mirror of
https://github.com/FFmpeg/FFmpeg.git
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229 lines
8.0 KiB
C
229 lines
8.0 KiB
C
/*
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* Copyright (c) 2001-2010 Vladimir Sadovnikov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "formats.h"
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#define MAX_HAAS_DELAY 40
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typedef struct HaasContext {
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const AVClass *class;
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int par_m_source;
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double par_delay0;
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double par_delay1;
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int par_phase0;
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int par_phase1;
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int par_middle_phase;
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double par_side_gain;
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double par_gain0;
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double par_gain1;
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double par_balance0;
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double par_balance1;
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double level_in;
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double level_out;
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double *buffer;
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size_t buffer_size;
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uint32_t write_ptr;
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uint32_t delay[2];
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double balance_l[2];
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double balance_r[2];
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double phase0;
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double phase1;
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} HaasContext;
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#define OFFSET(x) offsetof(HaasContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption haas_options[] = {
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{ "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
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{ "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
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{ "side_gain", "set side gain", OFFSET(par_side_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
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{ "middle_source", "set middle source", OFFSET(par_m_source), AV_OPT_TYPE_INT, {.i64=2}, 0, 3, A, "source" },
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{ "left", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "source" },
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{ "right", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "source" },
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{ "mid", "L+R", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "source" },
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{ "side", "L-R", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, "source" },
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{ "middle_phase", "set middle phase", OFFSET(par_middle_phase), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
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{ "left_delay", "set left delay", OFFSET(par_delay0), AV_OPT_TYPE_DOUBLE, {.dbl=2.05}, 0, MAX_HAAS_DELAY, A },
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{ "left_balance", "set left balance", OFFSET(par_balance0), AV_OPT_TYPE_DOUBLE, {.dbl=-1.0}, -1, 1, A },
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{ "left_gain", "set left gain", OFFSET(par_gain0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
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{ "left_phase", "set left phase", OFFSET(par_phase0), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
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{ "right_delay", "set right delay", OFFSET(par_delay1), AV_OPT_TYPE_DOUBLE, {.dbl=2.12}, 0, MAX_HAAS_DELAY, A },
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{ "right_balance", "set right balance", OFFSET(par_balance1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, -1, 1, A },
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{ "right_gain", "set right gain", OFFSET(par_gain1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
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{ "right_phase", "set right phase", OFFSET(par_phase1), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(haas);
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats = NULL;
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AVFilterChannelLayouts *layout = NULL;
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int ret;
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if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
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(ret = ff_set_common_formats (ctx , formats )) < 0 ||
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(ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO)) < 0 ||
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(ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
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return ret;
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formats = ff_all_samplerates();
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return ff_set_common_samplerates(ctx, formats);
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}
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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HaasContext *s = ctx->priv;
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size_t min_buf_size = (size_t)(inlink->sample_rate * MAX_HAAS_DELAY * 0.001);
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size_t new_buf_size = 1;
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while (new_buf_size < min_buf_size)
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new_buf_size <<= 1;
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av_freep(&s->buffer);
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s->buffer = av_calloc(new_buf_size, sizeof(*s->buffer));
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if (!s->buffer)
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return AVERROR(ENOMEM);
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s->buffer_size = new_buf_size;
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s->write_ptr = 0;
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s->delay[0] = (uint32_t)(s->par_delay0 * 0.001 * inlink->sample_rate);
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s->delay[1] = (uint32_t)(s->par_delay1 * 0.001 * inlink->sample_rate);
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s->phase0 = s->par_phase0 ? 1.0 : -1.0;
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s->phase1 = s->par_phase1 ? 1.0 : -1.0;
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s->balance_l[0] = (s->par_balance0 + 1) / 2 * s->par_gain0 * s->phase0;
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s->balance_r[0] = (1.0 - (s->par_balance0 + 1) / 2) * (s->par_gain0) * s->phase0;
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s->balance_l[1] = (s->par_balance1 + 1) / 2 * s->par_gain1 * s->phase1;
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s->balance_r[1] = (1.0 - (s->par_balance1 + 1) / 2) * (s->par_gain1) * s->phase1;
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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HaasContext *s = ctx->priv;
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const double *src = (const double *)in->data[0];
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const double level_in = s->level_in;
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const double level_out = s->level_out;
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const uint32_t mask = s->buffer_size - 1;
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double *buffer = s->buffer;
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AVFrame *out;
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double *dst;
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int n;
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if (av_frame_is_writable(in)) {
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out = in;
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} else {
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out = ff_get_audio_buffer(outlink, in->nb_samples);
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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}
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dst = (double *)out->data[0];
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for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
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double mid, side[2], side_l, side_r;
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uint32_t s0_ptr, s1_ptr;
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switch (s->par_m_source) {
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case 0: mid = src[0]; break;
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case 1: mid = src[1]; break;
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case 2: mid = (src[0] + src[1]) * 0.5; break;
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case 3: mid = (src[0] - src[1]) * 0.5; break;
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}
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mid *= level_in;
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buffer[s->write_ptr] = mid;
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s0_ptr = (s->write_ptr + s->buffer_size - s->delay[0]) & mask;
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s1_ptr = (s->write_ptr + s->buffer_size - s->delay[1]) & mask;
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if (s->par_middle_phase)
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mid = -mid;
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side[0] = buffer[s0_ptr] * s->par_side_gain;
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side[1] = buffer[s1_ptr] * s->par_side_gain;
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side_l = side[0] * s->balance_l[0] - side[1] * s->balance_l[1];
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side_r = side[1] * s->balance_r[1] - side[0] * s->balance_r[0];
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dst[0] = (mid + side_l) * level_out;
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dst[1] = (mid + side_r) * level_out;
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s->write_ptr = (s->write_ptr + 1) & mask;
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}
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if (out != in)
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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HaasContext *s = ctx->priv;
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av_freep(&s->buffer);
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s->buffer_size = 0;
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}
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static const AVFilterPad inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.config_props = config_input,
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},
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{ NULL }
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};
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static const AVFilterPad outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL }
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};
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AVFilter ff_af_haas = {
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.name = "haas",
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.description = NULL_IF_CONFIG_SMALL("Apply Haas Stereo Enhancer."),
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.query_formats = query_formats,
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.priv_size = sizeof(HaasContext),
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.priv_class = &haas_class,
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.uninit = uninit,
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.inputs = inputs,
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.outputs = outputs,
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};
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