From: Stefano Sabatini <[email protected]>
---
Made changes recommended by Diego and squashed with patch 2.
Changelog | 1 +
doc/filters.texi | 50 ++++++++++++
libavfilter/Makefile | 1 +
libavfilter/af_volume.c | 201 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 255 insertions(+), 1 deletions(-)
create mode 100644 libavfilter/af_volume.c
diff --git a/Changelog b/Changelog
index ad3e211..e0ccc8f 100644
--- a/Changelog
+++ b/Changelog
@@ -48,6 +48,7 @@ version <next>:
- Microsoft Screen 2 decoder
- RTP depacketization of JPEG
- Smooth Streaming live segmenter muxer
+- audio volume filter
version 0.8:
diff --git a/doc/filters.texi b/doc/filters.texi
index 4825b0d..8bb2040 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -318,6 +318,56 @@ not meant to be used directly, it is inserted
automatically by libavfilter
whenever conversion is needed. Use the @var{aformat} filter to force a specific
conversion.
+@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 decresed.
+
+Output values are clipped to the maximum value.
+
+If @var{vol} is expressed as a decimal number, and 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
+Halve 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
+
@c man end AUDIO FILTERS
@chapter Audio Sources
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 530aa57..876025f 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -34,6 +34,7 @@ OBJS-$(CONFIG_CHANNELMAP_FILTER) +=
af_channelmap.o
OBJS-$(CONFIG_CHANNELSPLIT_FILTER) += af_channelsplit.o
OBJS-$(CONFIG_JOIN_FILTER) += af_join.o
OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
+OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o
diff --git a/libavfilter/af_volume.c b/libavfilter/af_volume.c
new file mode 100644
index 0000000..aa42e13
--- /dev/null
+++ b/libavfilter/af_volume.c
@@ -0,0 +1,201 @@
+/*
+ * Copyright (c) 2011 Stefano Sabatini
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * audio volume filter
+ */
+
+#include "libavutil/audioconvert.h"
+#include "libavutil/common.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_samples(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 = 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 = (int16_t *)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 = (int32_t *)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 = (float *)insamples->data[0];
+ float scale = (float)volume;
+ for (i = 0; i < nb_samples; i++) {
+ *p++ *= scale;
+ }
+ break;
+ }
+ case AV_SAMPLE_FMT_DBL:
+ {
+ double *p = (double *)insamples->data[0];
+ for (i = 0; i < nb_samples; i++) {
+ *p *= volume;
+ p++;
+ }
+ break;
+ }
+ }
+ }
+ return ff_filter_samples(outlink, insamples);
+}
+
+static const AVFilterPad avfilter_af_volume_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_samples = filter_samples,
+ .min_perms = AV_PERM_READ | AV_PERM_WRITE,
+ },
+ { NULL }
+};
+
+static const AVFilterPad avfilter_af_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 = avfilter_af_volume_inputs,
+ .outputs = avfilter_af_volume_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 94b3115..29ded71 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -45,6 +45,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (CHANNELSPLIT,channelsplit,af);
REGISTER_FILTER (JOIN, join, af);
REGISTER_FILTER (RESAMPLE, resample, af);
+ REGISTER_FILTER (VOLUME, volume, af);
REGISTER_FILTER (ANULLSRC, anullsrc, asrc);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 0e72a47..eb5326b 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -29,7 +29,7 @@
#include "libavutil/avutil.h"
#define LIBAVFILTER_VERSION_MAJOR 3
-#define LIBAVFILTER_VERSION_MINOR 1
+#define LIBAVFILTER_VERSION_MINOR 2
#define LIBAVFILTER_VERSION_MICRO 0
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
--
1.7.1
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