| Line | Branch | Exec | Source |
|---|---|---|---|
| 1 | /* | ||
| 2 | * Copyright (c) 2001-2010 Vladimir Sadovnikov | ||
| 3 | * | ||
| 4 | * This file is part of FFmpeg. | ||
| 5 | * | ||
| 6 | * FFmpeg is free software; you can redistribute it and/or | ||
| 7 | * modify it under the terms of the GNU Lesser General Public | ||
| 8 | * License as published by the Free Software Foundation; either | ||
| 9 | * version 2.1 of the License, or (at your option) any later version. | ||
| 10 | * | ||
| 11 | * FFmpeg is distributed in the hope that it will be useful, | ||
| 12 | * but WITHOUT ANY WARRANTY; without even the implied warranty of | ||
| 13 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | ||
| 14 | * Lesser General Public License for more details. | ||
| 15 | * | ||
| 16 | * You should have received a copy of the GNU Lesser General Public | ||
| 17 | * License along with FFmpeg; if not, write to the Free Software | ||
| 18 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | ||
| 19 | */ | ||
| 20 | |||
| 21 | #include "libavutil/channel_layout.h" | ||
| 22 | #include "libavutil/mem.h" | ||
| 23 | #include "libavutil/opt.h" | ||
| 24 | #include "avfilter.h" | ||
| 25 | #include "audio.h" | ||
| 26 | #include "filters.h" | ||
| 27 | #include "formats.h" | ||
| 28 | |||
| 29 | #define MAX_HAAS_DELAY 40 | ||
| 30 | |||
| 31 | typedef struct HaasContext { | ||
| 32 | const AVClass *class; | ||
| 33 | |||
| 34 | int par_m_source; | ||
| 35 | double par_delay0; | ||
| 36 | double par_delay1; | ||
| 37 | int par_phase0; | ||
| 38 | int par_phase1; | ||
| 39 | int par_middle_phase; | ||
| 40 | double par_side_gain; | ||
| 41 | double par_gain0; | ||
| 42 | double par_gain1; | ||
| 43 | double par_balance0; | ||
| 44 | double par_balance1; | ||
| 45 | double level_in; | ||
| 46 | double level_out; | ||
| 47 | |||
| 48 | double *buffer; | ||
| 49 | size_t buffer_size; | ||
| 50 | uint32_t write_ptr; | ||
| 51 | uint32_t delay[2]; | ||
| 52 | double balance_l[2]; | ||
| 53 | double balance_r[2]; | ||
| 54 | double phase0; | ||
| 55 | double phase1; | ||
| 56 | } HaasContext; | ||
| 57 | |||
| 58 | #define OFFSET(x) offsetof(HaasContext, x) | ||
| 59 | #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM | ||
| 60 | |||
| 61 | static const AVOption haas_options[] = { | ||
| 62 | { "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, | ||
| 63 | { "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, | ||
| 64 | { "side_gain", "set side gain", OFFSET(par_side_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, | ||
| 65 | { "middle_source", "set middle source", OFFSET(par_m_source), AV_OPT_TYPE_INT, {.i64=2}, 0, 3, A, .unit = "source" }, | ||
| 66 | { "left", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "source" }, | ||
| 67 | { "right", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "source" }, | ||
| 68 | { "mid", "L+R", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, .unit = "source" }, | ||
| 69 | { "side", "L-R", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, .unit = "source" }, | ||
| 70 | { "middle_phase", "set middle phase", OFFSET(par_middle_phase), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A }, | ||
| 71 | { "left_delay", "set left delay", OFFSET(par_delay0), AV_OPT_TYPE_DOUBLE, {.dbl=2.05}, 0, MAX_HAAS_DELAY, A }, | ||
| 72 | { "left_balance", "set left balance", OFFSET(par_balance0), AV_OPT_TYPE_DOUBLE, {.dbl=-1.0}, -1, 1, A }, | ||
| 73 | { "left_gain", "set left gain", OFFSET(par_gain0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, | ||
| 74 | { "left_phase", "set left phase", OFFSET(par_phase0), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A }, | ||
| 75 | { "right_delay", "set right delay", OFFSET(par_delay1), AV_OPT_TYPE_DOUBLE, {.dbl=2.12}, 0, MAX_HAAS_DELAY, A }, | ||
| 76 | { "right_balance", "set right balance", OFFSET(par_balance1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, -1, 1, A }, | ||
| 77 | { "right_gain", "set right gain", OFFSET(par_gain1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, | ||
| 78 | { "right_phase", "set right phase", OFFSET(par_phase1), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A }, | ||
| 79 | { NULL } | ||
| 80 | }; | ||
| 81 | |||
| 82 | AVFILTER_DEFINE_CLASS(haas); | ||
| 83 | |||
| 84 | ✗ | static int query_formats(const AVFilterContext *ctx, | |
| 85 | AVFilterFormatsConfig **cfg_in, | ||
| 86 | AVFilterFormatsConfig **cfg_out) | ||
| 87 | { | ||
| 88 | static const enum AVSampleFormat formats[] = { | ||
| 89 | AV_SAMPLE_FMT_DBL, | ||
| 90 | AV_SAMPLE_FMT_NONE, | ||
| 91 | }; | ||
| 92 | static const AVChannelLayout layouts[] = { | ||
| 93 | AV_CHANNEL_LAYOUT_STEREO, | ||
| 94 | { .nb_channels = 0 }, | ||
| 95 | }; | ||
| 96 | int ret; | ||
| 97 | |||
| 98 | ✗ | ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, formats); | |
| 99 | ✗ | if (ret < 0) | |
| 100 | ✗ | return ret; | |
| 101 | |||
| 102 | ✗ | ret = ff_set_common_channel_layouts_from_list2(ctx, cfg_in, cfg_out, layouts); | |
| 103 | ✗ | if (ret < 0) | |
| 104 | ✗ | return ret; | |
| 105 | |||
| 106 | ✗ | return 0; | |
| 107 | } | ||
| 108 | |||
| 109 | ✗ | static int config_input(AVFilterLink *inlink) | |
| 110 | { | ||
| 111 | ✗ | AVFilterContext *ctx = inlink->dst; | |
| 112 | ✗ | HaasContext *s = ctx->priv; | |
| 113 | ✗ | size_t min_buf_size = (size_t)(inlink->sample_rate * MAX_HAAS_DELAY * 0.001); | |
| 114 | ✗ | size_t new_buf_size = 1; | |
| 115 | |||
| 116 | ✗ | while (new_buf_size < min_buf_size) | |
| 117 | ✗ | new_buf_size <<= 1; | |
| 118 | |||
| 119 | ✗ | av_freep(&s->buffer); | |
| 120 | ✗ | s->buffer = av_calloc(new_buf_size, sizeof(*s->buffer)); | |
| 121 | ✗ | if (!s->buffer) | |
| 122 | ✗ | return AVERROR(ENOMEM); | |
| 123 | |||
| 124 | ✗ | s->buffer_size = new_buf_size; | |
| 125 | ✗ | s->write_ptr = 0; | |
| 126 | |||
| 127 | ✗ | s->delay[0] = (uint32_t)(s->par_delay0 * 0.001 * inlink->sample_rate); | |
| 128 | ✗ | s->delay[1] = (uint32_t)(s->par_delay1 * 0.001 * inlink->sample_rate); | |
| 129 | |||
| 130 | ✗ | s->phase0 = s->par_phase0 ? 1.0 : -1.0; | |
| 131 | ✗ | s->phase1 = s->par_phase1 ? 1.0 : -1.0; | |
| 132 | |||
| 133 | ✗ | s->balance_l[0] = (s->par_balance0 + 1) / 2 * s->par_gain0 * s->phase0; | |
| 134 | ✗ | s->balance_r[0] = (1.0 - (s->par_balance0 + 1) / 2) * (s->par_gain0) * s->phase0; | |
| 135 | ✗ | s->balance_l[1] = (s->par_balance1 + 1) / 2 * s->par_gain1 * s->phase1; | |
| 136 | ✗ | s->balance_r[1] = (1.0 - (s->par_balance1 + 1) / 2) * (s->par_gain1) * s->phase1; | |
| 137 | |||
| 138 | ✗ | return 0; | |
| 139 | } | ||
| 140 | |||
| 141 | ✗ | static int filter_frame(AVFilterLink *inlink, AVFrame *in) | |
| 142 | { | ||
| 143 | ✗ | AVFilterContext *ctx = inlink->dst; | |
| 144 | ✗ | AVFilterLink *outlink = ctx->outputs[0]; | |
| 145 | ✗ | HaasContext *s = ctx->priv; | |
| 146 | ✗ | const double *src = (const double *)in->data[0]; | |
| 147 | ✗ | const double level_in = s->level_in; | |
| 148 | ✗ | const double level_out = s->level_out; | |
| 149 | ✗ | const uint32_t mask = s->buffer_size - 1; | |
| 150 | ✗ | double *buffer = s->buffer; | |
| 151 | AVFrame *out; | ||
| 152 | double *dst; | ||
| 153 | int n; | ||
| 154 | |||
| 155 | ✗ | if (av_frame_is_writable(in)) { | |
| 156 | ✗ | out = in; | |
| 157 | } else { | ||
| 158 | ✗ | out = ff_get_audio_buffer(outlink, in->nb_samples); | |
| 159 | ✗ | if (!out) { | |
| 160 | ✗ | av_frame_free(&in); | |
| 161 | ✗ | return AVERROR(ENOMEM); | |
| 162 | } | ||
| 163 | ✗ | av_frame_copy_props(out, in); | |
| 164 | } | ||
| 165 | ✗ | dst = (double *)out->data[0]; | |
| 166 | |||
| 167 | ✗ | for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) { | |
| 168 | double mid, side[2], side_l, side_r; | ||
| 169 | uint32_t s0_ptr, s1_ptr; | ||
| 170 | |||
| 171 | ✗ | switch (s->par_m_source) { | |
| 172 | ✗ | case 0: mid = src[0]; break; | |
| 173 | ✗ | case 1: mid = src[1]; break; | |
| 174 | ✗ | case 2: mid = (src[0] + src[1]) * 0.5; break; | |
| 175 | ✗ | case 3: mid = (src[0] - src[1]) * 0.5; break; | |
| 176 | } | ||
| 177 | |||
| 178 | ✗ | mid *= level_in; | |
| 179 | |||
| 180 | ✗ | buffer[s->write_ptr] = mid; | |
| 181 | |||
| 182 | ✗ | s0_ptr = (s->write_ptr + s->buffer_size - s->delay[0]) & mask; | |
| 183 | ✗ | s1_ptr = (s->write_ptr + s->buffer_size - s->delay[1]) & mask; | |
| 184 | |||
| 185 | ✗ | if (s->par_middle_phase) | |
| 186 | ✗ | mid = -mid; | |
| 187 | |||
| 188 | ✗ | side[0] = buffer[s0_ptr] * s->par_side_gain; | |
| 189 | ✗ | side[1] = buffer[s1_ptr] * s->par_side_gain; | |
| 190 | ✗ | side_l = side[0] * s->balance_l[0] - side[1] * s->balance_l[1]; | |
| 191 | ✗ | side_r = side[1] * s->balance_r[1] - side[0] * s->balance_r[0]; | |
| 192 | |||
| 193 | ✗ | dst[0] = (mid + side_l) * level_out; | |
| 194 | ✗ | dst[1] = (mid + side_r) * level_out; | |
| 195 | |||
| 196 | ✗ | s->write_ptr = (s->write_ptr + 1) & mask; | |
| 197 | } | ||
| 198 | |||
| 199 | ✗ | if (out != in) | |
| 200 | ✗ | av_frame_free(&in); | |
| 201 | ✗ | return ff_filter_frame(outlink, out); | |
| 202 | } | ||
| 203 | |||
| 204 | ✗ | static av_cold void uninit(AVFilterContext *ctx) | |
| 205 | { | ||
| 206 | ✗ | HaasContext *s = ctx->priv; | |
| 207 | |||
| 208 | ✗ | av_freep(&s->buffer); | |
| 209 | ✗ | s->buffer_size = 0; | |
| 210 | ✗ | } | |
| 211 | |||
| 212 | static const AVFilterPad inputs[] = { | ||
| 213 | { | ||
| 214 | .name = "default", | ||
| 215 | .type = AVMEDIA_TYPE_AUDIO, | ||
| 216 | .filter_frame = filter_frame, | ||
| 217 | .config_props = config_input, | ||
| 218 | }, | ||
| 219 | }; | ||
| 220 | |||
| 221 | const AVFilter ff_af_haas = { | ||
| 222 | .name = "haas", | ||
| 223 | .description = NULL_IF_CONFIG_SMALL("Apply Haas Stereo Enhancer."), | ||
| 224 | .priv_size = sizeof(HaasContext), | ||
| 225 | .priv_class = &haas_class, | ||
| 226 | .uninit = uninit, | ||
| 227 | FILTER_INPUTS(inputs), | ||
| 228 | FILTER_OUTPUTS(ff_audio_default_filterpad), | ||
| 229 | FILTER_QUERY_FUNC2(query_formats), | ||
| 230 | }; | ||
| 231 |