mirror of https://github.com/FFmpeg/FFmpeg.git
Signed-off-by: Paul B Mahol <onemda@gmail.com>pull/297/head
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6 changed files with 219 additions and 1 deletions
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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 "audio.h" |
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#include "avfilter.h" |
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#include "internal.h" |
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typedef struct ADerivativeContext { |
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const AVClass *class; |
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AVFrame *prev; |
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void (*filter)(void **dst, void **prv, const void **src, |
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int nb_samples, int channels); |
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} ADerivativeContext; |
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static int query_formats(AVFilterContext *ctx) |
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{ |
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AVFilterFormats *formats = NULL; |
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AVFilterChannelLayouts *layouts = NULL; |
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static const enum AVSampleFormat derivative_sample_fmts[] = { |
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AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLTP, |
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AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_DBLP, |
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AV_SAMPLE_FMT_NONE |
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}; |
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static const enum AVSampleFormat integral_sample_fmts[] = { |
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, |
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AV_SAMPLE_FMT_NONE |
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}; |
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int ret; |
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formats = ff_make_format_list(strcmp(ctx->filter->name, "aintegral") ? |
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derivative_sample_fmts : integral_sample_fmts); |
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if (!formats) |
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return AVERROR(ENOMEM); |
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ret = ff_set_common_formats(ctx, formats); |
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if (ret < 0) |
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return ret; |
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layouts = ff_all_channel_counts(); |
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if (!layouts) |
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return AVERROR(ENOMEM); |
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ret = ff_set_common_channel_layouts(ctx, layouts); |
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if (ret < 0) |
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return ret; |
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formats = ff_all_samplerates(); |
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return ff_set_common_samplerates(ctx, formats); |
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} |
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#define DERIVATIVE(name, type) \ |
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static void aderivative_## name ##p(void **d, void **p, const void **s, \
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int nb_samples, int channels) \
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{ \
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int n, c; \
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\
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for (c = 0; c < channels; c++) { \
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const type *src = s[c]; \
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type *dst = d[c]; \
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type *prv = p[c]; \
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\
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for (n = 0; n < nb_samples; n++) { \
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const type current = src[n]; \
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\
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dst[n] = current - prv[0]; \
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prv[0] = current; \
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} \
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} \
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} |
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DERIVATIVE(flt, float) |
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DERIVATIVE(dbl, double) |
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DERIVATIVE(s16, int16_t) |
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DERIVATIVE(s32, int32_t) |
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#define INTEGRAL(name, type) \ |
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static void aintegral_## name ##p(void **d, void **p, const void **s, \
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int nb_samples, int channels) \
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{ \
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int n, c; \
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\
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for (c = 0; c < channels; c++) { \
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const type *src = s[c]; \
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type *dst = d[c]; \
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type *prv = p[c]; \
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\
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for (n = 0; n < nb_samples; n++) { \
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const type current = src[n]; \
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\
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dst[n] = current + prv[0]; \
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prv[0] = dst[n]; \
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} \
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} \
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} |
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INTEGRAL(flt, float) |
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INTEGRAL(dbl, double) |
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static int config_input(AVFilterLink *inlink) |
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{ |
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AVFilterContext *ctx = inlink->dst; |
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ADerivativeContext *s = ctx->priv; |
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switch (inlink->format) { |
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case AV_SAMPLE_FMT_FLTP: s->filter = aderivative_fltp; break; |
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case AV_SAMPLE_FMT_DBLP: s->filter = aderivative_dblp; break; |
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case AV_SAMPLE_FMT_S32P: s->filter = aderivative_s32p; break; |
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case AV_SAMPLE_FMT_S16P: s->filter = aderivative_s16p; break; |
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} |
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if (strcmp(ctx->filter->name, "aintegral")) |
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return 0; |
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switch (inlink->format) { |
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case AV_SAMPLE_FMT_FLTP: s->filter = aintegral_fltp; break; |
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case AV_SAMPLE_FMT_DBLP: s->filter = aintegral_dblp; break; |
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} |
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return 0; |
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} |
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static int filter_frame(AVFilterLink *inlink, AVFrame *in) |
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{ |
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AVFilterContext *ctx = inlink->dst; |
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ADerivativeContext *s = ctx->priv; |
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AVFilterLink *outlink = ctx->outputs[0]; |
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AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples); |
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if (!out) { |
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av_frame_free(&in); |
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return AVERROR(ENOMEM); |
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} |
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av_frame_copy_props(out, in); |
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if (!s->prev) { |
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s->prev = ff_get_audio_buffer(inlink, 1); |
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if (!s->prev) { |
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av_frame_free(&in); |
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return AVERROR(ENOMEM); |
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} |
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} |
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s->filter((void **)out->extended_data, (void **)s->prev->extended_data, (const void **)in->extended_data, |
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in->nb_samples, in->channels); |
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av_frame_free(&in); |
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return ff_filter_frame(outlink, out); |
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} |
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static av_cold void uninit(AVFilterContext *ctx) |
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{ |
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ADerivativeContext *s = ctx->priv; |
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av_frame_free(&s->prev); |
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} |
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static const AVFilterPad aderivative_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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.config_props = config_input, |
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}, |
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{ NULL } |
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}; |
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static const AVFilterPad aderivative_outputs[] = { |
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{ |
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.name = "default", |
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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_aderivative = { |
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.name = "aderivative", |
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.description = NULL_IF_CONFIG_SMALL("Compute derivative of input audio."), |
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.query_formats = query_formats, |
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.priv_size = sizeof(ADerivativeContext), |
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.uninit = uninit, |
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.inputs = aderivative_inputs, |
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.outputs = aderivative_outputs, |
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}; |
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AVFilter ff_af_aintegral = { |
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.name = "aintegral", |
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.description = NULL_IF_CONFIG_SMALL("Compute integral of input audio."), |
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.query_formats = query_formats, |
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.priv_size = sizeof(ADerivativeContext), |
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.uninit = uninit, |
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.inputs = aderivative_inputs, |
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.outputs = aderivative_outputs, |
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}; |
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