@@ -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:

@@ -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.

+@section volume

+

+Adjust the input audio volume.

+

+The filter accepts the following named parameters:

+@table @option

+

+@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:

+@example

+@var{output_volume} = @var{volume} * @var{input_volume}

+@end example

+

+Default value for @var{volume} is 1.0.

+

+@item precision

+Mathematical precision.

+

+This determines which input sample formats will be allowed, which affects the

+precision of the volume scaling.

+

+@table @option

+@item fixed

+8-bit fixed-point; limits input sample format to U8, S16, and S32.

+@item float

+32-bit floating-point; limits input sample format to FLT. (default)

+@item double

+64-bit floating-point; limits input sample format to DBL.

+@end table

+@end table

+

+@subsection Examples

+

+@itemize

+@item

+Halve the input audio volume:

+@example

+volume=volume=0.5

+volume=volume=1/2

+volume=volume=-6.0206dB

+@end example

+

+@item

+Increase input audio power by 6 decibels using fixed-point precision:

+@example

+volume=volume=6dB:precision=fixed

+@end example

+@end itemize

+

 @c man end AUDIO FILTERS

 @chapter Audio Sources

@@ -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

new file mode 100644

@@ -0,0 +1,314 @@

+/*

+ * Copyright (c) 2011 Stefano Sabatini

+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@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,

+};

new file mode 100644

@@ -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 */

@@ -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);

@@ -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, \