/* * 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/mem.h" #include "libavutil/opt.h" #include "avfilter.h" #include "audio.h" #include "filters.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, .unit = "source" }, { "left", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "source" }, { "right", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "source" }, { "mid", "L+R", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, .unit = "source" }, { "side", "L-R", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, .unit = "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 , &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO)) < 0 || (ret = ff_set_common_channel_layouts (ctx , layout )) < 0) return ret; return ff_set_common_all_samplerates(ctx); } 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, }, }; const AVFilter ff_af_haas = { .name = "haas", .description = NULL_IF_CONFIG_SMALL("Apply Haas Stereo Enhancer."), .priv_size = sizeof(HaasContext), .priv_class = &haas_class, .uninit = uninit, FILTER_INPUTS(inputs), FILTER_OUTPUTS(ff_audio_default_filterpad), FILTER_QUERY_FUNC(query_formats), };