2 * Copyright (c) 2011 Stefano Sabatini
3 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
5 * This file is part of FFmpeg.
7 * FFmpeg 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 * FFmpeg 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 FFmpeg; 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
44 #define F AV_OPT_FLAG_FILTERING_PARAM
46 static const AVOption volume_options[] = {
47 { "volume", "set volume adjustment",
48 OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0, 0x7fffff, A|F },
49 { "precision", "select mathematical precision",
50 OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A|F, "precision" },
51 { "fixed", "select 8-bit fixed-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED }, INT_MIN, INT_MAX, A|F, "precision" },
52 { "float", "select 32-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT }, INT_MIN, INT_MAX, A|F, "precision" },
53 { "double", "select 64-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A|F, "precision" },
57 AVFILTER_DEFINE_CLASS(volume);
59 static av_cold int init(AVFilterContext *ctx)
61 VolumeContext *vol = ctx->priv;
63 if (vol->precision == PRECISION_FIXED) {
64 vol->volume_i = (int)(vol->volume * 256 + 0.5);
65 vol->volume = vol->volume_i / 256.0;
66 av_log(ctx, AV_LOG_VERBOSE, "volume:(%d/256)(%f)(%1.2fdB) precision:fixed\n",
67 vol->volume_i, vol->volume, 20.0*log(vol->volume)/M_LN10);
69 av_log(ctx, AV_LOG_VERBOSE, "volume:(%f)(%1.2fdB) precision:%s\n",
70 vol->volume, 20.0*log(vol->volume)/M_LN10,
71 precision_str[vol->precision]);
77 static int query_formats(AVFilterContext *ctx)
79 VolumeContext *vol = ctx->priv;
80 AVFilterFormats *formats = NULL;
81 AVFilterChannelLayouts *layouts;
82 static const enum AVSampleFormat sample_fmts[][7] = {
97 [PRECISION_DOUBLE] = {
104 layouts = ff_all_channel_layouts();
106 return AVERROR(ENOMEM);
107 ff_set_common_channel_layouts(ctx, layouts);
109 formats = ff_make_format_list(sample_fmts[vol->precision]);
111 return AVERROR(ENOMEM);
112 ff_set_common_formats(ctx, formats);
114 formats = ff_all_samplerates();
116 return AVERROR(ENOMEM);
117 ff_set_common_samplerates(ctx, formats);
122 static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
123 int nb_samples, int volume)
126 for (i = 0; i < nb_samples; i++)
127 dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
130 static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
131 int nb_samples, int volume)
134 for (i = 0; i < nb_samples; i++)
135 dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
138 static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,
139 int nb_samples, int volume)
142 int16_t *smp_dst = (int16_t *)dst;
143 const int16_t *smp_src = (const int16_t *)src;
144 for (i = 0; i < nb_samples; i++)
145 smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8);
148 static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
149 int nb_samples, int volume)
152 int16_t *smp_dst = (int16_t *)dst;
153 const int16_t *smp_src = (const int16_t *)src;
154 for (i = 0; i < nb_samples; i++)
155 smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
158 static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
159 int nb_samples, int volume)
162 int32_t *smp_dst = (int32_t *)dst;
163 const int32_t *smp_src = (const int32_t *)src;
164 for (i = 0; i < nb_samples; i++)
165 smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
168 static av_cold void volume_init(VolumeContext *vol)
170 vol->samples_align = 1;
172 switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
173 case AV_SAMPLE_FMT_U8:
174 if (vol->volume_i < 0x1000000)
175 vol->scale_samples = scale_samples_u8_small;
177 vol->scale_samples = scale_samples_u8;
179 case AV_SAMPLE_FMT_S16:
180 if (vol->volume_i < 0x10000)
181 vol->scale_samples = scale_samples_s16_small;
183 vol->scale_samples = scale_samples_s16;
185 case AV_SAMPLE_FMT_S32:
186 vol->scale_samples = scale_samples_s32;
188 case AV_SAMPLE_FMT_FLT:
189 avpriv_float_dsp_init(&vol->fdsp, 0);
190 vol->samples_align = 4;
192 case AV_SAMPLE_FMT_DBL:
193 avpriv_float_dsp_init(&vol->fdsp, 0);
194 vol->samples_align = 8;
199 ff_volume_init_x86(vol);
202 static int config_output(AVFilterLink *outlink)
204 AVFilterContext *ctx = outlink->src;
205 VolumeContext *vol = ctx->priv;
206 AVFilterLink *inlink = ctx->inputs[0];
208 vol->sample_fmt = inlink->format;
209 vol->channels = av_get_channel_layout_nb_channels(inlink->channel_layout);
210 vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
217 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
219 VolumeContext *vol = inlink->dst->priv;
220 AVFilterLink *outlink = inlink->dst->outputs[0];
221 int nb_samples = buf->nb_samples;
224 if (vol->volume == 1.0 || vol->volume_i == 256)
225 return ff_filter_frame(outlink, buf);
227 /* do volume scaling in-place if input buffer is writable */
228 if (av_frame_is_writable(buf)) {
231 out_buf = ff_get_audio_buffer(inlink, nb_samples);
233 return AVERROR(ENOMEM);
234 av_frame_copy_props(out_buf, buf);
237 if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
238 int p, plane_samples;
240 if (av_sample_fmt_is_planar(buf->format))
241 plane_samples = FFALIGN(nb_samples, vol->samples_align);
243 plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
245 if (vol->precision == PRECISION_FIXED) {
246 for (p = 0; p < vol->planes; p++) {
247 vol->scale_samples(out_buf->extended_data[p],
248 buf->extended_data[p], plane_samples,
251 } else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) {
252 for (p = 0; p < vol->planes; p++) {
253 vol->fdsp.vector_fmul_scalar((float *)out_buf->extended_data[p],
254 (const float *)buf->extended_data[p],
255 vol->volume, plane_samples);
258 for (p = 0; p < vol->planes; p++) {
259 vol->fdsp.vector_dmul_scalar((double *)out_buf->extended_data[p],
260 (const double *)buf->extended_data[p],
261 vol->volume, plane_samples);
269 return ff_filter_frame(outlink, out_buf);
272 static const AVFilterPad avfilter_af_volume_inputs[] = {
275 .type = AVMEDIA_TYPE_AUDIO,
276 .filter_frame = filter_frame,
281 static const AVFilterPad avfilter_af_volume_outputs[] = {
284 .type = AVMEDIA_TYPE_AUDIO,
285 .config_props = config_output,
290 AVFilter avfilter_af_volume = {
292 .description = NULL_IF_CONFIG_SMALL("Change input volume."),
293 .query_formats = query_formats,
294 .priv_size = sizeof(VolumeContext),
295 .priv_class = &volume_class,
297 .inputs = avfilter_af_volume_inputs,
298 .outputs = avfilter_af_volume_outputs,
299 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,