Signed-off-by: Paul B Mahol <onemda@gmail.com>
Paul B Mahol authored on 2017/11/18 18:28:27... | ... |
@@ -429,6 +429,16 @@ How much to use compressed signal in output. Default is 1. |
429 | 429 |
Range is between 0 and 1. |
430 | 430 |
@end table |
431 | 431 |
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+@section acontrast |
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+Simple audio dynamic range commpression/expansion filter. |
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+ |
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+The filter accepts the following options: |
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+ |
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+@table @option |
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+@item contrast |
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+Set contrast. Default is 33. Allowed range is between 0 and 100. |
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+@end table |
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+ |
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432 | 442 |
@section acopy |
433 | 443 |
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434 | 444 |
Copy the input audio source unchanged to the output. This is mainly useful for |
... | ... |
@@ -31,6 +31,7 @@ OBJS-$(CONFIG_QSVVPP) += qsvvpp.o |
31 | 31 |
# audio filters |
32 | 32 |
OBJS-$(CONFIG_ABENCH_FILTER) += f_bench.o |
33 | 33 |
OBJS-$(CONFIG_ACOMPRESSOR_FILTER) += af_sidechaincompress.o |
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+OBJS-$(CONFIG_ACONTRAST_FILTER) += af_acontrast.o |
|
34 | 35 |
OBJS-$(CONFIG_ACOPY_FILTER) += af_acopy.o |
35 | 36 |
OBJS-$(CONFIG_ACROSSFADE_FILTER) += af_afade.o |
36 | 37 |
OBJS-$(CONFIG_ACRUSHER_FILTER) += af_acrusher.o |
37 | 38 |
new file mode 100644 |
... | ... |
@@ -0,0 +1,219 @@ |
0 |
+/* |
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+ * Copyright (c) 2008 Rob Sykes |
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+ * Copyright (c) 2017 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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+ |
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+#include "libavutil/channel_layout.h" |
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+#include "libavutil/opt.h" |
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+#include "avfilter.h" |
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+#include "audio.h" |
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+#include "formats.h" |
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+ |
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+typedef struct AudioContrastContext { |
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+ const AVClass *class; |
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+ float contrast; |
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+ void (*filter)(void **dst, const void **src, |
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+ int nb_samples, int channels, float contrast); |
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+} AudioContrastContext; |
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+ |
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+#define OFFSET(x) offsetof(AudioContrastContext, x) |
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+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
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+ |
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+static const AVOption acontrast_options[] = { |
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+ { "contrast", "set contrast", OFFSET(contrast), AV_OPT_TYPE_FLOAT, {.dbl=33}, 0, 100, A }, |
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+ { NULL } |
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+}; |
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+ |
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+AVFILTER_DEFINE_CLASS(acontrast); |
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+ |
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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 sample_fmts[] = { |
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+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, |
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+ AV_SAMPLE_FMT_DBL, 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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+ |
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+ formats = ff_make_format_list(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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+ |
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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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+ |
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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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+ |
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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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+ |
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+static void filter_flt(void **d, const void **s, |
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+ int nb_samples, int channels, |
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+ float contrast) |
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+{ |
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+ const float *src = s[0]; |
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+ float *dst = d[0]; |
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+ int n, c; |
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+ |
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+ for (n = 0; n < nb_samples; n++) { |
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+ for (c = 0; c < channels; c++) { |
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+ float d = src[c] * M_PI_2; |
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+ |
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+ dst[c] = sinf(d + contrast * sinf(d * 4)); |
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+ } |
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+ |
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+ dst += c; |
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+ src += c; |
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+ } |
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+} |
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+ |
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+static void filter_dbl(void **d, const void **s, |
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+ int nb_samples, int channels, |
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+ float contrast) |
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+{ |
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+ const double *src = s[0]; |
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+ double *dst = d[0]; |
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+ int n, c; |
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+ |
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+ for (n = 0; n < nb_samples; n++) { |
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+ for (c = 0; c < channels; c++) { |
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+ double d = src[c] * M_PI_2; |
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+ |
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+ dst[c] = sin(d + contrast * sin(d * 4)); |
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+ } |
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+ |
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+ dst += c; |
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+ src += c; |
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+ } |
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+} |
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+ |
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+static void filter_fltp(void **d, const void **s, |
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+ int nb_samples, int channels, |
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+ float contrast) |
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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 float *src = s[c]; |
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+ float *dst = d[c]; |
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+ |
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+ for (n = 0; n < nb_samples; n++) { |
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+ float d = src[n] * M_PI_2; |
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+ |
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+ dst[n] = sinf(d + contrast * sinf(d * 4)); |
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+ } |
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+ } |
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+} |
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+ |
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+static void filter_dblp(void **d, const void **s, |
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+ int nb_samples, int channels, |
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+ float contrast) |
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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 double *src = s[c]; |
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+ double *dst = d[c]; |
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+ |
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+ for (n = 0; n < nb_samples; n++) { |
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+ double d = src[n] * M_PI_2; |
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+ |
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+ dst[n] = sin(d + contrast * sin(d * 4)); |
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+ } |
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+ } |
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+} |
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+ |
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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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+ AudioContrastContext *s = ctx->priv; |
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+ |
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+ switch (inlink->format) { |
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+ case AV_SAMPLE_FMT_FLT: s->filter = filter_flt; break; |
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+ case AV_SAMPLE_FMT_DBL: s->filter = filter_dbl; break; |
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+ case AV_SAMPLE_FMT_FLTP: s->filter = filter_fltp; break; |
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+ case AV_SAMPLE_FMT_DBLP: s->filter = filter_dblp; break; |
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+ } |
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+ |
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+ return 0; |
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+} |
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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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+ AVFilterLink *outlink = ctx->outputs[0]; |
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+ AudioContrastContext *s = ctx->priv; |
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+ AVFrame *out; |
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+ |
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+ if (av_frame_is_writable(in)) { |
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+ out = in; |
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+ } else { |
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+ out = ff_get_audio_buffer(inlink, 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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+ } |
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+ |
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+ s->filter((void **)out->extended_data, (const void **)in->extended_data, |
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+ in->nb_samples, in->channels, s->contrast / 750); |
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+ |
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+ if (out != in) |
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+ av_frame_free(&in); |
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+ |
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+ return ff_filter_frame(outlink, out); |
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+} |
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+ |
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+static const AVFilterPad 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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+ |
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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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+ }, |
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+ { NULL } |
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+}; |
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+ |
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+AVFilter ff_af_acontrast = { |
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+ .name = "acontrast", |
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+ .description = NULL_IF_CONFIG_SMALL("Simple audio dynamic range compression/expansion filter."), |
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+ .query_formats = query_formats, |
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+ .priv_size = sizeof(AudioContrastContext), |
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+ .priv_class = &acontrast_class, |
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+ .inputs = inputs, |
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+ .outputs = outputs, |
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+}; |
... | ... |
@@ -42,6 +42,7 @@ static void register_all(void) |
42 | 42 |
{ |
43 | 43 |
REGISTER_FILTER(ABENCH, abench, af); |
44 | 44 |
REGISTER_FILTER(ACOMPRESSOR, acompressor, af); |
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+ REGISTER_FILTER(ACONTRAST, acontrast, af); |
|
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REGISTER_FILTER(ACOPY, acopy, af); |
46 | 47 |
REGISTER_FILTER(ACROSSFADE, acrossfade, af); |
47 | 48 |
REGISTER_FILTER(ACRUSHER, acrusher, af); |
... | ... |
@@ -30,7 +30,7 @@ |
30 | 30 |
#include "libavutil/version.h" |
31 | 31 |
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#define LIBAVFILTER_VERSION_MAJOR 7 |
33 |
-#define LIBAVFILTER_VERSION_MINOR 1 |
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+#define LIBAVFILTER_VERSION_MINOR 2 |
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#define LIBAVFILTER_VERSION_MICRO 100 |
35 | 35 |
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#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \ |