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b4f5201967
If one looks at the many query_formats callbacks in existence, one will immediately recognize that there is one type of default callback for video and a slightly different default callback for audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);" for video with a filter-specific pix_fmts list. For audio, it is the same with a filter-specific sample_fmts list together with ff_set_common_all_samplerates() and ff_set_common_all_channel_counts(). This commit allows to remove the boilerplate query_formats callbacks by replacing said callback with a union consisting the old callback and pointers for pixel and sample format arrays. For the not uncommon case in which these lists only contain a single entry (besides the sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also added to the union to store them directly in the AVFilter, thereby avoiding a relocation. The state of said union will be contained in a new, dedicated AVFilter field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t in order to create a hole for this new field; this is no problem, as the maximum of all the nb_inputs is four; for nb_outputs it is only two). The state's default value coincides with the earlier default of query_formats being unset, namely that the filter accepts all formats (and also sample rates and channel counts/layouts for audio) provided that these properties agree coincide for all inputs and outputs. By using different union members for audio and video filters the type-unsafety of using the same functions for audio and video lists will furthermore be more confined to formats.c than before. When the new fields are used, they will also avoid allocations: Currently something nearly equivalent to ff_default_query_formats() is called after every successful call to a query_formats callback; yet in the common case that the newly allocated AVFilterFormats are not used at all (namely if there are no free links) these newly allocated AVFilterFormats are freed again without ever being used. Filters no longer using the callback will not exhibit this any more. Reviewed-by: Paul B Mahol <onemda@gmail.com> Reviewed-by: Nicolas George <george@nsup.org> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
226 lines
8.0 KiB
C
226 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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return ff_set_common_all_samplerates(ctx);
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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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};
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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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};
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const 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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.priv_size = sizeof(HaasContext),
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.priv_class = &haas_class,
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.uninit = uninit,
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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(outputs),
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FILTER_QUERY_FUNC(query_formats),
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};
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