From 10db70d5e9c92f0464089476694d9a440cdac321 Mon Sep 17 00:00:00 2001 From: Stefano Sabatini Date: Sat, 8 Dec 2012 12:07:03 +0100 Subject: lavfi: drop af_volume_stefano.c in favor of af_volume_justin.c Justin's version has more features but is otherwise equivalent from the point of view of the syntax. --- doc/filters.texi | 66 ++------- libavfilter/Makefile | 3 +- libavfilter/af_volume.c | 311 ++++++++++++++++++++++++++++++++++++++++ libavfilter/af_volume_justin.c | 311 ---------------------------------------- libavfilter/af_volume_stefano.c | 201 -------------------------- libavfilter/allfilters.c | 1 - libavfilter/version.h | 4 +- 7 files changed, 325 insertions(+), 572 deletions(-) create mode 100644 libavfilter/af_volume.c delete mode 100644 libavfilter/af_volume_justin.c delete mode 100644 libavfilter/af_volume_stefano.c diff --git a/doc/filters.texi b/doc/filters.texi index 8e70cdb357..56a213ccc6 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. -@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 diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 7f9f0ef2a5..377bd4d701 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.c b/libavfilter/af_volume.c new file mode 100644 index 0000000000..5ffa1fea4f --- /dev/null +++ b/libavfilter/af_volume.c @@ -0,0 +1,311 @@ +/* + * Copyright (c) 2011 Stefano Sabatini + * Copyright (c) 2012 Justin Ruggles + * + * 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, +}; diff --git a/libavfilter/af_volume_justin.c b/libavfilter/af_volume_justin.c deleted file mode 100644 index 0ba466a348..0000000000 --- a/libavfilter/af_volume_justin.c +++ /dev/null @@ -1,311 +0,0 @@ -/* - * Copyright (c) 2011 Stefano Sabatini - * Copyright (c) 2012 Justin Ruggles - * - * 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, -}; diff --git a/libavfilter/af_volume_stefano.c b/libavfilter/af_volume_stefano.c deleted file mode 100644 index 7608083640..0000000000 --- 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 f9cacfc9bf..ffde5ce112 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); diff --git a/libavfilter/version.h b/libavfilter/version.h index 68a3543179..ed17978c2e 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -29,8 +29,8 @@ #include "libavutil/avutil.h" #define LIBAVFILTER_VERSION_MAJOR 3 -#define LIBAVFILTER_VERSION_MINOR 25 -#define LIBAVFILTER_VERSION_MICRO 102 +#define LIBAVFILTER_VERSION_MINOR 26 +#define LIBAVFILTER_VERSION_MICRO 100 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \ LIBAVFILTER_VERSION_MINOR, \ -- cgit v1.2.3