[FFmpeg-devel] [PATCH] lavfi: drop af_volume_stefano.c, in favor of af_volume_justin

Stefano Sabatini stefasab at gmail.com
Sat Dec 8 12:10:10 CET 2012


Justin's version has more features but is otherwise equivalent from the
point of view of the syntax.

TODO: bump micro
---
 doc/filters.texi                                |   66 ++------
 libavfilter/Makefile                            |    3 +-
 libavfilter/{af_volume_justin.c => af_volume.c} |    4 +-
 libavfilter/af_volume_stefano.c                 |  201 -----------------------
 libavfilter/allfilters.c                        |    1 -
 5 files changed, 14 insertions(+), 261 deletions(-)
 rename libavfilter/{af_volume_justin.c => af_volume.c} (99%)
 delete mode 100644 libavfilter/af_volume_stefano.c

diff --git a/doc/filters.texi b/doc/filters.texi
index cb2c899..a2d4a19 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -829,56 +829,6 @@ out
 Convert the audio sample format, sample rate and channel layout. This filter is
 not meant to be used directly.
 
- at 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:
- at example
- at var{output_volume} = @var{vol} * @var{input_volume}
- at 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:
- at example
- at var{output_volume} = 10^(@var{vol}/20) * @var{input_volume}
- at 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.
-
- at subsection Examples
-
- at itemize
- at item
-Half the input audio volume:
- at example
-volume=0.5
- at end example
-
-The above example is equivalent to:
- at example
-volume=1/2
- at end example
-
- at item
-Decrease input audio power by 12 decibels:
- at example
-volume=-12dB
- at end example
- at 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.
 
- at section volume_justin
+ at 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
+ at end example
+
+In all the above example the named key for @option{volume} can be
+omitted, for example like in:
+ at 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
 
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 7f9f0ef..377bd4d 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -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
diff --git a/libavfilter/af_volume_justin.c b/libavfilter/af_volume.c
similarity index 99%
rename from libavfilter/af_volume_justin.c
rename to libavfilter/af_volume.c
index 0ba466a..5ffa1fe 100644
--- a/libavfilter/af_volume_justin.c
+++ b/libavfilter/af_volume.c
@@ -299,8 +299,8 @@ static const AVFilterPad avfilter_af_volume_outputs[] = {
     { NULL }
 };
 
-AVFilter avfilter_af_volume_justin = {
-    .name           = "volume_justin",
+AVFilter avfilter_af_volume = {
+    .name           = "volume",
     .description    = NULL_IF_CONFIG_SMALL("Change input volume."),
     .query_formats  = query_formats,
     .priv_size      = sizeof(VolumeContext),
diff --git a/libavfilter/af_volume_stefano.c b/libavfilter/af_volume_stefano.c
deleted file mode 100644
index 7608083..0000000
--- a/libavfilter/af_volume_stefano.c
+++ /dev/null
@@ -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);
-                *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;
-        }
-        }
-    }
-    return ff_filter_frame(outlink, insamples);
-}
-
-static const AVFilterPad volume_inputs[] = {
-    {
-        .name         = "default",
-        .type         = AVMEDIA_TYPE_AUDIO,
-        .filter_frame = filter_frame,
-        .min_perms    = AV_PERM_READ | AV_PERM_WRITE,
-    },
-    { NULL },
-};
-
-static const AVFilterPad volume_outputs[] = {
-    {
-        .name = "default",
-        .type = AVMEDIA_TYPE_AUDIO,
-    },
-    { 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         = volume_inputs,
-    .outputs        = volume_outputs,
-};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index f9cacfc..ffde5ce 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -64,7 +64,6 @@ void avfilter_register_all(void)
     REGISTER_FILTER (RESAMPLE,    resample,    af);
     REGISTER_FILTER (SILENCEDETECT, silencedetect, af);
     REGISTER_FILTER (VOLUME,      volume,      af);
-    REGISTER_FILTER (VOLUME_JUSTIN, volume_justin, af);
     REGISTER_FILTER (VOLUMEDETECT,volumedetect,af);
 
     REGISTER_FILTER (AEVALSRC,    aevalsrc,    asrc);
-- 
1.7.9.5



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