2 * Copyright (c) 2011 Stefano Sabatini
3 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
5 * This file is part of Libav.
7 * Libav is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * Libav is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
27 #include "libavutil/channel_layout.h"
28 #include "libavutil/common.h"
29 #include "libavutil/eval.h"
30 #include "libavutil/float_dsp.h"
31 #include "libavutil/opt.h"
36 #include "af_volume.h"
38 static const char *precision_str[] = {
39 "fixed", "float", "double"
42 #define OFFSET(x) offsetof(VolumeContext, x)
43 #define A AV_OPT_FLAG_AUDIO_PARAM
45 static const AVOption options[] = {
46 { "volume", "Volume adjustment.",
47 OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0, 0x7fffff, A },
48 { "precision", "Mathematical precision.",
49 OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A, "precision" },
50 { "fixed", "8-bit fixed-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED }, INT_MIN, INT_MAX, A, "precision" },
51 { "float", "32-bit floating-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT }, INT_MIN, INT_MAX, A, "precision" },
52 { "double", "64-bit floating-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A, "precision" },
56 static const AVClass volume_class = {
57 .class_name = "volume filter",
58 .item_name = av_default_item_name,
60 .version = LIBAVUTIL_VERSION_INT,
63 static av_cold int init(AVFilterContext *ctx)
65 VolumeContext *vol = ctx->priv;
67 if (vol->precision == PRECISION_FIXED) {
68 vol->volume_i = (int)(vol->volume * 256 + 0.5);
69 vol->volume = vol->volume_i / 256.0;
70 av_log(ctx, AV_LOG_VERBOSE, "volume:(%d/256)(%f)(%1.2fdB) precision:fixed\n",
71 vol->volume_i, vol->volume, 20.0*log(vol->volume)/M_LN10);
73 av_log(ctx, AV_LOG_VERBOSE, "volume:(%f)(%1.2fdB) precision:%s\n",
74 vol->volume, 20.0*log(vol->volume)/M_LN10,
75 precision_str[vol->precision]);
81 static int query_formats(AVFilterContext *ctx)
83 VolumeContext *vol = ctx->priv;
84 AVFilterFormats *formats = NULL;
85 AVFilterChannelLayouts *layouts;
86 static const enum AVSampleFormat sample_fmts[][7] = {
103 /* PRECISION_DOUBLE */
111 layouts = ff_all_channel_layouts();
113 return AVERROR(ENOMEM);
114 ff_set_common_channel_layouts(ctx, layouts);
116 formats = ff_make_format_list(sample_fmts[vol->precision]);
118 return AVERROR(ENOMEM);
119 ff_set_common_formats(ctx, formats);
121 formats = ff_all_samplerates();
123 return AVERROR(ENOMEM);
124 ff_set_common_samplerates(ctx, formats);
129 static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
130 int nb_samples, int volume)
133 for (i = 0; i < nb_samples; i++)
134 dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
137 static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
138 int nb_samples, int volume)
141 for (i = 0; i < nb_samples; i++)
142 dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
145 static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,
146 int nb_samples, int volume)
149 int16_t *smp_dst = (int16_t *)dst;
150 const int16_t *smp_src = (const int16_t *)src;
151 for (i = 0; i < nb_samples; i++)
152 smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8);
155 static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
156 int nb_samples, int volume)
159 int16_t *smp_dst = (int16_t *)dst;
160 const int16_t *smp_src = (const int16_t *)src;
161 for (i = 0; i < nb_samples; i++)
162 smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
165 static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
166 int nb_samples, int volume)
169 int32_t *smp_dst = (int32_t *)dst;
170 const int32_t *smp_src = (const int32_t *)src;
171 for (i = 0; i < nb_samples; i++)
172 smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
177 static av_cold void volume_init(VolumeContext *vol)
179 vol->samples_align = 1;
181 switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
182 case AV_SAMPLE_FMT_U8:
183 if (vol->volume_i < 0x1000000)
184 vol->scale_samples = scale_samples_u8_small;
186 vol->scale_samples = scale_samples_u8;
188 case AV_SAMPLE_FMT_S16:
189 if (vol->volume_i < 0x10000)
190 vol->scale_samples = scale_samples_s16_small;
192 vol->scale_samples = scale_samples_s16;
194 case AV_SAMPLE_FMT_S32:
195 vol->scale_samples = scale_samples_s32;
197 case AV_SAMPLE_FMT_FLT:
198 avpriv_float_dsp_init(&vol->fdsp, 0);
199 vol->samples_align = 4;
201 case AV_SAMPLE_FMT_DBL:
202 avpriv_float_dsp_init(&vol->fdsp, 0);
203 vol->samples_align = 8;
208 ff_volume_init_x86(vol);
211 static int config_output(AVFilterLink *outlink)
213 AVFilterContext *ctx = outlink->src;
214 VolumeContext *vol = ctx->priv;
215 AVFilterLink *inlink = ctx->inputs[0];
217 vol->sample_fmt = inlink->format;
218 vol->channels = av_get_channel_layout_nb_channels(inlink->channel_layout);
219 vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
226 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
228 VolumeContext *vol = inlink->dst->priv;
229 AVFilterLink *outlink = inlink->dst->outputs[0];
230 int nb_samples = buf->nb_samples;
233 if (vol->volume == 1.0 || vol->volume_i == 256)
234 return ff_filter_frame(outlink, buf);
236 /* do volume scaling in-place if input buffer is writable */
237 if (av_frame_is_writable(buf)) {
240 out_buf = ff_get_audio_buffer(inlink, nb_samples);
242 return AVERROR(ENOMEM);
243 out_buf->pts = buf->pts;
246 if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
247 int p, plane_samples;
249 if (av_sample_fmt_is_planar(buf->format))
250 plane_samples = FFALIGN(nb_samples, vol->samples_align);
252 plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
254 if (vol->precision == PRECISION_FIXED) {
255 for (p = 0; p < vol->planes; p++) {
256 vol->scale_samples(out_buf->extended_data[p],
257 buf->extended_data[p], plane_samples,
260 } else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) {
261 for (p = 0; p < vol->planes; p++) {
262 vol->fdsp.vector_fmul_scalar((float *)out_buf->extended_data[p],
263 (const float *)buf->extended_data[p],
264 vol->volume, plane_samples);
267 for (p = 0; p < vol->planes; p++) {
268 vol->fdsp.vector_dmul_scalar((double *)out_buf->extended_data[p],
269 (const double *)buf->extended_data[p],
270 vol->volume, plane_samples);
278 return ff_filter_frame(outlink, out_buf);
281 static const AVFilterPad avfilter_af_volume_inputs[] = {
284 .type = AVMEDIA_TYPE_AUDIO,
285 .filter_frame = filter_frame,
290 static const AVFilterPad avfilter_af_volume_outputs[] = {
293 .type = AVMEDIA_TYPE_AUDIO,
294 .config_props = config_output,
299 AVFilter ff_af_volume = {
301 .description = NULL_IF_CONFIG_SMALL("Change input volume."),
302 .query_formats = query_formats,
303 .priv_size = sizeof(VolumeContext),
304 .priv_class = &volume_class,
306 .inputs = avfilter_af_volume_inputs,
307 .outputs = avfilter_af_volume_outputs,