| /* |
| * 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" |
| #include "generate_wave_table.h" |
| |
| typedef struct AudioPhaserContext { |
| const AVClass *class; |
| double in_gain, out_gain; |
| double delay; |
| double decay; |
| double speed; |
| |
| int 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 *s, |
| 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 *s = ctx->priv; |
| |
| if (s->in_gain > (1 - s->decay * s->decay)) |
| av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n"); |
| if (s->in_gain / (1 - s->decay) > 1 / s->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 |
| }; |
| int ret; |
| |
| layouts = ff_all_channel_counts(); |
| if (!layouts) |
| return AVERROR(ENOMEM); |
| ret = ff_set_common_channel_layouts(ctx, layouts); |
| if (ret < 0) |
| return ret; |
| |
| formats = ff_make_format_list(sample_fmts); |
| if (!formats) |
| return AVERROR(ENOMEM); |
| ret = ff_set_common_formats(ctx, formats); |
| if (ret < 0) |
| return ret; |
| |
| formats = ff_all_samplerates(); |
| if (!formats) |
| return AVERROR(ENOMEM); |
| return ff_set_common_samplerates(ctx, formats); |
| } |
| |
| #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a)) |
| |
| #define PHASER_PLANAR(name, type) \ |
| static void phaser_## name ##p(AudioPhaserContext *s, \ |
| uint8_t * const *ssrc, uint8_t **ddst, \ |
| int nb_samples, int channels) \ |
| { \ |
| int i, c, delay_pos, modulation_pos; \ |
| \ |
| av_assert0(channels > 0); \ |
| for (c = 0; c < channels; c++) { \ |
| type *src = (type *)ssrc[c]; \ |
| type *dst = (type *)ddst[c]; \ |
| double *buffer = s->delay_buffer + \ |
| c * s->delay_buffer_length; \ |
| \ |
| delay_pos = s->delay_pos; \ |
| modulation_pos = s->modulation_pos; \ |
| \ |
| for (i = 0; i < nb_samples; i++, src++, dst++) { \ |
| double v = *src * s->in_gain + buffer[ \ |
| MOD(delay_pos + s->modulation_buffer[ \ |
| modulation_pos], \ |
| s->delay_buffer_length)] * s->decay; \ |
| \ |
| modulation_pos = MOD(modulation_pos + 1, \ |
| s->modulation_buffer_length); \ |
| delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \ |
| buffer[delay_pos] = v; \ |
| \ |
| *dst = v * s->out_gain; \ |
| } \ |
| } \ |
| \ |
| s->delay_pos = delay_pos; \ |
| s->modulation_pos = modulation_pos; \ |
| } |
| |
| #define PHASER(name, type) \ |
| static void phaser_## name (AudioPhaserContext *s, \ |
| uint8_t * const *ssrc, uint8_t **ddst, \ |
| int nb_samples, int channels) \ |
| { \ |
| int i, c, delay_pos, modulation_pos; \ |
| type *src = (type *)ssrc[0]; \ |
| type *dst = (type *)ddst[0]; \ |
| double *buffer = s->delay_buffer; \ |
| \ |
| delay_pos = s->delay_pos; \ |
| modulation_pos = s->modulation_pos; \ |
| \ |
| for (i = 0; i < nb_samples; i++) { \ |
| int pos = MOD(delay_pos + s->modulation_buffer[modulation_pos], \ |
| s->delay_buffer_length) * channels; \ |
| int npos; \ |
| \ |
| delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \ |
| npos = delay_pos * channels; \ |
| for (c = 0; c < channels; c++, src++, dst++) { \ |
| double v = *src * s->in_gain + buffer[pos + c] * s->decay; \ |
| \ |
| buffer[npos + c] = v; \ |
| \ |
| *dst = v * s->out_gain; \ |
| } \ |
| \ |
| modulation_pos = MOD(modulation_pos + 1, \ |
| s->modulation_buffer_length); \ |
| } \ |
| \ |
| s->delay_pos = delay_pos; \ |
| s->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 *s = outlink->src->priv; |
| AVFilterLink *inlink = outlink->src->inputs[0]; |
| |
| s->delay_buffer_length = s->delay * 0.001 * inlink->sample_rate + 0.5; |
| if (s->delay_buffer_length <= 0) { |
| av_log(outlink->src, AV_LOG_ERROR, "delay is too small\n"); |
| return AVERROR(EINVAL); |
| } |
| s->delay_buffer = av_calloc(s->delay_buffer_length, sizeof(*s->delay_buffer) * inlink->channels); |
| s->modulation_buffer_length = inlink->sample_rate / s->speed + 0.5; |
| s->modulation_buffer = av_malloc_array(s->modulation_buffer_length, sizeof(*s->modulation_buffer)); |
| |
| if (!s->modulation_buffer || !s->delay_buffer) |
| return AVERROR(ENOMEM); |
| |
| ff_generate_wave_table(s->type, AV_SAMPLE_FMT_S32, |
| s->modulation_buffer, s->modulation_buffer_length, |
| 1., s->delay_buffer_length, M_PI / 2.0); |
| |
| s->delay_pos = s->modulation_pos = 0; |
| |
| switch (inlink->format) { |
| case AV_SAMPLE_FMT_DBL: s->phaser = phaser_dbl; break; |
| case AV_SAMPLE_FMT_DBLP: s->phaser = phaser_dblp; break; |
| case AV_SAMPLE_FMT_FLT: s->phaser = phaser_flt; break; |
| case AV_SAMPLE_FMT_FLTP: s->phaser = phaser_fltp; break; |
| case AV_SAMPLE_FMT_S16: s->phaser = phaser_s16; break; |
| case AV_SAMPLE_FMT_S16P: s->phaser = phaser_s16p; break; |
| case AV_SAMPLE_FMT_S32: s->phaser = phaser_s32; break; |
| case AV_SAMPLE_FMT_S32P: s->phaser = phaser_s32p; break; |
| default: av_assert0(0); |
| } |
| |
| return 0; |
| } |
| |
| static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf) |
| { |
| AudioPhaserContext *s = 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(outlink, inbuf->nb_samples); |
| if (!outbuf) { |
| av_frame_free(&inbuf); |
| return AVERROR(ENOMEM); |
| } |
| av_frame_copy_props(outbuf, inbuf); |
| } |
| |
| s->phaser(s, inbuf->extended_data, outbuf->extended_data, |
| outbuf->nb_samples, outbuf->channels); |
| |
| if (inbuf != outbuf) |
| av_frame_free(&inbuf); |
| |
| return ff_filter_frame(outlink, outbuf); |
| } |
| |
| static av_cold void uninit(AVFilterContext *ctx) |
| { |
| AudioPhaserContext *s = ctx->priv; |
| |
| av_freep(&s->delay_buffer); |
| av_freep(&s->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, |
| }; |
