[FFmpeg-cvslog] lavfi: add volume filter
Justin Ruggles
git at videolan.org
Thu Dec 6 15:23:07 CET 2012
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Sat Sep 29 00:38:13 2012 -0400| [b384e031daeb1ac612620985e3e5377bc587559c] | committer: Justin Ruggles
lavfi: add volume filter
Based on the volume filter in FFmpeg written by Stefano Sabatini
<stefasab at gmail.com>.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b384e031daeb1ac612620985e3e5377bc587559c
---
Changelog | 1 +
doc/filters.texi | 53 ++++++++
libavfilter/Makefile | 1 +
libavfilter/af_volume.c | 314 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/af_volume.h | 53 ++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
7 files changed, 424 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 33e55a5..7216622 100644
--- a/Changelog
+++ b/Changelog
@@ -4,6 +4,7 @@ releases are sorted from youngest to oldest.
version <next>:
- ashowinfo audio filter
- 24-bit FLAC encoding
+- audio volume filter
version 9_beta2:
diff --git a/doc/filters.texi b/doc/filters.texi
index f092f3c..55e4468 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -359,6 +359,59 @@ 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.
+ at section volume
+
+Adjust the input audio volume.
+
+The filter accepts the following named parameters:
+ at table @option
+
+ at item volume
+Expresses how the audio volume will be increased or decreased.
+
+Output values are clipped to the maximum value.
+
+The output audio volume is given by the relation:
+ at example
+ at var{output_volume} = @var{volume} * @var{input_volume}
+ at end example
+
+Default value for @var{volume} is 1.0.
+
+ at item precision
+Mathematical precision.
+
+This determines which input sample formats will be allowed, which affects the
+precision of the volume scaling.
+
+ at table @option
+ at item fixed
+8-bit fixed-point; limits input sample format to U8, S16, and S32.
+ at item float
+32-bit floating-point; limits input sample format to FLT. (default)
+ at item double
+64-bit floating-point; limits input sample format to DBL.
+ at end table
+ at end table
+
+ at subsection Examples
+
+ at itemize
+ at item
+Halve the input audio volume:
+ at example
+volume=volume=0.5
+volume=volume=1/2
+volume=volume=-6.0206dB
+ at end example
+
+ at item
+Increase input audio power by 6 decibels using fixed-point precision:
+ at example
+volume=volume=6dB:precision=fixed
+ at end example
+ at end itemize
+
@c man end AUDIO FILTERS
@chapter Audio Sources
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 752ff40..2559e8a 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -35,6 +35,7 @@ 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..4a4e29f
--- /dev/null
+++ b/libavfilter/af_volume.c
@@ -0,0 +1,314 @@
+/*
+ * Copyright (c) 2011 Stefano Sabatini
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
+ *
+ * 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 "libavutil/float_dsp.h"
+#include "libavutil/opt.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "internal.h"
+#include "af_volume.h"
+
+static const char *precision_str[] = {
+ "fixed", "float", "double"
+};
+
+#define OFFSET(x) offsetof(VolumeContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM
+
+static const AVOption options[] = {
+ { "volume", "Volume adjustment.",
+ OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0, 0x7fffff, A },
+ { "precision", "Mathematical precision.",
+ OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A, "precision" },
+ { "fixed", "8-bit fixed-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED }, INT_MIN, INT_MAX, A, "precision" },
+ { "float", "32-bit floating-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT }, INT_MIN, INT_MAX, A, "precision" },
+ { "double", "64-bit floating-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A, "precision" },
+ { NULL },
+};
+
+static const AVClass volume_class = {
+ .class_name = "volume filter",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+static av_cold int init(AVFilterContext *ctx, const char *args)
+{
+ VolumeContext *vol = ctx->priv;
+ int ret;
+
+ vol->class = &volume_class;
+ av_opt_set_defaults(vol);
+
+ if ((ret = av_set_options_string(vol, args, "=", ":")) < 0) {
+ av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args);
+ return ret;
+ }
+
+ if (vol->precision == PRECISION_FIXED) {
+ vol->volume_i = (int)(vol->volume * 256 + 0.5);
+ vol->volume = vol->volume_i / 256.0;
+ av_log(ctx, AV_LOG_VERBOSE, "volume:(%d/256)(%f)(%1.2fdB) precision:fixed\n",
+ vol->volume_i, vol->volume, 20.0*log(vol->volume)/M_LN10);
+ } else {
+ av_log(ctx, AV_LOG_VERBOSE, "volume:(%f)(%1.2fdB) precision:%s\n",
+ vol->volume, 20.0*log(vol->volume)/M_LN10,
+ precision_str[vol->precision]);
+ }
+
+ av_opt_free(vol);
+ return ret;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ VolumeContext *vol = ctx->priv;
+ AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layouts;
+ static const enum AVSampleFormat sample_fmts[][7] = {
+ /* PRECISION_FIXED */
+ {
+ AV_SAMPLE_FMT_U8,
+ AV_SAMPLE_FMT_U8P,
+ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_S32,
+ AV_SAMPLE_FMT_S32P,
+ AV_SAMPLE_FMT_NONE
+ },
+ /* PRECISION_FLOAT */
+ {
+ AV_SAMPLE_FMT_FLT,
+ AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE
+ },
+ /* PRECISION_DOUBLE */
+ {
+ AV_SAMPLE_FMT_DBL,
+ AV_SAMPLE_FMT_DBLP,
+ 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[vol->precision]);
+ 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 inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
+ int nb_samples, int volume)
+{
+ int i;
+ for (i = 0; i < nb_samples; i++)
+ dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
+}
+
+static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
+ int nb_samples, int volume)
+{
+ int i;
+ for (i = 0; i < nb_samples; i++)
+ dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
+}
+
+static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,
+ int nb_samples, int volume)
+{
+ int i;
+ int16_t *smp_dst = (int16_t *)dst;
+ const int16_t *smp_src = (const int16_t *)src;
+ for (i = 0; i < nb_samples; i++)
+ smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8);
+}
+
+static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
+ int nb_samples, int volume)
+{
+ int i;
+ int16_t *smp_dst = (int16_t *)dst;
+ const int16_t *smp_src = (const int16_t *)src;
+ for (i = 0; i < nb_samples; i++)
+ smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
+}
+
+static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
+ int nb_samples, int volume)
+{
+ int i;
+ int32_t *smp_dst = (int32_t *)dst;
+ const int32_t *smp_src = (const int32_t *)src;
+ for (i = 0; i < nb_samples; i++)
+ smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
+}
+
+
+
+static void volume_init(VolumeContext *vol)
+{
+ vol->samples_align = 1;
+
+ switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
+ case AV_SAMPLE_FMT_U8:
+ if (vol->volume_i < 0x1000000)
+ vol->scale_samples = scale_samples_u8_small;
+ else
+ vol->scale_samples = scale_samples_u8;
+ break;
+ case AV_SAMPLE_FMT_S16:
+ if (vol->volume_i < 0x10000)
+ vol->scale_samples = scale_samples_s16_small;
+ else
+ vol->scale_samples = scale_samples_s16;
+ break;
+ case AV_SAMPLE_FMT_S32:
+ vol->scale_samples = scale_samples_s32;
+ break;
+ case AV_SAMPLE_FMT_FLT:
+ avpriv_float_dsp_init(&vol->fdsp, 0);
+ vol->samples_align = 4;
+ break;
+ case AV_SAMPLE_FMT_DBL:
+ avpriv_float_dsp_init(&vol->fdsp, 0);
+ vol->samples_align = 8;
+ break;
+ }
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ VolumeContext *vol = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
+
+ vol->sample_fmt = inlink->format;
+ vol->channels = av_get_channel_layout_nb_channels(inlink->channel_layout);
+ vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
+
+ volume_init(vol);
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+ VolumeContext *vol = inlink->dst->priv;
+ AVFilterLink *outlink = inlink->dst->outputs[0];
+ int nb_samples = buf->audio->nb_samples;
+ AVFilterBufferRef *out_buf;
+
+ if (vol->volume == 1.0 || vol->volume_i == 256)
+ return ff_filter_frame(outlink, buf);
+
+ /* do volume scaling in-place if input buffer is writable */
+ if (buf->perms & AV_PERM_WRITE) {
+ out_buf = buf;
+ } else {
+ out_buf = ff_get_audio_buffer(inlink, AV_PERM_WRITE, nb_samples);
+ if (!out_buf)
+ return AVERROR(ENOMEM);
+ out_buf->pts = buf->pts;
+ }
+
+ if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
+ int p, plane_samples;
+
+ if (av_sample_fmt_is_planar(buf->format))
+ plane_samples = FFALIGN(nb_samples, vol->samples_align);
+ else
+ plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
+
+ if (vol->precision == PRECISION_FIXED) {
+ for (p = 0; p < vol->planes; p++) {
+ vol->scale_samples(out_buf->extended_data[p],
+ buf->extended_data[p], plane_samples,
+ vol->volume_i);
+ }
+ } else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) {
+ for (p = 0; p < vol->planes; p++) {
+ vol->fdsp.vector_fmul_scalar((float *)out_buf->extended_data[p],
+ (const float *)buf->extended_data[p],
+ vol->volume, plane_samples);
+ }
+ } else {
+ for (p = 0; p < vol->planes; p++) {
+ vol->fdsp.vector_dmul_scalar((double *)out_buf->extended_data[p],
+ (const double *)buf->extended_data[p],
+ vol->volume, plane_samples);
+ }
+ }
+ }
+
+ if (buf != out_buf)
+ avfilter_unref_buffer(buf);
+
+ return ff_filter_frame(outlink, out_buf);
+}
+
+static const AVFilterPad avfilter_af_volume_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad avfilter_af_volume_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ },
+ { 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/af_volume.h b/libavfilter/af_volume.h
new file mode 100644
index 0000000..dec8767
--- /dev/null
+++ b/libavfilter/af_volume.h
@@ -0,0 +1,53 @@
+/*
+ * 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
+ */
+
+#ifndef AVFILTER_AF_VOLUME_H
+#define AVFILTER_AF_VOLUME_H
+
+#include "libavutil/common.h"
+#include "libavutil/float_dsp.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+
+enum PrecisionType {
+ PRECISION_FIXED = 0,
+ PRECISION_FLOAT,
+ PRECISION_DOUBLE,
+};
+
+typedef struct VolumeContext {
+ const AVClass *class;
+ AVFloatDSPContext fdsp;
+ enum PrecisionType precision;
+ double volume;
+ int volume_i;
+ int channels;
+ int planes;
+ enum AVSampleFormat sample_fmt;
+
+ void (*scale_samples)(uint8_t *dst, const uint8_t *src, int nb_samples,
+ int volume);
+ int samples_align;
+} VolumeContext;
+
+#endif /* AVFILTER_AF_VOLUME_H */
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index d7a7b07..0d7cbc2 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -46,6 +46,7 @@ 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 eb5326b..c09d44b 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 2
+#define LIBAVFILTER_VERSION_MINOR 3
#define LIBAVFILTER_VERSION_MICRO 0
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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