| /* |
| * Copyright (c) 2001-2010 Vladimir Sadovnikov |
| * |
| * 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/channel_layout.h" |
| #include "libavutil/opt.h" |
| #include "avfilter.h" |
| #include "audio.h" |
| #include "formats.h" |
| |
| #define MAX_HAAS_DELAY 40 |
| |
| typedef struct HaasContext { |
| const AVClass *class; |
| |
| int par_m_source; |
| double par_delay0; |
| double par_delay1; |
| int par_phase0; |
| int par_phase1; |
| int par_middle_phase; |
| double par_side_gain; |
| double par_gain0; |
| double par_gain1; |
| double par_balance0; |
| double par_balance1; |
| double level_in; |
| double level_out; |
| |
| double *buffer; |
| size_t buffer_size; |
| uint32_t write_ptr; |
| uint32_t delay[2]; |
| double balance_l[2]; |
| double balance_r[2]; |
| double phase0; |
| double phase1; |
| } HaasContext; |
| |
| #define OFFSET(x) offsetof(HaasContext, x) |
| #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
| |
| static const AVOption haas_options[] = { |
| { "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, |
| { "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, |
| { "side_gain", "set side gain", OFFSET(par_side_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, |
| { "middle_source", "set middle source", OFFSET(par_m_source), AV_OPT_TYPE_INT, {.i64=2}, 0, 3, A, "source" }, |
| { "left", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "source" }, |
| { "right", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "source" }, |
| { "mid", "L+R", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "source" }, |
| { "side", "L-R", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, "source" }, |
| { "middle_phase", "set middle phase", OFFSET(par_middle_phase), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A }, |
| { "left_delay", "set left delay", OFFSET(par_delay0), AV_OPT_TYPE_DOUBLE, {.dbl=2.05}, 0, MAX_HAAS_DELAY, A }, |
| { "left_balance", "set left balance", OFFSET(par_balance0), AV_OPT_TYPE_DOUBLE, {.dbl=-1.0}, -1, 1, A }, |
| { "left_gain", "set left gain", OFFSET(par_gain0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, |
| { "left_phase", "set left phase", OFFSET(par_phase0), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A }, |
| { "right_delay", "set right delay", OFFSET(par_delay1), AV_OPT_TYPE_DOUBLE, {.dbl=2.12}, 0, MAX_HAAS_DELAY, A }, |
| { "right_balance", "set right balance", OFFSET(par_balance1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, -1, 1, A }, |
| { "right_gain", "set right gain", OFFSET(par_gain1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, |
| { "right_phase", "set right phase", OFFSET(par_phase1), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A }, |
| { NULL } |
| }; |
| |
| AVFILTER_DEFINE_CLASS(haas); |
| |
| static int query_formats(AVFilterContext *ctx) |
| { |
| AVFilterFormats *formats = NULL; |
| AVFilterChannelLayouts *layout = NULL; |
| int ret; |
| |
| if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 || |
| (ret = ff_set_common_formats (ctx , formats )) < 0 || |
| (ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO)) < 0 || |
| (ret = ff_set_common_channel_layouts (ctx , layout )) < 0) |
| return ret; |
| |
| formats = ff_all_samplerates(); |
| return ff_set_common_samplerates(ctx, formats); |
| } |
| |
| static int config_input(AVFilterLink *inlink) |
| { |
| AVFilterContext *ctx = inlink->dst; |
| HaasContext *s = ctx->priv; |
| size_t min_buf_size = (size_t)(inlink->sample_rate * MAX_HAAS_DELAY * 0.001); |
| size_t new_buf_size = 1; |
| |
| while (new_buf_size < min_buf_size) |
| new_buf_size <<= 1; |
| |
| av_freep(&s->buffer); |
| s->buffer = av_calloc(new_buf_size, sizeof(*s->buffer)); |
| if (!s->buffer) |
| return AVERROR(ENOMEM); |
| |
| s->buffer_size = new_buf_size; |
| s->write_ptr = 0; |
| |
| s->delay[0] = (uint32_t)(s->par_delay0 * 0.001 * inlink->sample_rate); |
| s->delay[1] = (uint32_t)(s->par_delay1 * 0.001 * inlink->sample_rate); |
| |
| s->phase0 = s->par_phase0 ? 1.0 : -1.0; |
| s->phase1 = s->par_phase1 ? 1.0 : -1.0; |
| |
| s->balance_l[0] = (s->par_balance0 + 1) / 2 * s->par_gain0 * s->phase0; |
| s->balance_r[0] = (1.0 - (s->par_balance0 + 1) / 2) * (s->par_gain0) * s->phase0; |
| s->balance_l[1] = (s->par_balance1 + 1) / 2 * s->par_gain1 * s->phase1; |
| s->balance_r[1] = (1.0 - (s->par_balance1 + 1) / 2) * (s->par_gain1) * s->phase1; |
| |
| return 0; |
| } |
| |
| static int filter_frame(AVFilterLink *inlink, AVFrame *in) |
| { |
| AVFilterContext *ctx = inlink->dst; |
| AVFilterLink *outlink = ctx->outputs[0]; |
| HaasContext *s = ctx->priv; |
| const double *src = (const double *)in->data[0]; |
| const double level_in = s->level_in; |
| const double level_out = s->level_out; |
| const uint32_t mask = s->buffer_size - 1; |
| double *buffer = s->buffer; |
| AVFrame *out; |
| double *dst; |
| int n; |
| |
| 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); |
| } |
| dst = (double *)out->data[0]; |
| |
| for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) { |
| double mid, side[2], side_l, side_r; |
| uint32_t s0_ptr, s1_ptr; |
| |
| switch (s->par_m_source) { |
| case 0: mid = src[0]; break; |
| case 1: mid = src[1]; break; |
| case 2: mid = (src[0] + src[1]) * 0.5; break; |
| case 3: mid = (src[0] - src[1]) * 0.5; break; |
| } |
| |
| mid *= level_in; |
| |
| buffer[s->write_ptr] = mid; |
| |
| s0_ptr = (s->write_ptr + s->buffer_size - s->delay[0]) & mask; |
| s1_ptr = (s->write_ptr + s->buffer_size - s->delay[1]) & mask; |
| |
| if (s->par_middle_phase) |
| mid = -mid; |
| |
| side[0] = buffer[s0_ptr] * s->par_side_gain; |
| side[1] = buffer[s1_ptr] * s->par_side_gain; |
| side_l = side[0] * s->balance_l[0] - side[1] * s->balance_l[1]; |
| side_r = side[1] * s->balance_r[1] - side[0] * s->balance_r[0]; |
| |
| dst[0] = (mid + side_l) * level_out; |
| dst[1] = (mid + side_r) * level_out; |
| |
| s->write_ptr = (s->write_ptr + 1) & mask; |
| } |
| |
| if (out != in) |
| av_frame_free(&in); |
| return ff_filter_frame(outlink, out); |
| } |
| |
| static av_cold void uninit(AVFilterContext *ctx) |
| { |
| HaasContext *s = ctx->priv; |
| |
| av_freep(&s->buffer); |
| s->buffer_size = 0; |
| } |
| |
| 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_haas = { |
| .name = "haas", |
| .description = NULL_IF_CONFIG_SMALL("Apply Haas Stereo Enhancer."), |
| .query_formats = query_formats, |
| .priv_size = sizeof(HaasContext), |
| .priv_class = &haas_class, |
| .uninit = uninit, |
| .inputs = inputs, |
| .outputs = outputs, |
| }; |