mirror of https://github.com/FFmpeg/FFmpeg.git
Paul B Mahol
7 years ago
parent
e59da0f7ff
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6 changed files with 296 additions and 1 deletions
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/*
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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(inlink, 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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