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Diffstat (limited to 'libavfilter/af_flanger.c')
| -rw-r--r-- | libavfilter/af_flanger.c | 246 |
1 files changed, 246 insertions, 0 deletions
diff --git a/libavfilter/af_flanger.c b/libavfilter/af_flanger.c new file mode 100644 index 0000000000..b7497a12ed --- /dev/null +++ b/libavfilter/af_flanger.c @@ -0,0 +1,246 @@ +/* + * Copyright (c) 2006 Rob Sykes <robs@users.sourceforge.net> + * + * 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 + */ + +#include "libavutil/avstring.h" +#include "libavutil/opt.h" +#include "libavutil/samplefmt.h" +#include "avfilter.h" +#include "audio.h" +#include "internal.h" +#include "generate_wave_table.h" + +#define INTERPOLATION_LINEAR 0 +#define INTERPOLATION_QUADRATIC 1 + +typedef struct FlangerContext { + const AVClass *class; + double delay_min; + double delay_depth; + double feedback_gain; + double delay_gain; + double speed; + int wave_shape; + double channel_phase; + int interpolation; + double in_gain; + int max_samples; + uint8_t **delay_buffer; + int delay_buf_pos; + double *delay_last; + float *lfo; + int lfo_length; + int lfo_pos; +} FlangerContext; + +#define OFFSET(x) offsetof(FlangerContext, x) +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption flanger_options[] = { + { "delay", "base delay in milliseconds", OFFSET(delay_min), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, A }, + { "depth", "added swept delay in milliseconds", OFFSET(delay_depth), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, A }, + { "regen", "percentage regeneration (delayed signal feedback)", OFFSET(feedback_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95, A }, + { "width", "percentage of delayed signal mixed with original", OFFSET(delay_gain), AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100, A }, + { "speed", "sweeps per second (Hz)", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.1, 10, A }, + { "shape", "swept wave shape", OFFSET(wave_shape), AV_OPT_TYPE_INT, {.i64=WAVE_SIN}, WAVE_SIN, WAVE_NB-1, A, "type" }, + { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" }, + { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" }, + { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" }, + { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" }, + { "phase", "swept wave percentage phase-shift for multi-channel", OFFSET(channel_phase), AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100, A }, + { "interp", "delay-line interpolation", OFFSET(interpolation), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "itype" }, + { "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_LINEAR}, 0, 0, A, "itype" }, + { "quadratic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, "itype" }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(flanger); + +static int init(AVFilterContext *ctx) +{ + FlangerContext *s = ctx->priv; + + s->feedback_gain /= 100; + s->delay_gain /= 100; + s->channel_phase /= 100; + s->delay_min /= 1000; + s->delay_depth /= 1000; + s->in_gain = 1 / (1 + s->delay_gain); + s->delay_gain /= 1 + s->delay_gain; + s->delay_gain *= 1 - fabs(s->feedback_gain); + + return 0; +} + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterChannelLayouts *layouts; + AVFilterFormats *formats; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_DBLP, 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); +} + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + FlangerContext *s = ctx->priv; + + s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5; + s->lfo_length = inlink->sample_rate / s->speed; + s->delay_last = av_calloc(inlink->channels, sizeof(*s->delay_last)); + s->lfo = av_calloc(s->lfo_length, sizeof(*s->lfo)); + if (!s->lfo || !s->delay_last) + return AVERROR(ENOMEM); + + ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length, + rint(s->delay_min * inlink->sample_rate), + s->max_samples - 2., 3 * M_PI_2); + + return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL, + inlink->channels, s->max_samples, + inlink->format, 0); +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *frame) +{ + AVFilterContext *ctx = inlink->dst; + FlangerContext *s = ctx->priv; + AVFrame *out_frame; + int chan, i; + + if (av_frame_is_writable(frame)) { + out_frame = frame; + } else { + out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples); + if (!out_frame) { + av_frame_free(&frame); + return AVERROR(ENOMEM); + } + av_frame_copy_props(out_frame, frame); + } + + for (i = 0; i < frame->nb_samples; i++) { + + s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples; + + for (chan = 0; chan < inlink->channels; chan++) { + double *src = (double *)frame->extended_data[chan]; + double *dst = (double *)out_frame->extended_data[chan]; + double delayed_0, delayed_1; + double delayed; + double in, out; + int channel_phase = chan * s->lfo_length * s->channel_phase + .5; + double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length]; + int int_delay = (int)delay; + double frac_delay = modf(delay, &delay); + double *delay_buffer = (double *)s->delay_buffer[chan]; + + in = src[i]; + delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] * + s->feedback_gain; + delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples]; + delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples]; + + if (s->interpolation == INTERPOLATION_LINEAR) { + delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay; + } else { + double a, b; + double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples]; + delayed_2 -= delayed_0; + delayed_1 -= delayed_0; + a = delayed_2 * .5 - delayed_1; + b = delayed_1 * 2 - delayed_2 *.5; + delayed = delayed_0 + (a * frac_delay + b) * frac_delay; + } + + s->delay_last[chan] = delayed; + out = in * s->in_gain + delayed * s->delay_gain; + dst[i] = out; + } + s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length; + } + + if (frame != out_frame) + av_frame_free(&frame); + + return ff_filter_frame(ctx->outputs[0], out_frame); +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + FlangerContext *s = ctx->priv; + + av_freep(&s->lfo); + av_freep(&s->delay_last); + + if (s->delay_buffer) + av_freep(&s->delay_buffer[0]); + av_freep(&s->delay_buffer); +} + +static const AVFilterPad flanger_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_input, + .filter_frame = filter_frame, + }, + { NULL } +}; + +static const AVFilterPad flanger_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +AVFilter ff_af_flanger = { + .name = "flanger", + .description = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."), + .query_formats = query_formats, + .priv_size = sizeof(FlangerContext), + .priv_class = &flanger_class, + .init = init, + .uninit = uninit, + .inputs = flanger_inputs, + .outputs = flanger_outputs, +}; |