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. --- libavfilter/af_volume.c | 311 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 311 insertions(+) create mode 100644 libavfilter/af_volume.c (limited to 'libavfilter/af_volume.c') diff --git a/libavfilter/af_volume.c b/libavfilter/af_volume.c new file mode 100644 index 0000000..5ffa1fe --- /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, +}; -- cgit v1.1