lavfi: add volume filter
authorStefano Sabatini <stefasab@gmail.com>
Tue, 1 Nov 2011 20:42:14 +0000 (21:42 +0100)
committerStefano Sabatini <stefasab@gmail.com>
Sat, 5 Nov 2011 01:13:37 +0000 (02:13 +0100)
Changelog
doc/filters.texi
libavfilter/Makefile
libavfilter/af_volume.c [new file with mode: 0644]
libavfilter/allfilters.c
libavfilter/avfilter.h

index ceeead5..f35477e 100644 (file)
--- 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:
 
index 0da5702..d21ddf1 100644 (file)
@@ -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
 
+@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
+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
+
 @c man end AUDIO FILTERS
 
 @chapter Audio Sources
index cfe5d74..edfb12f 100644 (file)
@@ -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 (file)
index 0000000..74e0bbb
--- /dev/null
@@ -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}},
+};
index 3c77adb..e80fc17 100644 (file)
@@ -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);
index 9c67b35..4021522 100644 (file)
@@ -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, \