[FFmpeg-devel] [PATCH] lavfi: add avolume filter
Stefano Sabatini
stefasab at gmail.com
Tue Nov 1 21:45:39 CET 2011
---
doc/filters.texi | 42 ++++++++++
libavfilter/Makefile | 1 +
libavfilter/af_avolume.c | 189 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 233 insertions(+), 0 deletions(-)
create mode 100644 libavfilter/af_avolume.c
diff --git a/doc/filters.texi b/doc/filters.texi
index 7085ae1..9b6f614 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -224,6 +224,48 @@ expressed in the form "[@var{c0} @var{c1} @var{c2} @var{c3} @var{c4} @var{c5}
@var{c6} @var{c7}]"
@end table
+ at section avolume
+
+Adjust the input audio volume. The filter accepts exactly one
+parameter, which express how the audio volume will be increased or
+decresed.
+
+Output values are clipped to the maximum value.
+
+If the parameter value is an integer, baseline value is 256, and the
+output audio volume is given by the relation:
+ at example
+ at var{output_volume}=@var{vol}/256 * @var{input_volume}
+ at end example
+
+If the parameter is expressed as a decimal number, baseline value is
+1.0, and the output audio volume is given by the relation:
+ at example
+ at var{output_volume}=@var{vol} * @var{input_volume}
+ at end example
+
+Note that when choosing the decimal number representation, "." must be
+specified in the parameter or it will be interpreted as an integer
+value.
+
+The expressed value must to be a non-negative value either way.
+Default parameter value is 1.0.
+
+ at subsection Examples
+
+ at itemize
+ at item
+Half the input audio volume:
+ at example
+avolume=0.5
+ at end example
+
+The above example is equivalent to:
+ at example
+avolume=128
+ at end example
+ at end itemize
+
@section earwax
Make audio easier to listen to on headphones. Adds `cues' to 44.1kHz
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 583dd6d..2eaa74e 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -27,6 +27,7 @@ OBJS-$(CONFIG_ACONVERT_FILTER) += af_aconvert.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
+OBJS-$(CONFIG_AVOLUME_FILTER) += af_avolume.o
OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
OBJS-$(CONFIG_LADSPA_FILTER) += af_ladspa.o
diff --git a/libavfilter/af_avolume.c b/libavfilter/af_avolume.c
new file mode 100644
index 0000000..62f2c97
--- /dev/null
+++ b/libavfilter/af_avolume.c
@@ -0,0 +1,189 @@
+/*
+ * 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 "avfilter.h"
+
+typedef struct {
+ /* todo: express it as a float (0-1) or int (0-256) */
+ int volume_i;
+ double volume_d;
+} VolumeContext;
+
+static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
+{
+ VolumeContext *vol = ctx->priv;
+ char *tail;
+ vol->volume_i = 256;
+ vol->volume_d = 1.0;
+
+ if (args) {
+ /* parse the number as an integer */
+ long int li = strtol(args, &tail, 10);
+
+ if (!*tail) {
+ if (li < 0 || li >= INT_MAX) {
+ av_log(ctx, AV_LOG_ERROR,
+ "Invalid or too big integer %ld, must be >= 0\n", li);
+ return AVERROR(EINVAL);
+ }
+ vol->volume_i = li;
+ vol->volume_d = (double)vol->volume_i / 256;
+ } else {
+ /* parse the number as a decimal number */
+ double d = strtod(args, &tail);
+ int err = errno;
+
+ if (*tail) {
+ av_log(ctx, AV_LOG_ERROR,
+ "String provided '%s' cannot be parsed as a number\n",
+ args);
+ return AVERROR(EINVAL);
+ }
+ if (d < 0 || err) {
+ av_log(ctx, AV_LOG_ERROR,
+ "Invalid or too big decimal number %f, must be >= 0.0\n", d);
+ return AVERROR(EINVAL);
+ }
+ vol->volume_d = d;
+ vol->volume_i = (int)(vol->volume_d * 256 + 0.5);
+ }
+ }
+
+ av_log(ctx, AV_LOG_INFO, "volume_i:%d volume_d=%f\n", vol->volume_i, vol->volume_d);
+ 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];
+ int nb_samples = insamples->audio->nb_samples *
+ av_get_channel_layout_nb_channels(insamples->audio->channel_layout);
+ const int volume_i = vol->volume_i;
+ const double volume_d = vol->volume_d;
+ 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 = ((*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_d;
+ 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_d;
+ p++;
+ }
+ break;
+ }
+ }
+ }
+ avfilter_filter_samples(outlink, insamples);
+}
+
+AVFilter avfilter_af_avolume = {
+ .name = "avolume",
+ .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 185114c..37ca7f0 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 (AVOLUME, avolume, af);
REGISTER_FILTER (EARWAX, earwax, af);
REGISTER_FILTER (LADSPA, ladspa, af);
REGISTER_FILTER (SOX, sox, af);
--
1.7.4.1
More information about the ffmpeg-devel
mailing list