@@ -829,56 +829,6 @@ out

 Convert the audio sample format, sample rate and channel layout. This filter is

 not meant to be used directly.

-@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 decreased.

-

-Output values are clipped to the maximum value.

-

-If @var{vol} is expressed as a decimal number, 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

-

 @section volumedetect

 Detect the volume of the input video.

@@ -919,7 +869,7 @@ There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.

 In other words, raising the volume by +4 dB does not cause any clipping,

 raising it by +5 dB causes clipping for 6 samples, etc.

-@section volume_justin

+@section volume

 Adjust the input audio volume.

@@ -966,15 +916,21 @@ precision of the volume scaling.

 @item

 Halve the input audio volume:

 @example

-volume_justin=volume=0.5

-volume_justin=volume=1/2

-volume_justin=volume=-6.0206dB

+volume=volume=0.5

+volume=volume=1/2

+volume=volume=-6.0206dB

+@end example

+

+In all the above example the named key for @option{volume} can be

+omitted, for example like in:

+@example

+volume=0.5

 @end example

 @item

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

 @example

-volume_justin=volume=6dB:precision=fixed

+volume=volume=6dB:precision=fixed

 @end example

 @end itemize

@@ -71,8 +71,7 @@ OBJS-$(CONFIG_JOIN_FILTER)                   += af_join.o

 OBJS-$(CONFIG_PAN_FILTER)                    += af_pan.o

 OBJS-$(CONFIG_RESAMPLE_FILTER)               += af_resample.o

 OBJS-$(CONFIG_SILENCEDETECT_FILTER)          += af_silencedetect.o

-OBJS-$(CONFIG_VOLUME_FILTER)                 += af_volume_stefano.o

-OBJS-$(CONFIG_VOLUME_JUSTIN_FILTER)          += af_volume_justin.o

+OBJS-$(CONFIG_VOLUME_FILTER)                 += af_volume.o

 OBJS-$(CONFIG_VOLUMEDETECT_FILTER)           += af_volumedetect.o

 OBJS-$(CONFIG_AEVALSRC_FILTER)               += asrc_aevalsrc.o

new file mode 100644

@@ -0,0 +1,311 @@

+/*

+ * Copyright (c) 2011 Stefano Sabatini

+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>

+ *

+ * 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

+ */

+

+#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

+#define F AV_OPT_FLAG_FILTERING_PARAM

+

+static const AVOption volume_options[] = {

+    { "volume", "set volume adjustment",

+            OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0, 0x7fffff, A|F },

+    { "precision", "select mathematical precision",

+            OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A|F, "precision" },

+        { "fixed",  "select 8-bit fixed-point",     0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED  }, INT_MIN, INT_MAX, A|F, "precision" },

+        { "float",  "select 32-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT  }, INT_MIN, INT_MAX, A|F, "precision" },

+        { "double", "select 64-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A|F, "precision" },

+    { NULL },

+};

+

+AVFILTER_DEFINE_CLASS(volume);

+

+static av_cold int init(AVFilterContext *ctx, const char *args)

+{

+    VolumeContext *vol = ctx->priv;

+    static const char *shorthand[] = { "volume", "precision", NULL };

+    int ret;

+

+    vol->class = &volume_class;

+    av_opt_set_defaults(vol);

+

+    if ((ret = av_opt_set_from_string(vol, args, shorthand, "=", ":")) < 0)

+        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;

+    }

+

+    if (ARCH_X86)

+        ff_volume_init_x86(vol);

+}

+

+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,

+    .priv_class     = &volume_class,

+};

deleted file mode 100644

@@ -1,311 +0,0 @@

-/*

- * Copyright (c) 2011 Stefano Sabatini

- * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>

- *

- * 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

- */

-

-#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

-#define F AV_OPT_FLAG_FILTERING_PARAM

-

-static const AVOption volume_options[] = {

-    { "volume", "set volume adjustment",

-            OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0, 0x7fffff, A|F },

-    { "precision", "select mathematical precision",

-            OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A|F, "precision" },

-        { "fixed",  "select 8-bit fixed-point",     0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED  }, INT_MIN, INT_MAX, A|F, "precision" },

-        { "float",  "select 32-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT  }, INT_MIN, INT_MAX, A|F, "precision" },

-        { "double", "select 64-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A|F, "precision" },

-    { NULL },

-};

-

-AVFILTER_DEFINE_CLASS(volume);

-

-static av_cold int init(AVFilterContext *ctx, const char *args)

-{

-    VolumeContext *vol = ctx->priv;

-    static const char *shorthand[] = { "volume", "precision", NULL };

-    int ret;

-

-    vol->class = &volume_class;

-    av_opt_set_defaults(vol);

-

-    if ((ret = av_opt_set_from_string(vol, args, shorthand, "=", ":")) < 0)

-        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;

-    }

-

-    if (ARCH_X86)

-        ff_volume_init_x86(vol);

-}

-

-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_justin = {

-    .name           = "volume_justin",

-    .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,

-    .priv_class     = &volume_class,

-};

deleted file mode 100644

@@ -1,201 +0,0 @@

-/*

- * 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/channel_layout.h"

-#include "libavutil/eval.h"

-#include "audio.h"

-#include "avfilter.h"

-#include "formats.h"

-

-typedef struct {

-    double volume;

-    int    volume_i;

-} VolumeContext;

-

-static av_cold int init(AVFilterContext *ctx, const char *args)

-{

-    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 */

-                d = pow(10, 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_VERBOSE, "volume=%f\n", vol->volume);

-    return 0;

-}

-

-static int query_formats(AVFilterContext *ctx)

-{

-    AVFilterFormats *formats = NULL;

-    AVFilterChannelLayouts *layouts;

-    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

-    };

-

-    layouts = ff_all_channel_layouts();

-    if (!layouts)

-        return AVERROR(ENOMEM);

-    ff_set_common_channel_layouts(ctx, layouts);

-

-    formats = ff_make_format_list(sample_fmts);

-    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 int filter_frame(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);