/* * 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/channel_layout.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) { VolumeContext *vol = ctx->priv; 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]); } return 0; } 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 av_cold 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, AVFrame *buf) { VolumeContext *vol = inlink->dst->priv; AVFilterLink *outlink = inlink->dst->outputs[0]; int nb_samples = buf->nb_samples; AVFrame *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 (av_frame_is_writable(buf)) { out_buf = buf; } else { out_buf = ff_get_audio_buffer(inlink, nb_samples); if (!out_buf) return AVERROR(ENOMEM); av_frame_copy_props(out_buf, buf); } 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) av_frame_free(&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 ff_af_volume = { .name = "volume", .description = NULL_IF_CONFIG_SMALL("Change input volume."), .query_formats = query_formats, .priv_size = sizeof(VolumeContext), .priv_class = &volume_class, .init = init, .inputs = avfilter_af_volume_inputs, .outputs = avfilter_af_volume_outputs, .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, };