diff options
Diffstat (limited to 'libavfilter/af_aphaser.c')
| -rw-r--r-- | libavfilter/af_aphaser.c | 358 |
1 files changed, 358 insertions, 0 deletions
diff --git a/libavfilter/af_aphaser.c b/libavfilter/af_aphaser.c new file mode 100644 index 0000000000..0fc45eab43 --- /dev/null +++ b/libavfilter/af_aphaser.c @@ -0,0 +1,358 @@ +/* + * Copyright (c) 2013 Paul B Mahol + * + * 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 + * phaser audio filter + */ + +#include "libavutil/avassert.h" +#include "libavutil/opt.h" +#include "audio.h" +#include "avfilter.h" +#include "internal.h" + +enum WaveType { + WAVE_SIN, + WAVE_TRI, + WAVE_NB, +}; + +typedef struct AudioPhaserContext { + const AVClass *class; + double in_gain, out_gain; + double delay; + double decay; + double speed; + + enum WaveType type; + + int delay_buffer_length; + double *delay_buffer; + + int modulation_buffer_length; + int32_t *modulation_buffer; + + int delay_pos, modulation_pos; + + void (*phaser)(struct AudioPhaserContext *p, + uint8_t * const *src, uint8_t **dst, + int nb_samples, int channels); +} AudioPhaserContext; + +#define OFFSET(x) offsetof(AudioPhaserContext, x) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption aphaser_options[] = { + { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS }, + { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS }, + { "delay", "set delay in milliseconds", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS }, + { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS }, + { "speed", "set modulation speed", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS }, + { "type", "set modulation type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" }, + { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" }, + { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" }, + { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" }, + { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(aphaser); + +static av_cold int init(AVFilterContext *ctx) +{ + AudioPhaserContext *p = ctx->priv; + + if (p->in_gain > (1 - p->decay * p->decay)) + av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n"); + if (p->in_gain / (1 - p->decay) > 1 / p->out_gain) + av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n"); + + return 0; +} + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats; + AVFilterChannelLayouts *layouts; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP, + AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, + AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P, + AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P, + 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); + 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 void generate_wave_table(enum WaveType wave_type, enum AVSampleFormat sample_fmt, + void *table, int table_size, + double min, double max, double phase) +{ + uint32_t i, phase_offset = phase / M_PI / 2 * table_size + 0.5; + + for (i = 0; i < table_size; i++) { + uint32_t point = (i + phase_offset) % table_size; + double d; + + switch (wave_type) { + case WAVE_SIN: + d = (sin((double)point / table_size * 2 * M_PI) + 1) / 2; + break; + case WAVE_TRI: + d = (double)point * 2 / table_size; + switch (4 * point / table_size) { + case 0: d = d + 0.5; break; + case 1: + case 2: d = 1.5 - d; break; + case 3: d = d - 1.5; break; + } + break; + default: + av_assert0(0); + } + + d = d * (max - min) + min; + switch (sample_fmt) { + case AV_SAMPLE_FMT_FLT: { + float *fp = (float *)table; + *fp++ = (float)d; + table = fp; + continue; } + case AV_SAMPLE_FMT_DBL: { + double *dp = (double *)table; + *dp++ = d; + table = dp; + continue; } + } + + d += d < 0 ? -0.5 : 0.5; + switch (sample_fmt) { + case AV_SAMPLE_FMT_S16: { + int16_t *sp = table; + *sp++ = (int16_t)d; + table = sp; + continue; } + case AV_SAMPLE_FMT_S32: { + int32_t *ip = table; + *ip++ = (int32_t)d; + table = ip; + continue; } + default: + av_assert0(0); + } + } +} + +#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a)) + +#define PHASER_PLANAR(name, type) \ +static void phaser_## name ##p(AudioPhaserContext *p, \ + uint8_t * const *src, uint8_t **dst, \ + int nb_samples, int channels) \ +{ \ + int i, c, delay_pos, modulation_pos; \ + \ + av_assert0(channels > 0); \ + for (c = 0; c < channels; c++) { \ + type *s = (type *)src[c]; \ + type *d = (type *)dst[c]; \ + double *buffer = p->delay_buffer + \ + c * p->delay_buffer_length; \ + \ + delay_pos = p->delay_pos; \ + modulation_pos = p->modulation_pos; \ + \ + for (i = 0; i < nb_samples; i++, s++, d++) { \ + double v = *s * p->in_gain + buffer[ \ + MOD(delay_pos + p->modulation_buffer[ \ + modulation_pos], \ + p->delay_buffer_length)] * p->decay; \ + \ + modulation_pos = MOD(modulation_pos + 1, \ + p->modulation_buffer_length); \ + delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \ + buffer[delay_pos] = v; \ + \ + *d = v * p->out_gain; \ + } \ + } \ + \ + p->delay_pos = delay_pos; \ + p->modulation_pos = modulation_pos; \ +} + +#define PHASER(name, type) \ +static void phaser_## name (AudioPhaserContext *p, \ + uint8_t * const *src, uint8_t **dst, \ + int nb_samples, int channels) \ +{ \ + int i, c, delay_pos, modulation_pos; \ + type *s = (type *)src[0]; \ + type *d = (type *)dst[0]; \ + double *buffer = p->delay_buffer; \ + \ + delay_pos = p->delay_pos; \ + modulation_pos = p->modulation_pos; \ + \ + for (i = 0; i < nb_samples; i++) { \ + int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \ + p->delay_buffer_length) * channels; \ + int npos; \ + \ + delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \ + npos = delay_pos * channels; \ + for (c = 0; c < channels; c++, s++, d++) { \ + double v = *s * p->in_gain + buffer[pos + c] * p->decay; \ + \ + buffer[npos + c] = v; \ + \ + *d = v * p->out_gain; \ + } \ + \ + modulation_pos = MOD(modulation_pos + 1, \ + p->modulation_buffer_length); \ + } \ + \ + p->delay_pos = delay_pos; \ + p->modulation_pos = modulation_pos; \ +} + +PHASER_PLANAR(dbl, double) +PHASER_PLANAR(flt, float) +PHASER_PLANAR(s16, int16_t) +PHASER_PLANAR(s32, int32_t) + +PHASER(dbl, double) +PHASER(flt, float) +PHASER(s16, int16_t) +PHASER(s32, int32_t) + +static int config_output(AVFilterLink *outlink) +{ + AudioPhaserContext *p = outlink->src->priv; + AVFilterLink *inlink = outlink->src->inputs[0]; + + p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5; + p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels); + p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5; + p->modulation_buffer = av_malloc(p->modulation_buffer_length * sizeof(*p->modulation_buffer)); + + if (!p->modulation_buffer || !p->delay_buffer) + return AVERROR(ENOMEM); + + generate_wave_table(p->type, AV_SAMPLE_FMT_S32, + p->modulation_buffer, p->modulation_buffer_length, + 1., p->delay_buffer_length, M_PI / 2.0); + + p->delay_pos = p->modulation_pos = 0; + + switch (inlink->format) { + case AV_SAMPLE_FMT_DBL: p->phaser = phaser_dbl; break; + case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break; + case AV_SAMPLE_FMT_FLT: p->phaser = phaser_flt; break; + case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break; + case AV_SAMPLE_FMT_S16: p->phaser = phaser_s16; break; + case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break; + case AV_SAMPLE_FMT_S32: p->phaser = phaser_s32; break; + case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break; + default: av_assert0(0); + } + + return 0; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf) +{ + AudioPhaserContext *p = inlink->dst->priv; + AVFilterLink *outlink = inlink->dst->outputs[0]; + AVFrame *outbuf; + + if (av_frame_is_writable(inbuf)) { + outbuf = inbuf; + } else { + outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples); + if (!outbuf) + return AVERROR(ENOMEM); + av_frame_copy_props(outbuf, inbuf); + } + + p->phaser(p, inbuf->extended_data, outbuf->extended_data, + outbuf->nb_samples, av_frame_get_channels(outbuf)); + + if (inbuf != outbuf) + av_frame_free(&inbuf); + + return ff_filter_frame(outlink, outbuf); +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + AudioPhaserContext *p = ctx->priv; + + av_freep(&p->delay_buffer); + av_freep(&p->modulation_buffer); +} + +static const AVFilterPad aphaser_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + }, + { NULL } +}; + +static const AVFilterPad aphaser_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_output, + }, + { NULL } +}; + +AVFilter ff_af_aphaser = { + .name = "aphaser", + .description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."), + .query_formats = query_formats, + .priv_size = sizeof(AudioPhaserContext), + .init = init, + .uninit = uninit, + .inputs = aphaser_inputs, + .outputs = aphaser_outputs, + .priv_class = &aphaser_class, +}; |