avcodec/vdpau: fix assertion failure and < vs > error
[ffmpeg.git] / libavfilter / af_aphaser.c
1 /*
2  * Copyright (c) 2013 Paul B Mahol
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20
21 /**
22  * @file
23  * phaser audio filter
24  */
25
26 #include "libavutil/avassert.h"
27 #include "libavutil/opt.h"
28 #include "audio.h"
29 #include "avfilter.h"
30 #include "internal.h"
31 #include "generate_wave_table.h"
32
33 typedef struct AudioPhaserContext {
34     const AVClass *class;
35     double in_gain, out_gain;
36     double delay;
37     double decay;
38     double speed;
39
40     enum WaveType type;
41
42     int delay_buffer_length;
43     double *delay_buffer;
44
45     int modulation_buffer_length;
46     int32_t *modulation_buffer;
47
48     int delay_pos, modulation_pos;
49
50     void (*phaser)(struct AudioPhaserContext *p,
51                    uint8_t * const *src, uint8_t **dst,
52                    int nb_samples, int channels);
53 } AudioPhaserContext;
54
55 #define OFFSET(x) offsetof(AudioPhaserContext, x)
56 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
57
58 static const AVOption aphaser_options[] = {
59     { "in_gain",  "set input gain",            OFFSET(in_gain),  AV_OPT_TYPE_DOUBLE, {.dbl=.4},  0,  1,   FLAGS },
60     { "out_gain", "set output gain",           OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0,  1e9, FLAGS },
61     { "delay",    "set delay in milliseconds", OFFSET(delay),    AV_OPT_TYPE_DOUBLE, {.dbl=3.},  0,  5,   FLAGS },
62     { "decay",    "set decay",                 OFFSET(decay),    AV_OPT_TYPE_DOUBLE, {.dbl=.4},  0, .99,  FLAGS },
63     { "speed",    "set modulation speed",      OFFSET(speed),    AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1,  2,   FLAGS },
64     { "type",     "set modulation type",       OFFSET(type),     AV_OPT_TYPE_INT,    {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" },
65     { "triangular",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
66     { "t",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
67     { "sinusoidal",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
68     { "s",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
69     { NULL }
70 };
71
72 AVFILTER_DEFINE_CLASS(aphaser);
73
74 static av_cold int init(AVFilterContext *ctx)
75 {
76     AudioPhaserContext *p = ctx->priv;
77
78     if (p->in_gain > (1 - p->decay * p->decay))
79         av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
80     if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)
81         av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
82
83     return 0;
84 }
85
86 static int query_formats(AVFilterContext *ctx)
87 {
88     AVFilterFormats *formats;
89     AVFilterChannelLayouts *layouts;
90     static const enum AVSampleFormat sample_fmts[] = {
91         AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
92         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
93         AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
94         AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
95         AV_SAMPLE_FMT_NONE
96     };
97
98     layouts = ff_all_channel_layouts();
99     if (!layouts)
100         return AVERROR(ENOMEM);
101     ff_set_common_channel_layouts(ctx, layouts);
102
103     formats = ff_make_format_list(sample_fmts);
104     if (!formats)
105         return AVERROR(ENOMEM);
106     ff_set_common_formats(ctx, formats);
107
108     formats = ff_all_samplerates();
109     if (!formats)
110         return AVERROR(ENOMEM);
111     ff_set_common_samplerates(ctx, formats);
112
113     return 0;
114 }
115
116 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
117
118 #define PHASER_PLANAR(name, type)                                      \
119 static void phaser_## name ##p(AudioPhaserContext *p,                  \
120                                uint8_t * const *src, uint8_t **dst,    \
121                                int nb_samples, int channels)           \
122 {                                                                      \
123     int i, c, delay_pos, modulation_pos;                               \
124                                                                        \
125     av_assert0(channels > 0);                                          \
126     for (c = 0; c < channels; c++) {                                   \
127         type *s = (type *)src[c];                                      \
128         type *d = (type *)dst[c];                                      \
129         double *buffer = p->delay_buffer +                             \
130                          c * p->delay_buffer_length;                   \
131                                                                        \
132         delay_pos      = p->delay_pos;                                 \
133         modulation_pos = p->modulation_pos;                            \
134                                                                        \
135         for (i = 0; i < nb_samples; i++, s++, d++) {                   \
136             double v = *s * p->in_gain + buffer[                       \
137                        MOD(delay_pos + p->modulation_buffer[           \
138                        modulation_pos],                                \
139                        p->delay_buffer_length)] * p->decay;            \
140                                                                        \
141             modulation_pos = MOD(modulation_pos + 1,                   \
142                              p->modulation_buffer_length);             \
143             delay_pos = MOD(delay_pos + 1, p->delay_buffer_length);    \
144             buffer[delay_pos] = v;                                     \
145                                                                        \
146             *d = v * p->out_gain;                                      \
147         }                                                              \
148     }                                                                  \
149                                                                        \
150     p->delay_pos      = delay_pos;                                     \
151     p->modulation_pos = modulation_pos;                                \
152 }
153
154 #define PHASER(name, type)                                              \
155 static void phaser_## name (AudioPhaserContext *p,                      \
156                             uint8_t * const *src, uint8_t **dst,        \
157                             int nb_samples, int channels)               \
158 {                                                                       \
159     int i, c, delay_pos, modulation_pos;                                \
160     type *s = (type *)src[0];                                           \
161     type *d = (type *)dst[0];                                           \
162     double *buffer = p->delay_buffer;                                   \
163                                                                         \
164     delay_pos      = p->delay_pos;                                      \
165     modulation_pos = p->modulation_pos;                                 \
166                                                                         \
167     for (i = 0; i < nb_samples; i++) {                                  \
168         int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \
169                    p->delay_buffer_length) * channels;                  \
170         int npos;                                                       \
171                                                                         \
172         delay_pos = MOD(delay_pos + 1, p->delay_buffer_length);         \
173         npos = delay_pos * channels;                                    \
174         for (c = 0; c < channels; c++, s++, d++) {                      \
175             double v = *s * p->in_gain + buffer[pos + c] * p->decay;    \
176                                                                         \
177             buffer[npos + c] = v;                                       \
178                                                                         \
179             *d = v * p->out_gain;                                       \
180         }                                                               \
181                                                                         \
182         modulation_pos = MOD(modulation_pos + 1,                        \
183                          p->modulation_buffer_length);                  \
184     }                                                                   \
185                                                                         \
186     p->delay_pos      = delay_pos;                                      \
187     p->modulation_pos = modulation_pos;                                 \
188 }
189
190 PHASER_PLANAR(dbl, double)
191 PHASER_PLANAR(flt, float)
192 PHASER_PLANAR(s16, int16_t)
193 PHASER_PLANAR(s32, int32_t)
194
195 PHASER(dbl, double)
196 PHASER(flt, float)
197 PHASER(s16, int16_t)
198 PHASER(s32, int32_t)
199
200 static int config_output(AVFilterLink *outlink)
201 {
202     AudioPhaserContext *p = outlink->src->priv;
203     AVFilterLink *inlink = outlink->src->inputs[0];
204
205     p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;
206     p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);
207     p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;
208     p->modulation_buffer = av_malloc_array(p->modulation_buffer_length, sizeof(*p->modulation_buffer));
209
210     if (!p->modulation_buffer || !p->delay_buffer)
211         return AVERROR(ENOMEM);
212
213     ff_generate_wave_table(p->type, AV_SAMPLE_FMT_S32,
214                            p->modulation_buffer, p->modulation_buffer_length,
215                            1., p->delay_buffer_length, M_PI / 2.0);
216
217     p->delay_pos = p->modulation_pos = 0;
218
219     switch (inlink->format) {
220     case AV_SAMPLE_FMT_DBL:  p->phaser = phaser_dbl;  break;
221     case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;
222     case AV_SAMPLE_FMT_FLT:  p->phaser = phaser_flt;  break;
223     case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;
224     case AV_SAMPLE_FMT_S16:  p->phaser = phaser_s16;  break;
225     case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;
226     case AV_SAMPLE_FMT_S32:  p->phaser = phaser_s32;  break;
227     case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;
228     default: av_assert0(0);
229     }
230
231     return 0;
232 }
233
234 static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
235 {
236     AudioPhaserContext *p = inlink->dst->priv;
237     AVFilterLink *outlink = inlink->dst->outputs[0];
238     AVFrame *outbuf;
239
240     if (av_frame_is_writable(inbuf)) {
241         outbuf = inbuf;
242     } else {
243         outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);
244         if (!outbuf)
245             return AVERROR(ENOMEM);
246         av_frame_copy_props(outbuf, inbuf);
247     }
248
249     p->phaser(p, inbuf->extended_data, outbuf->extended_data,
250               outbuf->nb_samples, av_frame_get_channels(outbuf));
251
252     if (inbuf != outbuf)
253         av_frame_free(&inbuf);
254
255     return ff_filter_frame(outlink, outbuf);
256 }
257
258 static av_cold void uninit(AVFilterContext *ctx)
259 {
260     AudioPhaserContext *p = ctx->priv;
261
262     av_freep(&p->delay_buffer);
263     av_freep(&p->modulation_buffer);
264 }
265
266 static const AVFilterPad aphaser_inputs[] = {
267     {
268         .name         = "default",
269         .type         = AVMEDIA_TYPE_AUDIO,
270         .filter_frame = filter_frame,
271     },
272     { NULL }
273 };
274
275 static const AVFilterPad aphaser_outputs[] = {
276     {
277         .name         = "default",
278         .type         = AVMEDIA_TYPE_AUDIO,
279         .config_props = config_output,
280     },
281     { NULL }
282 };
283
284 AVFilter ff_af_aphaser = {
285     .name          = "aphaser",
286     .description   = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
287     .query_formats = query_formats,
288     .priv_size     = sizeof(AudioPhaserContext),
289     .init          = init,
290     .uninit        = uninit,
291     .inputs        = aphaser_inputs,
292     .outputs       = aphaser_outputs,
293     .priv_class    = &aphaser_class,
294 };