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-rw-r--r--doc/filters.texi66
-rw-r--r--libavfilter/Makefile3
-rw-r--r--libavfilter/af_volume.c (renamed from libavfilter/af_volume_justin.c)4
-rw-r--r--libavfilter/af_volume_stefano.c201
-rw-r--r--libavfilter/allfilters.c1
-rw-r--r--libavfilter/version.h4
6 files changed, 16 insertions, 263 deletions
diff --git a/doc/filters.texi b/doc/filters.texi
index 8e70cdb..56a213c 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -829,56 +829,6 @@ out
Convert the audio sample format, sample rate and channel layout. This filter is
not meant to be used directly.
-@section volume
-
-Adjust the input audio volume.
-
-The filter accepts exactly one parameter @var{vol}, which expresses
-how the audio volume will be increased or decreased.
-
-Output values are clipped to the maximum value.
-
-If @var{vol} is expressed as a decimal number, the output audio
-volume is given by the relation:
-@example
-@var{output_volume} = @var{vol} * @var{input_volume}
-@end example
-
-If @var{vol} is expressed as a decimal number followed by the string
-"dB", the value represents the requested change in decibels of the
-input audio power, and the output audio volume is given by the
-relation:
-@example
-@var{output_volume} = 10^(@var{vol}/20) * @var{input_volume}
-@end example
-
-Otherwise @var{vol} is considered an expression and its evaluated
-value is used for computing the output audio volume according to the
-first relation.
-
-Default value for @var{vol} is 1.0.
-
-@subsection Examples
-
-@itemize
-@item
-Half the input audio volume:
-@example
-volume=0.5
-@end example
-
-The above example is equivalent to:
-@example
-volume=1/2
-@end example
-
-@item
-Decrease input audio power by 12 decibels:
-@example
-volume=-12dB
-@end example
-@end itemize
-
@section volumedetect
Detect the volume of the input video.
@@ -919,7 +869,7 @@ There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.
In other words, raising the volume by +4 dB does not cause any clipping,
raising it by +5 dB causes clipping for 6 samples, etc.
-@section volume_justin
+@section volume
Adjust the input audio volume.
@@ -966,15 +916,21 @@ precision of the volume scaling.
@item
Halve the input audio volume:
@example
-volume_justin=volume=0.5
-volume_justin=volume=1/2
-volume_justin=volume=-6.0206dB
+volume=volume=0.5
+volume=volume=1/2
+volume=volume=-6.0206dB
+@end example
+
+In all the above example the named key for @option{volume} can be
+omitted, for example like in:
+@example
+volume=0.5
@end example
@item
Increase input audio power by 6 decibels using fixed-point precision:
@example
-volume_justin=volume=6dB:precision=fixed
+volume=volume=6dB:precision=fixed
@end example
@end itemize
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 7f9f0ef..377bd4d 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -71,8 +71,7 @@ OBJS-$(CONFIG_JOIN_FILTER) += af_join.o
OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o
-OBJS-$(CONFIG_VOLUME_FILTER) += af_volume_stefano.o
-OBJS-$(CONFIG_VOLUME_JUSTIN_FILTER) += af_volume_justin.o
+OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
OBJS-$(CONFIG_VOLUMEDETECT_FILTER) += af_volumedetect.o
OBJS-$(CONFIG_AEVALSRC_FILTER) += asrc_aevalsrc.o
diff --git a/libavfilter/af_volume_justin.c b/libavfilter/af_volume.c
index 0ba466a..5ffa1fe 100644
--- a/libavfilter/af_volume_justin.c
+++ b/libavfilter/af_volume.c
@@ -299,8 +299,8 @@ static const AVFilterPad avfilter_af_volume_outputs[] = {
{ NULL }
};
-AVFilter avfilter_af_volume_justin = {
- .name = "volume_justin",
+AVFilter avfilter_af_volume = {
+ .name = "volume",
.description = NULL_IF_CONFIG_SMALL("Change input volume."),
.query_formats = query_formats,
.priv_size = sizeof(VolumeContext),
diff --git a/libavfilter/af_volume_stefano.c b/libavfilter/af_volume_stefano.c
deleted file mode 100644
index 7608083..0000000
--- a/libavfilter/af_volume_stefano.c
+++ /dev/null
@@ -1,201 +0,0 @@
-/*
- * Copyright (c) 2011 Stefano Sabatini
- *
- * 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
- * audio volume filter
- * based on ffmpeg.c code
- */
-
-#include "libavutil/channel_layout.h"
-#include "libavutil/eval.h"
-#include "audio.h"
-#include "avfilter.h"
-#include "formats.h"
-
-typedef struct {
- double volume;
- int volume_i;
-} VolumeContext;
-
-static av_cold int init(AVFilterContext *ctx, const char *args)
-{
- VolumeContext *vol = ctx->priv;
- char *tail;
- int ret = 0;
-
- vol->volume = 1.0;
-
- if (args) {
- /* parse the number as a decimal number */
- double d = strtod(args, &tail);
-
- if (*tail) {
- if (!strcmp(tail, "dB")) {
- /* consider the argument an adjustement in decibels */
- d = pow(10, d/20);
- } else {
- /* parse the argument as an expression */
- ret = av_expr_parse_and_eval(&d, args, NULL, NULL,
- NULL, NULL, NULL, NULL,
- NULL, 0, ctx);
- }
- }
-
- if (ret < 0) {
- av_log(ctx, AV_LOG_ERROR,
- "Invalid volume argument '%s'\n", args);
- return AVERROR(EINVAL);
- }
-
- if (d < 0 || d > 65536) { /* 65536 = INT_MIN / (128 * 256) */
- av_log(ctx, AV_LOG_ERROR,
- "Negative or too big volume value %f\n", d);
- return AVERROR(EINVAL);
- }
-
- vol->volume = d;
- }
-
- vol->volume_i = (int)(vol->volume * 256 + 0.5);
- av_log(ctx, AV_LOG_VERBOSE, "volume=%f\n", vol->volume);
- return 0;
-}
-
-static int query_formats(AVFilterContext *ctx)
-{
- AVFilterFormats *formats = NULL;
- AVFilterChannelLayouts *layouts;
- enum AVSampleFormat sample_fmts[] = {
- AV_SAMPLE_FMT_U8,
- AV_SAMPLE_FMT_S16,
- AV_SAMPLE_FMT_S32,
- AV_SAMPLE_FMT_FLT,
- AV_SAMPLE_FMT_DBL,
- AV_SAMPLE_FMT_NONE
- };
-
- layouts = ff_all_channel_layouts();
- if (!layouts)
- return AVERROR(ENOMEM);
- ff_set_common_channel_layouts(ctx, layouts);
-
- formats = ff_make_format_list(sample_fmts);
- if (!formats)
- return AVERROR(ENOMEM);
- ff_set_common_formats(ctx, formats);
-
- formats = ff_all_samplerates();
- if (!formats)
- return AVERROR(ENOMEM);
- ff_set_common_samplerates(ctx, formats);
-
- return 0;
-}
-
-static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
-{
- VolumeContext *vol = inlink->dst->priv;
- AVFilterLink *outlink = inlink->dst->outputs[0];
- const int nb_samples = insamples->audio->nb_samples *
- av_get_channel_layout_nb_channels(insamples->audio->channel_layout);
- const double volume = vol->volume;
- const int volume_i = vol->volume_i;
- int i;
-
- if (volume_i != 256) {
- switch (insamples->format) {
- case AV_SAMPLE_FMT_U8:
- {
- uint8_t *p = (void *)insamples->data[0];
- for (i = 0; i < nb_samples; i++) {
- int v = (((*p - 128) * volume_i + 128) >> 8) + 128;
- *p++ = av_clip_uint8(v);
- }
- break;
- }
- case AV_SAMPLE_FMT_S16:
- {
- int16_t *p = (void *)insamples->data[0];
- for (i = 0; i < nb_samples; i++) {
- int v = ((int64_t)*p * volume_i + 128) >> 8;
- *p++ = av_clip_int16(v);
- }
- break;
- }
- case AV_SAMPLE_FMT_S32:
- {
- int32_t *p = (void *)insamples->data[0];
- for (i = 0; i < nb_samples; i++) {
- int64_t v = (((int64_t)*p * volume_i + 128) >> 8);
- *p++ = av_clipl_int32(v);
- }
- break;
- }
- case AV_SAMPLE_FMT_FLT:
- {
- float *p = (void *)insamples->data[0];
- float scale = (float)volume;
- for (i = 0; i < nb_samples; i++) {
- *p++ *= scale;
- }
- break;
- }
- case AV_SAMPLE_FMT_DBL:
- {
- double *p = (void *)insamples->data[0];
- for (i = 0; i < nb_samples; i++) {
- *p *= volume;
- p++;
- }
- break;
- }
- }
- }
- return ff_filter_frame(outlink, insamples);
-}
-
-static const AVFilterPad volume_inputs[] = {
- {
- .name = "default",
- .type = AVMEDIA_TYPE_AUDIO,
- .filter_frame = filter_frame,
- .min_perms = AV_PERM_READ | AV_PERM_WRITE,
- },
- { NULL },
-};
-
-static const AVFilterPad volume_outputs[] = {
- {
- .name = "default",
- .type = AVMEDIA_TYPE_AUDIO,
- },
- { NULL },
-};
-
-AVFilter avfilter_af_volume = {
- .name = "volume",
- .description = NULL_IF_CONFIG_SMALL("Change input volume."),
- .query_formats = query_formats,
- .priv_size = sizeof(VolumeContext),
- .init = init,
- .inputs = volume_inputs,
- .outputs = volume_outputs,
-};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index f9cacfc..ffde5ce 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -64,7 +64,6 @@ void avfilter_register_all(void)
REGISTER_FILTER (RESAMPLE, resample, af);
REGISTER_FILTER (SILENCEDETECT, silencedetect, af);
REGISTER_FILTER (VOLUME, volume, af);
- REGISTER_FILTER (VOLUME_JUSTIN, volume_justin, af);
REGISTER_FILTER (VOLUMEDETECT,volumedetect,af);
REGISTER_FILTER (AEVALSRC, aevalsrc, asrc);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 68a3543..ed17978 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -29,8 +29,8 @@
#include "libavutil/avutil.h"
#define LIBAVFILTER_VERSION_MAJOR 3
-#define LIBAVFILTER_VERSION_MINOR 25
-#define LIBAVFILTER_VERSION_MICRO 102
+#define LIBAVFILTER_VERSION_MINOR 26
+#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \
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