318 lines
10 KiB
C
318 lines
10 KiB
C
/*
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* Copyright (c) 2011 Stefano Sabatini
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* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* audio volume filter
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*/
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#include "libavutil/audioconvert.h"
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#include "libavutil/common.h"
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#include "libavutil/eval.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "formats.h"
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#include "internal.h"
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#include "af_volume.h"
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static const char *precision_str[] = {
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"fixed", "float", "double"
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};
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#define OFFSET(x) offsetof(VolumeContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM
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static const AVOption options[] = {
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{ "volume", "Volume adjustment.",
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OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0, 0x7fffff, A },
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{ "precision", "Mathematical precision.",
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OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A, "precision" },
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{ "fixed", "8-bit fixed-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED }, INT_MIN, INT_MAX, A, "precision" },
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{ "float", "32-bit floating-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT }, INT_MIN, INT_MAX, A, "precision" },
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{ "double", "64-bit floating-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A, "precision" },
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{ NULL },
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};
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static const AVClass volume_class = {
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.class_name = "volume filter",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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static av_cold int init(AVFilterContext *ctx, const char *args)
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{
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VolumeContext *vol = ctx->priv;
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int ret;
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vol->class = &volume_class;
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av_opt_set_defaults(vol);
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if ((ret = av_set_options_string(vol, args, "=", ":")) < 0) {
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av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args);
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return ret;
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}
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if (vol->precision == PRECISION_FIXED) {
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vol->volume_i = (int)(vol->volume * 256 + 0.5);
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vol->volume = vol->volume_i / 256.0;
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av_log(ctx, AV_LOG_VERBOSE, "volume:(%d/256)(%f)(%1.2fdB) precision:fixed\n",
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vol->volume_i, vol->volume, 20.0*log(vol->volume)/M_LN10);
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} else {
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av_log(ctx, AV_LOG_VERBOSE, "volume:(%f)(%1.2fdB) precision:%s\n",
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vol->volume, 20.0*log(vol->volume)/M_LN10,
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precision_str[vol->precision]);
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}
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av_opt_free(vol);
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return ret;
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}
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static int query_formats(AVFilterContext *ctx)
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{
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VolumeContext *vol = ctx->priv;
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AVFilterFormats *formats = NULL;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[][7] = {
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/* PRECISION_FIXED */
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{
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AV_SAMPLE_FMT_U8,
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AV_SAMPLE_FMT_U8P,
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S32,
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AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_NONE
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},
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/* PRECISION_FLOAT */
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{
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AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_NONE
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},
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/* PRECISION_DOUBLE */
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{
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AV_SAMPLE_FMT_DBL,
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AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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}
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};
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layouts = ff_all_channel_layouts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ff_set_common_channel_layouts(ctx, layouts);
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formats = ff_make_format_list(sample_fmts[vol->precision]);
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if (!formats)
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return AVERROR(ENOMEM);
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ff_set_common_formats(ctx, formats);
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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ff_set_common_samplerates(ctx, formats);
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return 0;
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}
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static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
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int nb_samples, int volume)
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{
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int i;
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for (i = 0; i < nb_samples; i++)
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dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
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}
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static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
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int nb_samples, int volume)
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{
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int i;
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for (i = 0; i < nb_samples; i++)
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dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
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}
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static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,
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int nb_samples, int volume)
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{
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int i;
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int16_t *smp_dst = (int16_t *)dst;
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const int16_t *smp_src = (const int16_t *)src;
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for (i = 0; i < nb_samples; i++)
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smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8);
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}
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static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
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int nb_samples, int volume)
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{
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int i;
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int16_t *smp_dst = (int16_t *)dst;
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const int16_t *smp_src = (const int16_t *)src;
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for (i = 0; i < nb_samples; i++)
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smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
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}
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static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
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int nb_samples, int volume)
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{
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int i;
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int32_t *smp_dst = (int32_t *)dst;
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const int32_t *smp_src = (const int32_t *)src;
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for (i = 0; i < nb_samples; i++)
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smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
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}
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static void volume_init(VolumeContext *vol)
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{
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vol->samples_align = 1;
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switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
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case AV_SAMPLE_FMT_U8:
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if (vol->volume_i < 0x1000000)
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vol->scale_samples = scale_samples_u8_small;
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else
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vol->scale_samples = scale_samples_u8;
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break;
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case AV_SAMPLE_FMT_S16:
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if (vol->volume_i < 0x10000)
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vol->scale_samples = scale_samples_s16_small;
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else
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vol->scale_samples = scale_samples_s16;
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break;
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case AV_SAMPLE_FMT_S32:
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vol->scale_samples = scale_samples_s32;
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break;
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case AV_SAMPLE_FMT_FLT:
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avpriv_float_dsp_init(&vol->fdsp, 0);
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vol->samples_align = 4;
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break;
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case AV_SAMPLE_FMT_DBL:
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avpriv_float_dsp_init(&vol->fdsp, 0);
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vol->samples_align = 8;
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break;
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}
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if (ARCH_X86)
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ff_volume_init_x86(vol);
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}
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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VolumeContext *vol = ctx->priv;
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AVFilterLink *inlink = ctx->inputs[0];
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vol->sample_fmt = inlink->format;
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vol->channels = av_get_channel_layout_nb_channels(inlink->channel_layout);
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vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
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volume_init(vol);
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
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{
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VolumeContext *vol = inlink->dst->priv;
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AVFilterLink *outlink = inlink->dst->outputs[0];
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int nb_samples = buf->audio->nb_samples;
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AVFilterBufferRef *out_buf;
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if (vol->volume == 1.0 || vol->volume_i == 256)
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return ff_filter_frame(outlink, buf);
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/* do volume scaling in-place if input buffer is writable */
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if (buf->perms & AV_PERM_WRITE) {
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out_buf = buf;
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} else {
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out_buf = ff_get_audio_buffer(inlink, AV_PERM_WRITE, nb_samples);
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if (!out_buf)
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return AVERROR(ENOMEM);
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out_buf->pts = buf->pts;
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}
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if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
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int p, plane_samples;
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if (av_sample_fmt_is_planar(buf->format))
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plane_samples = FFALIGN(nb_samples, vol->samples_align);
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else
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plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
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if (vol->precision == PRECISION_FIXED) {
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for (p = 0; p < vol->planes; p++) {
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vol->scale_samples(out_buf->extended_data[p],
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buf->extended_data[p], plane_samples,
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vol->volume_i);
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}
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} else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) {
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for (p = 0; p < vol->planes; p++) {
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vol->fdsp.vector_fmul_scalar((float *)out_buf->extended_data[p],
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(const float *)buf->extended_data[p],
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vol->volume, plane_samples);
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}
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} else {
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for (p = 0; p < vol->planes; p++) {
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vol->fdsp.vector_dmul_scalar((double *)out_buf->extended_data[p],
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(const double *)buf->extended_data[p],
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vol->volume, plane_samples);
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}
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}
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}
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if (buf != out_buf)
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avfilter_unref_buffer(buf);
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return ff_filter_frame(outlink, out_buf);
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}
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static const AVFilterPad avfilter_af_volume_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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},
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{ NULL }
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};
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static const AVFilterPad avfilter_af_volume_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_output,
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},
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{ NULL }
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};
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AVFilter avfilter_af_volume = {
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.name = "volume",
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.description = NULL_IF_CONFIG_SMALL("Change input volume."),
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.query_formats = query_formats,
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.priv_size = sizeof(VolumeContext),
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.init = init,
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.inputs = avfilter_af_volume_inputs,
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.outputs = avfilter_af_volume_outputs,
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};
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