| /* |
| * Copyright (c) 2019 The FFmpeg Project |
| * |
| * 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/avassert.h" |
| #include "libavutil/channel_layout.h" |
| #include "libavutil/opt.h" |
| #include "libswresample/swresample.h" |
| #include "avfilter.h" |
| #include "audio.h" |
| #include "formats.h" |
| |
| enum ASoftClipTypes { |
| ASC_HARD = -1, |
| ASC_TANH, |
| ASC_ATAN, |
| ASC_CUBIC, |
| ASC_EXP, |
| ASC_ALG, |
| ASC_QUINTIC, |
| ASC_SIN, |
| ASC_ERF, |
| NB_TYPES, |
| }; |
| |
| typedef struct ASoftClipContext { |
| const AVClass *class; |
| |
| int type; |
| int oversample; |
| int64_t delay; |
| double threshold; |
| double output; |
| double param; |
| |
| SwrContext *up_ctx; |
| SwrContext *down_ctx; |
| |
| AVFrame *frame; |
| |
| void (*filter)(struct ASoftClipContext *s, void **dst, const void **src, |
| int nb_samples, int channels, int start, int end); |
| } ASoftClipContext; |
| |
| #define OFFSET(x) offsetof(ASoftClipContext, x) |
| #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM |
| #define F AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
| |
| static const AVOption asoftclip_options[] = { |
| { "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, -1, NB_TYPES-1, A, "types" }, |
| { "hard", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_HARD}, 0, 0, A, "types" }, |
| { "tanh", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_TANH}, 0, 0, A, "types" }, |
| { "atan", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ATAN}, 0, 0, A, "types" }, |
| { "cubic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_CUBIC}, 0, 0, A, "types" }, |
| { "exp", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_EXP}, 0, 0, A, "types" }, |
| { "alg", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ALG}, 0, 0, A, "types" }, |
| { "quintic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_QUINTIC},0, 0, A, "types" }, |
| { "sin", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_SIN}, 0, 0, A, "types" }, |
| { "erf", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ERF}, 0, 0, A, "types" }, |
| { "threshold", "set softclip threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 1, A }, |
| { "output", "set softclip output gain", OFFSET(output), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 16, A }, |
| { "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A }, |
| { "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, 32, F }, |
| { NULL } |
| }; |
| |
| AVFILTER_DEFINE_CLASS(asoftclip); |
| |
| static int query_formats(AVFilterContext *ctx) |
| { |
| AVFilterFormats *formats = NULL; |
| AVFilterChannelLayouts *layouts = NULL; |
| static const enum AVSampleFormat sample_fmts[] = { |
| AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, |
| AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP, |
| AV_SAMPLE_FMT_NONE |
| }; |
| int 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; |
| |
| 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_all_samplerates(); |
| return ff_set_common_samplerates(ctx, formats); |
| } |
| |
| static void filter_flt(ASoftClipContext *s, |
| void **dptr, const void **sptr, |
| int nb_samples, int channels, |
| int start, int end) |
| { |
| float threshold = s->threshold; |
| float gain = s->output * threshold; |
| float factor = 1.f / threshold; |
| float param = s->param; |
| |
| for (int c = start; c < end; c++) { |
| const float *src = sptr[c]; |
| float *dst = dptr[c]; |
| |
| switch (s->type) { |
| case ASC_HARD: |
| for (int n = 0; n < nb_samples; n++) { |
| dst[n] = av_clipf(src[n] * factor, -1.f, 1.f); |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_TANH: |
| for (int n = 0; n < nb_samples; n++) { |
| dst[n] = tanhf(src[n] * factor * param); |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_ATAN: |
| for (int n = 0; n < nb_samples; n++) { |
| dst[n] = 2.f / M_PI * atanf(src[n] * factor * param); |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_CUBIC: |
| for (int n = 0; n < nb_samples; n++) { |
| float sample = src[n] * factor; |
| |
| if (FFABS(sample) >= 1.5f) |
| dst[n] = FFSIGN(sample); |
| else |
| dst[n] = sample - 0.1481f * powf(sample, 3.f); |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_EXP: |
| for (int n = 0; n < nb_samples; n++) { |
| dst[n] = 2.f / (1.f + expf(-2.f * src[n] * factor)) - 1.; |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_ALG: |
| for (int n = 0; n < nb_samples; n++) { |
| float sample = src[n] * factor; |
| |
| dst[n] = sample / (sqrtf(param + sample * sample)); |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_QUINTIC: |
| for (int n = 0; n < nb_samples; n++) { |
| float sample = src[n] * factor; |
| |
| if (FFABS(sample) >= 1.25) |
| dst[n] = FFSIGN(sample); |
| else |
| dst[n] = sample - 0.08192f * powf(sample, 5.f); |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_SIN: |
| for (int n = 0; n < nb_samples; n++) { |
| float sample = src[n] * factor; |
| |
| if (FFABS(sample) >= M_PI_2) |
| dst[n] = FFSIGN(sample); |
| else |
| dst[n] = sinf(sample); |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_ERF: |
| for (int n = 0; n < nb_samples; n++) { |
| dst[n] = erff(src[n] * factor); |
| dst[n] *= gain; |
| } |
| break; |
| default: |
| av_assert0(0); |
| } |
| } |
| } |
| |
| static void filter_dbl(ASoftClipContext *s, |
| void **dptr, const void **sptr, |
| int nb_samples, int channels, |
| int start, int end) |
| { |
| double threshold = s->threshold; |
| double gain = s->output * threshold; |
| double factor = 1. / threshold; |
| double param = s->param; |
| |
| for (int c = start; c < end; c++) { |
| const double *src = sptr[c]; |
| double *dst = dptr[c]; |
| |
| switch (s->type) { |
| case ASC_HARD: |
| for (int n = 0; n < nb_samples; n++) { |
| dst[n] = av_clipd(src[n] * factor, -1., 1.); |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_TANH: |
| for (int n = 0; n < nb_samples; n++) { |
| dst[n] = tanh(src[n] * factor * param); |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_ATAN: |
| for (int n = 0; n < nb_samples; n++) { |
| dst[n] = 2. / M_PI * atan(src[n] * factor * param); |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_CUBIC: |
| for (int n = 0; n < nb_samples; n++) { |
| double sample = src[n] * factor; |
| |
| if (FFABS(sample) >= 1.5) |
| dst[n] = FFSIGN(sample); |
| else |
| dst[n] = sample - 0.1481 * pow(sample, 3.); |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_EXP: |
| for (int n = 0; n < nb_samples; n++) { |
| dst[n] = 2. / (1. + exp(-2. * src[n] * factor)) - 1.; |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_ALG: |
| for (int n = 0; n < nb_samples; n++) { |
| double sample = src[n] * factor; |
| |
| dst[n] = sample / (sqrt(param + sample * sample)); |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_QUINTIC: |
| for (int n = 0; n < nb_samples; n++) { |
| double sample = src[n] * factor; |
| |
| if (FFABS(sample) >= 1.25) |
| dst[n] = FFSIGN(sample); |
| else |
| dst[n] = sample - 0.08192 * pow(sample, 5.); |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_SIN: |
| for (int n = 0; n < nb_samples; n++) { |
| double sample = src[n] * factor; |
| |
| if (FFABS(sample) >= M_PI_2) |
| dst[n] = FFSIGN(sample); |
| else |
| dst[n] = sin(sample); |
| dst[n] *= gain; |
| } |
| break; |
| case ASC_ERF: |
| for (int n = 0; n < nb_samples; n++) { |
| dst[n] = erf(src[n] * factor); |
| dst[n] *= gain; |
| } |
| break; |
| default: |
| av_assert0(0); |
| } |
| } |
| } |
| |
| static int config_input(AVFilterLink *inlink) |
| { |
| AVFilterContext *ctx = inlink->dst; |
| ASoftClipContext *s = ctx->priv; |
| int ret; |
| |
| switch (inlink->format) { |
| case AV_SAMPLE_FMT_FLT: |
| case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break; |
| case AV_SAMPLE_FMT_DBL: |
| case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; break; |
| default: av_assert0(0); |
| } |
| |
| if (s->oversample <= 1) |
| return 0; |
| |
| s->up_ctx = swr_alloc(); |
| s->down_ctx = swr_alloc(); |
| if (!s->up_ctx || !s->down_ctx) |
| return AVERROR(ENOMEM); |
| |
| av_opt_set_int(s->up_ctx, "in_channel_layout", inlink->channel_layout, 0); |
| av_opt_set_int(s->up_ctx, "in_sample_rate", inlink->sample_rate, 0); |
| av_opt_set_sample_fmt(s->up_ctx, "in_sample_fmt", inlink->format, 0); |
| |
| av_opt_set_int(s->up_ctx, "out_channel_layout", inlink->channel_layout, 0); |
| av_opt_set_int(s->up_ctx, "out_sample_rate", inlink->sample_rate * s->oversample, 0); |
| av_opt_set_sample_fmt(s->up_ctx, "out_sample_fmt", inlink->format, 0); |
| |
| av_opt_set_int(s->down_ctx, "in_channel_layout", inlink->channel_layout, 0); |
| av_opt_set_int(s->down_ctx, "in_sample_rate", inlink->sample_rate * s->oversample, 0); |
| av_opt_set_sample_fmt(s->down_ctx, "in_sample_fmt", inlink->format, 0); |
| |
| av_opt_set_int(s->down_ctx, "out_channel_layout", inlink->channel_layout, 0); |
| av_opt_set_int(s->down_ctx, "out_sample_rate", inlink->sample_rate, 0); |
| av_opt_set_sample_fmt(s->down_ctx, "out_sample_fmt", inlink->format, 0); |
| |
| ret = swr_init(s->up_ctx); |
| if (ret < 0) |
| return ret; |
| |
| ret = swr_init(s->down_ctx); |
| if (ret < 0) |
| return ret; |
| |
| return 0; |
| } |
| |
| typedef struct ThreadData { |
| AVFrame *in, *out; |
| int nb_samples; |
| int channels; |
| } ThreadData; |
| |
| static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) |
| { |
| ASoftClipContext *s = ctx->priv; |
| ThreadData *td = arg; |
| AVFrame *out = td->out; |
| AVFrame *in = td->in; |
| const int channels = td->channels; |
| const int nb_samples = td->nb_samples; |
| const int start = (channels * jobnr) / nb_jobs; |
| const int end = (channels * (jobnr+1)) / nb_jobs; |
| |
| s->filter(s, (void **)out->extended_data, (const void **)in->extended_data, |
| nb_samples, channels, start, end); |
| |
| return 0; |
| } |
| |
| static int filter_frame(AVFilterLink *inlink, AVFrame *in) |
| { |
| AVFilterContext *ctx = inlink->dst; |
| ASoftClipContext *s = ctx->priv; |
| AVFilterLink *outlink = ctx->outputs[0]; |
| int ret, nb_samples, channels; |
| ThreadData td; |
| AVFrame *out; |
| |
| if (av_frame_is_writable(in)) { |
| out = in; |
| } else { |
| out = ff_get_audio_buffer(outlink, in->nb_samples); |
| if (!out) { |
| av_frame_free(&in); |
| return AVERROR(ENOMEM); |
| } |
| av_frame_copy_props(out, in); |
| } |
| |
| if (av_sample_fmt_is_planar(in->format)) { |
| nb_samples = in->nb_samples; |
| channels = in->channels; |
| } else { |
| nb_samples = in->channels * in->nb_samples; |
| channels = 1; |
| } |
| |
| if (s->oversample > 1) { |
| s->frame = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample); |
| if (!s->frame) { |
| ret = AVERROR(ENOMEM); |
| goto fail; |
| } |
| |
| ret = swr_convert(s->up_ctx, (uint8_t**)s->frame->extended_data, in->nb_samples * s->oversample, |
| (const uint8_t **)in->extended_data, in->nb_samples); |
| if (ret < 0) |
| goto fail; |
| |
| td.in = s->frame; |
| td.out = s->frame; |
| td.nb_samples = av_sample_fmt_is_planar(in->format) ? ret : ret * in->channels; |
| td.channels = channels; |
| ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(channels, |
| ff_filter_get_nb_threads(ctx))); |
| |
| ret = swr_convert(s->down_ctx, (uint8_t**)out->extended_data, out->nb_samples, |
| (const uint8_t **)s->frame->extended_data, ret); |
| if (ret < 0) |
| goto fail; |
| |
| if (out->pts) |
| out->pts -= s->delay; |
| s->delay += in->nb_samples - ret; |
| out->nb_samples = ret; |
| |
| av_frame_free(&s->frame); |
| } else { |
| td.in = in; |
| td.out = out; |
| td.nb_samples = nb_samples; |
| td.channels = channels; |
| ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(channels, |
| ff_filter_get_nb_threads(ctx))); |
| } |
| |
| if (out != in) |
| av_frame_free(&in); |
| |
| return ff_filter_frame(outlink, out); |
| fail: |
| if (out != in) |
| av_frame_free(&out); |
| av_frame_free(&in); |
| av_frame_free(&s->frame); |
| |
| return ret; |
| } |
| |
| static av_cold void uninit(AVFilterContext *ctx) |
| { |
| ASoftClipContext *s = ctx->priv; |
| |
| swr_free(&s->up_ctx); |
| swr_free(&s->down_ctx); |
| } |
| |
| static const AVFilterPad inputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .filter_frame = filter_frame, |
| .config_props = config_input, |
| }, |
| { NULL } |
| }; |
| |
| static const AVFilterPad outputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| }, |
| { NULL } |
| }; |
| |
| AVFilter ff_af_asoftclip = { |
| .name = "asoftclip", |
| .description = NULL_IF_CONFIG_SMALL("Audio Soft Clipper."), |
| .query_formats = query_formats, |
| .priv_size = sizeof(ASoftClipContext), |
| .priv_class = &asoftclip_class, |
| .inputs = inputs, |
| .outputs = outputs, |
| .uninit = uninit, |
| .process_command = ff_filter_process_command, |
| .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC | |
| AVFILTER_FLAG_SLICE_THREADS, |
| }; |
