_u[ch]; |
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for (int n = 0; n < u->nb_samples; n++) { |
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sum_u += us[n] * us[n]; |
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sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]); |
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} |
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s->sum_uv[ch] = sum_uv; |
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s->sum_u[ch] = sum_u; |
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} |
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} |
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static int activate(AVFilterContext *ctx) |
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{ |
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AudioSDRContext *s = ctx->priv; |
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int ret, status; |
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int available; |
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int64_t pts; |
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FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx); |
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available = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]), ff_inlink_queued_samples(ctx->inputs[1])); |
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if (available > 0) { |
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AVFrame *out; |
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for (int i = 0; i < 2; i++) { |
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ret = ff_inlink_consume_samples(ctx->inputs[i], available, available, &s->cache[i]); |
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if (ret > 0) { |
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if (s->pts == AV_NOPTS_VALUE) |
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s->pts = s->cache[i]->pts; |
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} |
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} |
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sdr(ctx, s->cache[0], s->cache[1]); |
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av_frame_free(&s->cache[1]); |
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out = s->cache[0]; |
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out->nb_samples = available; |
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out->pts = s->pts; |
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s->pts += available; |
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s->cache[0] = NULL; |
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return ff_filter_frame(ctx->outputs[0], out); |
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} |
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for (int i = 0; i < 2; i++) { |
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if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) { |
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ff_outlink_set_status(ctx->outputs[0], status, s->pts); |
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return 0; |
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} |
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} |
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if (ff_outlink_frame_wanted(ctx->outputs[0])) { |
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for (int i = 0; i < 2; i++) { |
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if (ff_inlink_queued_samples(ctx->inputs[i]) > 0) |
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continue; |
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ff_inlink_request_frame(ctx->inputs[i]); |
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} |
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return 0; |
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} |
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return FFERROR_NOT_READY; |
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} |
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static int config_output(AVFilterLink *outlink) |
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{ |
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AVFilterContext *ctx = outlink->src; |
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AVFilterLink *inlink = ctx->inputs[0]; |
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AudioSDRContext *s = ctx->priv; |
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s->pts = AV_NOPTS_VALUE; |
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s->channels = inlink->channels; |
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s->sum_u = av_calloc(outlink->channels, sizeof(*s->sum_u)); |
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s->sum_uv = av_calloc(outlink->channels, sizeof(*s->sum_uv)); |
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if (!s->sum_u || !s->sum_uv) |
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return AVERROR(ENOMEM); |
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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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AudioSDRContext *s = ctx->priv; |
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for (int ch = 0; ch < s->channels; ch++) |
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av_log(ctx, AV_LOG_INFO, "SDR ch%d: %g dB\n", ch, 20. * log10(s->sum_u[ch] / s->sum_uv[ch])); |
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av_frame_free(&s->cache[0]); |
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av_frame_free(&s->cache[1]); |
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av_freep(&s->sum_u); |
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av_freep(&s->sum_uv); |
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} |
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static const AVFilterPad inputs[] = { |
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{ |
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.name = "input0", |
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.type = AVMEDIA_TYPE_AUDIO, |
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}, |
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{ |
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.name = "input1", |
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.type = AVMEDIA_TYPE_AUDIO, |
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}, |
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}; |
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static const AVFilterPad outputs[] = { |
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{ |
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.name = "default", |
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.type = AVMEDIA_TYPE_AUDIO, |
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.config_props = config_output, |
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}, |
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}; |
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const AVFilter ff_af_asdr = { |
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.name = "asdr", |
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.description = NULL_IF_CONFIG_SMALL("Measure Audio Signal-to-Distortion Ratio."), |
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.priv_size = sizeof(AudioSDRContext), |
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.activate = activate, |
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.uninit = uninit, |
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FILTER_INPUTS(inputs), |
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FILTER_OUTPUTS(outputs), |
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FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP), |
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}; |
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