[FFmpeg-cvslog] lavfi: add volume filter
Stefano Sabatini
git at videolan.org
Sat Nov 5 02:18:57 CET 2011
ffmpeg | branch: master | Stefano Sabatini <stefasab at gmail.com> | Tue Nov 1 21:42:14 2011 +0100| [618ac71354cf406a652109a90e6aa5e4e00d9463] | committer: Stefano Sabatini
lavfi: add volume filter
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=618ac71354cf406a652109a90e6aa5e4e00d9463
---
Changelog | 2 +
doc/filters.texi | 50 ++++++++++++
libavfilter/Makefile | 1 +
libavfilter/af_volume.c | 191 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/avfilter.h | 4 +-
6 files changed, 247 insertions(+), 2 deletions(-)
diff --git a/Changelog b/Changelog
index ceeead5..f35477e 100644
--- a/Changelog
+++ b/Changelog
@@ -73,6 +73,8 @@ easier to use. The changes are:
- Video Decoder Acceleration (VDA) HWAccel module.
- replacement Indeo 3 decoder
- new ffmpeg option: -map_channel
+- volume audio filter added
+
version 0.8:
diff --git a/doc/filters.texi b/doc/filters.texi
index 0da5702..d21ddf1 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -224,6 +224,56 @@ expressed in the form "[@var{c0} @var{c1} @var{c2} @var{c3} @var{c4} @var{c5}
@var{c6} @var{c7}]"
@end table
+ at 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:
+ at example
+ at var{output_volume} = @var{vol} * @var{input_volume}
+ at 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:
+ at example
+ at var{output_volume} = 10^(@var{vol}/20) * @var{input_volume}
+ at 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.
+
+ at subsection Examples
+
+ at itemize
+ at item
+Half the input audio volume:
+ at example
+volume=0.5
+ at end example
+
+The above example is equivalent to:
+ at example
+volume=1/2
+ at end example
+
+ at item
+Decrease input audio power by 12 decibels:
+ at example
+volume=-12dB
+ at end example
+ at end itemize
+
@c man end AUDIO FILTERS
@chapter Audio Sources
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index cfe5d74..edfb12f 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -28,6 +28,7 @@ OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
+OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
OBJS-$(CONFIG_ABUFFER_FILTER) += asrc_abuffer.o
OBJS-$(CONFIG_AEVALSRC_FILTER) += asrc_aevalsrc.o
diff --git a/libavfilter/af_volume.c b/libavfilter/af_volume.c
new file mode 100644
index 0000000..74e0bbb
--- /dev/null
+++ b/libavfilter/af_volume.c
@@ -0,0 +1,191 @@
+/*
+ * 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/audioconvert.h"
+#include "libavutil/eval.h"
+#include "avfilter.h"
+
+typedef struct {
+ double volume;
+ int volume_i;
+} VolumeContext;
+
+static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
+{
+ 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 */
+ if (!strcmp(tail, "dB")) {
+ d = exp10(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_INFO, "volume=%f\n", vol->volume);
+ return 0;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ 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
+ };
+ int packing_fmts[] = { AVFILTER_PACKED, -1 };
+
+ formats = avfilter_make_all_channel_layouts();
+ if (!formats)
+ return AVERROR(ENOMEM);
+ avfilter_set_common_channel_layouts(ctx, formats);
+
+ formats = avfilter_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ avfilter_set_common_sample_formats(ctx, formats);
+
+ formats = avfilter_make_format_list(packing_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ avfilter_set_common_packing_formats(ctx, formats);
+
+ return 0;
+}
+
+static void 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 = (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;
+ }
+ }
+ }
+ avfilter_filter_samples(outlink, insamples);
+}
+
+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 = (AVFilterPad[]) {{ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_samples = filter_samples,
+ .min_perms = AV_PERM_READ|AV_PERM_WRITE},
+ { .name = NULL}},
+
+ .outputs = (AVFilterPad[]) {{ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO, },
+ { .name = NULL}},
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 3c77adb..e80fc17 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -39,6 +39,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (ANULL, anull, af);
REGISTER_FILTER (ARESAMPLE, aresample, af);
REGISTER_FILTER (ASHOWINFO, ashowinfo, af);
+ REGISTER_FILTER (VOLUME, volume, af);
REGISTER_FILTER (ABUFFER, abuffer, asrc);
REGISTER_FILTER (AEVALSRC, aevalsrc, asrc);
diff --git a/libavfilter/avfilter.h b/libavfilter/avfilter.h
index 9c67b35..4021522 100644
--- a/libavfilter/avfilter.h
+++ b/libavfilter/avfilter.h
@@ -29,8 +29,8 @@
#include "libavutil/rational.h"
#define LIBAVFILTER_VERSION_MAJOR 2
-#define LIBAVFILTER_VERSION_MINOR 45
-#define LIBAVFILTER_VERSION_MICRO 3
+#define LIBAVFILTER_VERSION_MINOR 46
+#define LIBAVFILTER_VERSION_MICRO 0
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \
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