/* * Copyright (c) 2017 Paul B Mahol * * 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/avassert.h" #include "libavutil/channel_layout.h" #include "libavutil/mem.h" #include "libavutil/opt.h" #include "libavutil/tx.h" #include "avfilter.h" #include "audio.h" #include "filters.h" #include "formats.h" #include "window_func.h" enum SurroundChannel { SC_FL, SC_FR, SC_FC, SC_LF, SC_BL, SC_BR, SC_BC, SC_SL, SC_SR, SC_NB, }; static const int ch_map[SC_NB] = { [SC_FL] = AV_CHAN_FRONT_LEFT, [SC_FR] = AV_CHAN_FRONT_RIGHT, [SC_FC] = AV_CHAN_FRONT_CENTER, [SC_LF] = AV_CHAN_LOW_FREQUENCY, [SC_BL] = AV_CHAN_BACK_LEFT, [SC_BR] = AV_CHAN_BACK_RIGHT, [SC_BC] = AV_CHAN_BACK_CENTER, [SC_SL] = AV_CHAN_SIDE_LEFT, [SC_SR] = AV_CHAN_SIDE_RIGHT, }; static const int sc_map[16] = { [AV_CHAN_FRONT_LEFT ] = SC_FL, [AV_CHAN_FRONT_RIGHT ] = SC_FR, [AV_CHAN_FRONT_CENTER ] = SC_FC, [AV_CHAN_LOW_FREQUENCY] = SC_LF, [AV_CHAN_BACK_LEFT ] = SC_BL, [AV_CHAN_BACK_RIGHT ] = SC_BR, [AV_CHAN_BACK_CENTER ] = SC_BC, [AV_CHAN_SIDE_LEFT ] = SC_SL, [AV_CHAN_SIDE_RIGHT ] = SC_SR, }; typedef struct AudioSurroundContext { const AVClass *class; AVChannelLayout out_ch_layout; AVChannelLayout in_ch_layout; float level_in; float level_out; float f_i[SC_NB]; float f_o[SC_NB]; int lfe_mode; float smooth; float angle; float focus; int win_size; int win_func; float win_gain; float overlap; float all_x; float all_y; float f_x[SC_NB]; float f_y[SC_NB]; float *input_levels; float *output_levels; int output_lfe; int create_lfe; int lowcutf; int highcutf; float lowcut; float highcut; int nb_in_channels; int nb_out_channels; AVFrame *factors; AVFrame *sfactors; AVFrame *input_in; AVFrame *input; AVFrame *output; AVFrame *output_mag; AVFrame *output_ph; AVFrame *output_out; AVFrame *overlap_buffer; AVFrame *window; float *x_pos; float *y_pos; float *l_phase; float *r_phase; float *c_phase; float *c_mag; float *lfe_mag; float *lfe_phase; float *mag_total; int rdft_size; int hop_size; AVTXContext **rdft, **irdft; av_tx_fn tx_fn, itx_fn; float *window_func_lut; void (*filter)(AVFilterContext *ctx); void (*upmix)(AVFilterContext *ctx, int ch); void (*upmix_5_0)(AVFilterContext *ctx, float c_re, float c_im, float mag_totall, float mag_totalr, float fl_phase, float fr_phase, float bl_phase, float br_phase, float sl_phase, float sr_phase, float xl, float yl, float xr, float yr, int n); void (*upmix_5_1)(AVFilterContext *ctx, float c_re, float c_im, float lfe_re, float lfe_im, float mag_totall, float mag_totalr, float fl_phase, float fr_phase, float bl_phase, float br_phase, float sl_phase, float sr_phase, float xl, float yl, float xr, float yr, int n); } AudioSurroundContext; static int query_formats(AVFilterContext *ctx) { AudioSurroundContext *s = ctx->priv; AVFilterFormats *formats = NULL; AVFilterChannelLayouts *layouts = NULL; int ret; ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLTP); if (ret) return ret; ret = ff_set_common_formats(ctx, formats); if (ret) return ret; layouts = NULL; ret = ff_add_channel_layout(&layouts, &s->out_ch_layout); if (ret) return ret; ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts); if (ret) return ret; layouts = NULL; ret = ff_add_channel_layout(&layouts, &s->in_ch_layout); if (ret) return ret; ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts); if (ret) return ret; return ff_set_common_all_samplerates(ctx); } static void set_input_levels(AVFilterContext *ctx) { AudioSurroundContext *s = ctx->priv; for (int ch = 0; ch < s->nb_in_channels && s->level_in >= 0.f; ch++) s->input_levels[ch] = s->level_in; s->level_in = -1.f; for (int n = 0; n < SC_NB; n++) { const int ch = av_channel_layout_index_from_channel(&s->in_ch_layout, ch_map[n]); if (ch >= 0) s->input_levels[ch] = s->f_i[n]; } } static void set_output_levels(AVFilterContext *ctx) { AudioSurroundContext *s = ctx->priv; for (int ch = 0; ch < s->nb_out_channels && s->level_out >= 0.f; ch++) s->output_levels[ch] = s->level_out; s->level_out = -1.f; for (int n = 0; n < SC_NB; n++) { const int ch = av_channel_layout_index_from_channel(&s->out_ch_layout, ch_map[n]); if (ch >= 0) s->output_levels[ch] = s->f_o[n]; } } static int config_input(AVFilterLink *inlink) { AVFilterContext *ctx = inlink->dst; AudioSurroundContext *s = ctx->priv; int ret; s->rdft = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->rdft)); if (!s->rdft) return AVERROR(ENOMEM); s->nb_in_channels = inlink->ch_layout.nb_channels; for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) { float scale = 1.f; ret = av_tx_init(&s->rdft[ch], &s->tx_fn, AV_TX_FLOAT_RDFT, 0, s->win_size, &scale, 0); if (ret < 0) return ret; } s->input_levels = av_malloc_array(s->nb_in_channels, sizeof(*s->input_levels)); if (!s->input_levels) return AVERROR(ENOMEM); set_input_levels(ctx); s->window = ff_get_audio_buffer(inlink, s->win_size * 2); if (!s->window) return AVERROR(ENOMEM); s->input_in = ff_get_audio_buffer(inlink, s->win_size * 2); if (!s->input_in) return AVERROR(ENOMEM); s->input = ff_get_audio_buffer(inlink, s->win_size + 2); if (!s->input) return AVERROR(ENOMEM); s->lowcut = 1.f * s->lowcutf / (inlink->sample_rate * 0.5) * (s->win_size / 2); s->highcut = 1.f * s->highcutf / (inlink->sample_rate * 0.5) * (s->win_size / 2); return 0; } static int config_output(AVFilterLink *outlink) { AVFilterContext *ctx = outlink->src; AudioSurroundContext *s = ctx->priv; int ret; s->irdft = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->irdft)); if (!s->irdft) return AVERROR(ENOMEM); s->nb_out_channels = outlink->ch_layout.nb_channels; for (int ch = 0; ch < outlink->ch_layout.nb_channels; ch++) { float iscale = 1.f; ret = av_tx_init(&s->irdft[ch], &s->itx_fn, AV_TX_FLOAT_RDFT, 1, s->win_size, &iscale, 0); if (ret < 0) return ret; } s->output_levels = av_malloc_array(s->nb_out_channels, sizeof(*s->output_levels)); if (!s->output_levels) return AVERROR(ENOMEM); set_output_levels(ctx); s->factors = ff_get_audio_buffer(outlink, s->win_size + 2); s->sfactors = ff_get_audio_buffer(outlink, s->win_size + 2); s->output_ph = ff_get_audio_buffer(outlink, s->win_size + 2); s->output_mag = ff_get_audio_buffer(outlink, s->win_size + 2); s->output_out = ff_get_audio_buffer(outlink, s->win_size + 2); s->output = ff_get_audio_buffer(outlink, s->win_size + 2); s->overlap_buffer = ff_get_audio_buffer(outlink, s->win_size * 2); if (!s->overlap_buffer || !s->output || !s->output_out || !s->output_mag || !s->output_ph || !s->factors || !s->sfactors) return AVERROR(ENOMEM); s->rdft_size = s->win_size / 2 + 1; s->x_pos = av_calloc(s->rdft_size, sizeof(*s->x_pos)); s->y_pos = av_calloc(s->rdft_size, sizeof(*s->y_pos)); s->l_phase = av_calloc(s->rdft_size, sizeof(*s->l_phase)); s->r_phase = av_calloc(s->rdft_size, sizeof(*s->r_phase)); s->c_mag = av_calloc(s->rdft_size, sizeof(*s->c_mag)); s->c_phase = av_calloc(s->rdft_size, sizeof(*s->c_phase)); s->mag_total = av_calloc(s->rdft_size, sizeof(*s->mag_total)); s->lfe_mag = av_calloc(s->rdft_size, sizeof(*s->lfe_mag)); s->lfe_phase = av_calloc(s->rdft_size, sizeof(*s->lfe_phase)); if (!s->x_pos || !s->y_pos || !s->l_phase || !s->r_phase || !s->lfe_phase || !s->c_phase || !s->mag_total || !s->lfe_mag || !s->c_mag) return AVERROR(ENOMEM); return 0; } static float sqrf(float x) { return x * x; } static float r_distance(float a) { return fminf(sqrtf(1.f + sqrf(tanf(a))), sqrtf(1.f + sqrf(1.f / tanf(a)))); } #define MIN_MAG_SUM 0.00000001f static void angle_transform(float *x, float *y, float angle) { float reference, r, a; if (angle == 90.f) return; reference = angle * M_PIf / 180.f; r = hypotf(*x, *y); a = atan2f(*x, *y); r /= r_distance(a); if (fabsf(a) <= M_PI_4f) a *= reference / M_PI_2f; else a = M_PIf + (-2.f * M_PIf + reference) * (M_PIf - fabsf(a)) * FFDIFFSIGN(a, 0.f) / (3.f * M_PI_2f); r *= r_distance(a); *x = av_clipf(sinf(a) * r, -1.f, 1.f); *y = av_clipf(cosf(a) * r, -1.f, 1.f); } static void focus_transform(float *x, float *y, float focus) { float a, r, ra; if (focus == 0.f) return; a = atan2f(*x, *y); ra = r_distance(a); r = av_clipf(hypotf(*x, *y) / ra, 0.f, 1.f); r = focus > 0.f ? 1.f - powf(1.f - r, 1.f + focus * 20.f) : powf(r, 1.f - focus * 20.f); r *= ra; *x = av_clipf(sinf(a) * r, -1.f, 1.f); *y = av_clipf(cosf(a) * r, -1.f, 1.f); } static void stereo_position(float a, float p, float *x, float *y) { av_assert2(a >= -1.f && a <= 1.f); av_assert2(p >= 0.f && p <= M_PIf); *x = av_clipf(a+a*fmaxf(0.f, p*p-M_PI_2f), -1.f, 1.f); *y = av_clipf(cosf(a*M_PI_2f+M_PIf)*cosf(M_PI_2f-p/M_PIf)*M_LN10f+1.f, -1.f, 1.f); } static inline void get_lfe(int output_lfe, int n, float lowcut, float highcut, float *lfe_mag, float c_mag, float *mag_total, int lfe_mode) { if (output_lfe && n < highcut) { *lfe_mag = n < lowcut ? 1.f : .5f*(1.f+cosf(M_PIf*(lowcut-n)/(lowcut-highcut))); *lfe_mag *= c_mag; if (lfe_mode) *mag_total -= *lfe_mag; } else { *lfe_mag = 0.f; } } #define TRANSFORM \ dst[2 * n ] = mag * cosf(ph); \ dst[2 * n + 1] = mag * sinf(ph); static void calculate_factors(AVFilterContext *ctx, int ch, int chan) { AudioSurroundContext *s = ctx->priv; float *factor = (float *)s->factors->extended_data[ch]; const float f_x = s->f_x[sc_map[chan >= 0 ? chan : 0]]; const float f_y = s->f_y[sc_map[chan >= 0 ? chan : 0]]; const int rdft_size = s->rdft_size; const float *x = s->x_pos; const float *y = s->y_pos; switch (chan) { case AV_CHAN_FRONT_CENTER: for (int n = 0; n < rdft_size; n++) factor[n] = powf(1.f - fabsf(x[n]), f_x) * powf((y[n] + 1.f) * .5f, f_y); break; case AV_CHAN_FRONT_LEFT: for (int n = 0; n < rdft_size; n++) factor[n] = powf(.5f * ( x[n] + 1.f), f_x) * powf((y[n] + 1.f) * .5f, f_y); break; case AV_CHAN_FRONT_RIGHT: for (int n = 0; n < rdft_size; n++) factor[n] = powf(.5f * (-x[n] + 1.f), f_x) * powf((y[n] + 1.f) * .5f, f_y); break; case AV_CHAN_LOW_FREQUENCY: for (int n = 0; n < rdft_size; n++) factor[n] = powf(1.f - fabsf(x[n]), f_x) * powf((1.f - fabsf(y[n])), f_y); break; case AV_CHAN_BACK_CENTER: for (int n = 0; n < rdft_size; n++) factor[n] = powf(1.f - fabsf(x[n]), f_x) * powf((1.f - y[n]) * .5f, f_y); break; case AV_CHAN_BACK_LEFT: for (int n = 0; n < rdft_size; n++) factor[n] = powf(.5f * ( x[n] + 1.f), f_x) * powf(1.f - ((y[n] + 1.f) * .5f), f_y); break; case AV_CHAN_BACK_RIGHT: for (int n = 0; n < rdft_size; n++) factor[n] = powf(.5f * (-x[n] + 1.f), f_x) * powf(1.f - ((y[n] + 1.f) * .5f), f_y); break; case AV_CHAN_SIDE_LEFT: for (int n = 0; n < rdft_size; n++) factor[n] = powf(.5f * ( x[n] + 1.f), f_x) * powf(1.f - fabsf(y[n]), f_y); break; case AV_CHAN_SIDE_RIGHT: for (int n = 0; n < rdft_size; n++) factor[n] = powf(.5f * (-x[n] + 1.f), f_x) * powf(1.f - fabsf(y[n]), f_y); break; default: for (int n = 0; n < rdft_size; n++) factor[n] = 1.f; break; } } static void do_transform(AVFilterContext *ctx, int ch) { AudioSurroundContext *s = ctx->priv; float *sfactor = (float *)s->sfactors->extended_data[ch]; float *factor = (float *)s->factors->extended_data[ch]; float *omag = (float *)s->output_mag->extended_data[ch]; float *oph = (float *)s->output_ph->extended_data[ch]; float *dst = (float *)s->output->extended_data[ch]; const int rdft_size = s->rdft_size; const float smooth = s->smooth; if (smooth > 0.f) { for (int n = 0; n < rdft_size; n++) sfactor[n] = smooth * factor[n] + (1.f - smooth) * sfactor[n]; factor = sfactor; } for (int n = 0; n < rdft_size; n++) omag[n] *= factor[n]; for (int n = 0; n < rdft_size; n++) { const float mag = omag[n]; const float ph = oph[n]; TRANSFORM } } static void stereo_copy(AVFilterContext *ctx, int ch, int chan) { AudioSurroundContext *s = ctx->priv; float *omag = (float *)s->output_mag->extended_data[ch]; float *oph = (float *)s->output_ph->extended_data[ch]; const float *mag_total = s->mag_total; const int rdft_size = s->rdft_size; const float *c_phase = s->c_phase; const float *l_phase = s->l_phase; const float *r_phase = s->r_phase; const float *lfe_mag = s->lfe_mag; const float *c_mag = s->c_mag; switch (chan) { case AV_CHAN_FRONT_CENTER: memcpy(omag, c_mag, rdft_size * sizeof(*omag)); break; case AV_CHAN_LOW_FREQUENCY: memcpy(omag, lfe_mag, rdft_size * sizeof(*omag)); break; case AV_CHAN_FRONT_LEFT: case AV_CHAN_FRONT_RIGHT: case AV_CHAN_BACK_CENTER: case AV_CHAN_BACK_LEFT: case AV_CHAN_BACK_RIGHT: case AV_CHAN_SIDE_LEFT: case AV_CHAN_SIDE_RIGHT: memcpy(omag, mag_total, rdft_size * sizeof(*omag)); break; default: break; } switch (chan) { case AV_CHAN_FRONT_CENTER: case AV_CHAN_LOW_FREQUENCY: case AV_CHAN_BACK_CENTER: memcpy(oph, c_phase, rdft_size * sizeof(*oph)); break; case AV_CHAN_FRONT_LEFT: case AV_CHAN_BACK_LEFT: case AV_CHAN_SIDE_LEFT: memcpy(oph, l_phase, rdft_size * sizeof(*oph)); break; case AV_CHAN_FRONT_RIGHT: case AV_CHAN_BACK_RIGHT: case AV_CHAN_SIDE_RIGHT: memcpy(oph, r_phase, rdft_size * sizeof(*oph)); break; default: break; } } static void stereo_upmix(AVFilterContext *ctx, int ch) { AudioSurroundContext *s = ctx->priv; const int chan = av_channel_layout_channel_from_index(&s->out_ch_layout, ch); calculate_factors(ctx, ch, chan); stereo_copy(ctx, ch, chan); do_transform(ctx, ch); } static void l2_1_upmix(AVFilterContext *ctx, int ch) { AudioSurroundContext *s = ctx->priv; const int chan = av_channel_layout_channel_from_index(&s->out_ch_layout, ch); float *omag = (float *)s->output_mag->extended_data[ch]; float *oph = (float *)s->output_ph->extended_data[ch]; const float *mag_total = s->mag_total; const float *lfe_phase = s->lfe_phase; const int rdft_size = s->rdft_size; const float *c_phase = s->c_phase; const float *l_phase = s->l_phase; const float *r_phase = s->r_phase; const float *lfe_mag = s->lfe_mag; const float *c_mag = s->c_mag; switch (chan) { case AV_CHAN_LOW_FREQUENCY: calculate_factors(ctx, ch, -1); break; default: calculate_factors(ctx, ch, chan); break; } switch (chan) { case AV_CHAN_FRONT_CENTER: memcpy(omag, c_mag, rdft_size * sizeof(*omag)); break; case AV_CHAN_LOW_FREQUENCY: memcpy(omag, lfe_mag, rdft_size * sizeof(*omag)); break; case AV_CHAN_FRONT_LEFT: case AV_CHAN_FRONT_RIGHT: case AV_CHAN_BACK_CENTER: case AV_CHAN_BACK_LEFT: case AV_CHAN_BACK_RIGHT: case AV_CHAN_SIDE_LEFT: case AV_CHAN_SIDE_RIGHT: memcpy(omag, mag_total, rdft_size * sizeof(*omag)); break; default: break; } switch (chan) { case AV_CHAN_LOW_FREQUENCY: memcpy(oph, lfe_phase, rdft_size * sizeof(*oph)); break; case AV_CHAN_FRONT_CENTER: case AV_CHAN_BACK_CENTER: memcpy(oph, c_phase, rdft_size * sizeof(*oph)); break; case AV_CHAN_FRONT_LEFT: case AV_CHAN_BACK_LEFT: case AV_CHAN_SIDE_LEFT: memcpy(oph, l_phase, rdft_size * sizeof(*oph)); break; case AV_CHAN_FRONT_RIGHT: case AV_CHAN_BACK_RIGHT: case AV_CHAN_SIDE_RIGHT: memcpy(oph, r_phase, rdft_size * sizeof(*oph)); break; default: break; } do_transform(ctx, ch); } static void surround_upmix(AVFilterContext *ctx, int ch) { AudioSurroundContext *s = ctx->priv; const int chan = av_channel_layout_channel_from_index(&s->out_ch_layout, ch); switch (chan) { case AV_CHAN_FRONT_CENTER: calculate_factors(ctx, ch, -1); break; default: calculate_factors(ctx, ch, chan); break; } stereo_copy(ctx, ch, chan); do_transform(ctx, ch); } static void upmix_7_1_5_0_side(AVFilterContext *ctx, float c_re, float c_im, float mag_totall, float mag_totalr, float fl_phase, float fr_phase, float bl_phase, float br_phase, float sl_phase, float sr_phase, float xl, float yl, float xr, float yr, int n) { float fl_mag, fr_mag, ls_mag, rs_mag, lb_mag, rb_mag; float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb, *dstlfe; float lfe_mag, c_phase, mag_total = (mag_totall + mag_totalr) * 0.5f; AudioSurroundContext *s = ctx->priv; dstl = (float *)s->output->extended_data[0]; dstr = (float *)s->output->extended_data[1]; dstc = (float *)s->output->extended_data[2]; dstlfe = (float *)s->output->extended_data[3]; dstlb = (float *)s->output->extended_data[4]; dstrb = (float *)s->output->extended_data[5]; dstls = (float *)s->output->extended_data[6]; dstrs = (float *)s->output->extended_data[7]; c_phase = atan2f(c_im, c_re); get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, hypotf(c_re, c_im), &mag_total, s->lfe_mode); fl_mag = powf(.5f * (xl + 1.f), s->f_x[SC_FL]) * powf((yl + 1.f) * .5f, s->f_y[SC_FL]) * mag_totall; fr_mag = powf(.5f * (xr + 1.f), s->f_x[SC_FR]) * powf((yr + 1.f) * .5f, s->f_y[SC_FR]) * mag_totalr; lb_mag = powf(.5f * (-xl + 1.f), s->f_x[SC_BL]) * powf((yl + 1.f) * .5f, s->f_y[SC_BL]) * mag_totall; rb_mag = powf(.5f * (-xr + 1.f), s->f_x[SC_BR]) * powf((yr + 1.f) * .5f, s->f_y[SC_BR]) * mag_totalr; ls_mag = powf(1.f - fabsf(xl), s->f_x[SC_SL]) * powf((yl + 1.f) * .5f, s->f_y[SC_SL]) * mag_totall; rs_mag = powf(1.f - fabsf(xr), s->f_x[SC_SR]) * powf((yr + 1.f) * .5f, s->f_y[SC_SR]) * mag_totalr; dstl[2 * n ] = fl_mag * cosf(fl_phase); dstl[2 * n + 1] = fl_mag * sinf(fl_phase); dstr[2 * n ] = fr_mag * cosf(fr_phase); dstr[2 * n + 1] = fr_mag * sinf(fr_phase); dstc[2 * n ] = c_re; dstc[2 * n + 1] = c_im; dstlfe[2 * n ] = lfe_mag * cosf(c_phase); dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase); dstlb[2 * n ] = lb_mag * cosf(bl_phase); dstlb[2 * n + 1] = lb_mag * sinf(bl_phase); dstrb[2 * n ] = rb_mag * cosf(br_phase); dstrb[2 * n + 1] = rb_mag * sinf(br_phase); dstls[2 * n ] = ls_mag * cosf(sl_phase); dstls[2 * n + 1] = ls_mag * sinf(sl_phase); dstrs[2 * n ] = rs_mag * cosf(sr_phase); dstrs[2 * n + 1] = rs_mag * sinf(sr_phase); } static void upmix_7_1_5_1(AVFilterContext *ctx, float c_re, float c_im, float lfe_re, float lfe_im, float mag_totall, float mag_totalr, float fl_phase, float fr_phase, float bl_phase, float br_phase, float sl_phase, float sr_phase, float xl, float yl, float xr, float yr, int n) { float fl_mag, fr_mag, ls_mag, rs_mag, lb_mag, rb_mag; float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb, *dstlfe; AudioSurroundContext *s = ctx->priv; dstl = (float *)s->output->extended_data[0]; dstr = (float *)s->output->extended_data[1]; dstc = (float *)s->output->extended_data[2]; dstlfe = (float *)s->output->extended_data[3]; dstlb = (float *)s->output->extended_data[4]; dstrb = (float *)s->output->extended_data[5]; dstls = (float *)s->output->extended_data[6]; dstrs = (float *)s->output->extended_data[7]; fl_mag = powf(.5f * (xl + 1.f), s->f_x[SC_FL]) * powf((yl + 1.f) * .5f, s->f_y[SC_FL]) * mag_totall; fr_mag = powf(.5f * (xr + 1.f), s->f_x[SC_FR]) * powf((yr + 1.f) * .5f, s->f_y[SC_FR]) * mag_totalr; lb_mag = powf(.5f * (-xl + 1.f), s->f_x[SC_BL]) * powf((yl + 1.f) * .5f, s->f_y[SC_BL]) * mag_totall; rb_mag = powf(.5f * (-xr + 1.f), s->f_x[SC_BR]) * powf((yr + 1.f) * .5f, s->f_y[SC_BR]) * mag_totalr; ls_mag = powf(1.f - fabsf(xl), s->f_x[SC_SL]) * powf((yl + 1.f) * .5f, s->f_y[SC_SL]) * mag_totall; rs_mag = powf(1.f - fabsf(xr), s->f_x[SC_SR]) * powf((yr + 1.f) * .5f, s->f_y[SC_SR]) * mag_totalr; dstl[2 * n ] = fl_mag * cosf(fl_phase); dstl[2 * n + 1] = fl_mag * sinf(fl_phase); dstr[2 * n ] = fr_mag * cosf(fr_phase); dstr[2 * n + 1] = fr_mag * sinf(fr_phase); dstc[2 * n ] = c_re; dstc[2 * n + 1] = c_im; dstlfe[2 * n ] = lfe_re; dstlfe[2 * n + 1] = lfe_im; dstlb[2 * n ] = lb_mag * cosf(bl_phase); dstlb[2 * n + 1] = lb_mag * sinf(bl_phase); dstrb[2 * n ] = rb_mag * cosf(br_phase); dstrb[2 * n + 1] = rb_mag * sinf(br_phase); dstls[2 * n ] = ls_mag * cosf(sl_phase); dstls[2 * n + 1] = ls_mag * sinf(sl_phase); dstrs[2 * n ] = rs_mag * cosf(sr_phase); dstrs[2 * n + 1] = rs_mag * sinf(sr_phase); } static void filter_stereo(AVFilterContext *ctx) { AudioSurroundContext *s = ctx->priv; const float *srcl = (const float *)s->input->extended_data[0]; const float *srcr = (const float *)s->input->extended_data[1]; const int output_lfe = s->output_lfe && s->create_lfe; const int rdft_size = s->rdft_size; const int lfe_mode = s->lfe_mode; const float highcut = s->highcut; const float lowcut = s->lowcut; const float angle = s->angle; const float focus = s->focus; float *magtotal = s->mag_total; float *lfemag = s->lfe_mag; float *lphase = s->l_phase; float *rphase = s->r_phase; float *cphase = s->c_phase; float *cmag = s->c_mag; float *xpos = s->x_pos; float *ypos = s->y_pos; for (int n = 0; n < rdft_size; n++) { float l_re = srcl[2 * n], r_re = srcr[2 * n]; float l_im = srcl[2 * n + 1], r_im = srcr[2 * n + 1]; float c_phase = atan2f(l_im + r_im, l_re + r_re); float l_mag = hypotf(l_re, l_im); float r_mag = hypotf(r_re, r_im); float mag_total = hypotf(l_mag, r_mag); float l_phase = atan2f(l_im, l_re); float r_phase = atan2f(r_im, r_re); float phase_dif = fabsf(l_phase - r_phase); float mag_sum = l_mag + r_mag; float c_mag = mag_sum * 0.5f; float mag_dif, x, y; mag_sum = mag_sum < MIN_MAG_SUM ? 1.f : mag_sum; mag_dif = (l_mag - r_mag) / mag_sum; if (phase_dif > M_PIf) phase_dif = 2.f * M_PIf - phase_dif; stereo_position(mag_dif, phase_dif, &x, &y); angle_transform(&x, &y, angle); focus_transform(&x, &y, focus); get_lfe(output_lfe, n, lowcut, highcut, &lfemag[n], c_mag, &mag_total, lfe_mode); xpos[n] = x; ypos[n] = y; lphase[n] = l_phase; rphase[n] = r_phase; cmag[n] = c_mag; cphase[n] = c_phase; magtotal[n] = mag_total; } } static void filter_2_1(AVFilterContext *ctx) { AudioSurroundContext *s = ctx->priv; const float *srcl = (const float *)s->input->extended_data[0]; const float *srcr = (const float *)s->input->extended_data[1]; const float *srclfe = (const float *)s->input->extended_data[2]; const int rdft_size = s->rdft_size; const float angle = s->angle; const float focus = s->focus; float *magtotal = s->mag_total; float *lfephase = s->lfe_phase; float *lfemag = s->lfe_mag; float *lphase = s->l_phase; float *rphase = s->r_phase; float *cphase = s->c_phase; float *cmag = s->c_mag; float *xpos = s->x_pos; float *ypos = s->y_pos; for (int n = 0; n < rdft_size; n++) { float l_re = srcl[2 * n], r_re = srcr[2 * n]; float l_im = srcl[2 * n + 1], r_im = srcr[2 * n + 1]; float lfe_re = srclfe[2 * n], lfe_im = srclfe[2 * n + 1]; float c_phase = atan2f(l_im + r_im, l_re + r_re); float l_mag = hypotf(l_re, l_im); float r_mag = hypotf(r_re, r_im); float lfe_mag = hypotf(lfe_re, lfe_im); float lfe_phase = atan2f(lfe_im, lfe_re); float mag_total = hypotf(l_mag, r_mag); float l_phase = atan2f(l_im, l_re); float r_phase = atan2f(r_im, r_re); float phase_dif = fabsf(l_phase - r_phase); float mag_sum = l_mag + r_mag; float c_mag = mag_sum * 0.5f; float mag_dif, x, y; mag_sum = mag_sum < MIN_MAG_SUM ? 1.f : mag_sum; mag_dif = (l_mag - r_mag) / mag_sum; if (phase_dif > M_PIf) phase_dif = 2.f * M_PIf - phase_dif; stereo_position(mag_dif, phase_dif, &x, &y); angle_transform(&x, &y, angle); focus_transform(&x, &y, focus); xpos[n] = x; ypos[n] = y; lphase[n] = l_phase; rphase[n] = r_phase; cmag[n] = c_mag; cphase[n] = c_phase; lfemag[n] = lfe_mag; lfephase[n] = lfe_phase; magtotal[n] = mag_total; } } static void filter_surround(AVFilterContext *ctx) { AudioSurroundContext *s = ctx->priv; const float *srcl = (const float *)s->input->extended_data[0]; const float *srcr = (const float *)s->input->extended_data[1]; const float *srcc = (const float *)s->input->extended_data[2]; const int output_lfe = s->output_lfe && s->create_lfe; const int rdft_size = s->rdft_size; const int lfe_mode = s->lfe_mode; const float highcut = s->highcut; const float lowcut = s->lowcut; const float angle = s->angle; const float focus = s->focus; float *magtotal = s->mag_total; float *lfemag = s->lfe_mag; float *lphase = s->l_phase; float *rphase = s->r_phase; float *cphase = s->c_phase; float *cmag = s->c_mag; float *xpos = s->x_pos; float *ypos = s->y_pos; for (int n = 0; n < rdft_size; n++) { float l_re = srcl[2 * n], r_re = srcr[2 * n]; float l_im = srcl[2 * n + 1], r_im = srcr[2 * n + 1]; float c_re = srcc[2 * n], c_im = srcc[2 * n + 1]; float c_phase = atan2f(c_im, c_re); float c_mag = hypotf(c_re, c_im); float l_mag = hypotf(l_re, l_im); float r_mag = hypotf(r_re, r_im); float mag_total = hypotf(l_mag, r_mag); float l_phase = atan2f(l_im, l_re); float r_phase = atan2f(r_im, r_re); float phase_dif = fabsf(l_phase - r_phase); float mag_sum = l_mag + r_mag; float mag_dif, x, y; mag_sum = mag_sum < MIN_MAG_SUM ? 1.f : mag_sum; mag_dif = (l_mag - r_mag) / mag_sum; if (phase_dif > M_PIf) phase_dif = 2.f * M_PIf - phase_dif; stereo_position(mag_dif, phase_dif, &x, &y); angle_transform(&x, &y, angle); focus_transform(&x, &y, focus); get_lfe(output_lfe, n, lowcut, highcut, &lfemag[n], c_mag, &mag_total, lfe_mode); xpos[n] = x; ypos[n] = y; lphase[n] = l_phase; rphase[n] = r_phase; cmag[n] = c_mag; cphase[n] = c_phase; magtotal[n] = mag_total; } } static void filter_5_0_side(AVFilterContext *ctx) { AudioSurroundContext *s = ctx->priv; const int rdft_size = s->rdft_size; float *srcl, *srcr, *srcc, *srcsl, *srcsr; int n; srcl = (float *)s->input->extended_data[0]; srcr = (float *)s->input->extended_data[1]; srcc = (float *)s->input->extended_data[2]; srcsl = (float *)s->input->extended_data[3]; srcsr = (float *)s->input->extended_data[4]; for (n = 0; n < rdft_size; n++) { float fl_re = srcl[2 * n], fr_re = srcr[2 * n]; float fl_im = srcl[2 * n + 1], fr_im = srcr[2 * n + 1]; float c_re = srcc[2 * n], c_im = srcc[2 * n + 1]; float sl_re = srcsl[2 * n], sl_im = srcsl[2 * n + 1]; float sr_re = srcsr[2 * n], sr_im = srcsr[2 * n + 1]; float fl_mag = hypotf(fl_re, fl_im); float fr_mag = hypotf(fr_re, fr_im); float fl_phase = atan2f(fl_im, fl_re); float fr_phase = atan2f(fr_im, fr_re); float sl_mag = hypotf(sl_re, sl_im); float sr_mag = hypotf(sr_re, sr_im); float sl_phase = atan2f(sl_im, sl_re); float sr_phase = atan2f(sr_im, sr_re); float phase_difl = fabsf(fl_phase - sl_phase); float phase_difr = fabsf(fr_phase - sr_phase); float magl_sum = fl_mag + sl_mag; float magr_sum = fr_mag + sr_mag; float mag_difl = magl_sum < MIN_MAG_SUM ? FFDIFFSIGN(fl_mag, sl_mag) : (fl_mag - sl_mag) / magl_sum; float mag_difr = magr_sum < MIN_MAG_SUM ? FFDIFFSIGN(fr_mag, sr_mag) : (fr_mag - sr_mag) / magr_sum; float mag_totall = hypotf(fl_mag, sl_mag); float mag_totalr = hypotf(fr_mag, sr_mag); float bl_phase = atan2f(fl_im + sl_im, fl_re + sl_re); float br_phase = atan2f(fr_im + sr_im, fr_re + sr_re); float xl, yl; float xr, yr; if (phase_difl > M_PIf) phase_difl = 2.f * M_PIf - phase_difl; if (phase_difr > M_PIf) phase_difr = 2.f * M_PIf - phase_difr; stereo_position(mag_difl, phase_difl, &xl, &yl); stereo_position(mag_difr, phase_difr, &xr, &yr); s->upmix_5_0(ctx, c_re, c_im, mag_totall, mag_totalr, fl_phase, fr_phase, bl_phase, br_phase, sl_phase, sr_phase, xl, yl, xr, yr, n); } } static void filter_5_1_side(AVFilterContext *ctx) { AudioSurroundContext *s = ctx->priv; const int rdft_size = s->rdft_size; float *srcl, *srcr, *srcc, *srclfe, *srcsl, *srcsr; int n; srcl = (float *)s->input->extended_data[0]; srcr = (float *)s->input->extended_data[1]; srcc = (float *)s->input->extended_data[2]; srclfe = (float *)s->input->extended_data[3]; srcsl = (float *)s->input->extended_data[4]; srcsr = (float *)s->input->extended_data[5]; for (n = 0; n < rdft_size; n++) { float fl_re = srcl[2 * n], fr_re = srcr[2 * n]; float fl_im = srcl[2 * n + 1], fr_im = srcr[2 * n + 1]; float c_re = srcc[2 * n], c_im = srcc[2 * n + 1]; float lfe_re = srclfe[2 * n], lfe_im = srclfe[2 * n + 1]; float sl_re = srcsl[2 * n], sl_im = srcsl[2 * n + 1]; float sr_re = srcsr[2 * n], sr_im = srcsr[2 * n + 1]; float fl_mag = hypotf(fl_re, fl_im); float fr_mag = hypotf(fr_re, fr_im); float fl_phase = atan2f(fl_im, fl_re); float fr_phase = atan2f(fr_im, fr_re); float sl_mag = hypotf(sl_re, sl_im); float sr_mag = hypotf(sr_re, sr_im); float sl_phase = atan2f(sl_im, sl_re); float sr_phase = atan2f(sr_im, sr_re); float phase_difl = fabsf(fl_phase - sl_phase); float phase_difr = fabsf(fr_phase - sr_phase); float magl_sum = fl_mag + sl_mag; float magr_sum = fr_mag + sr_mag; float mag_difl = magl_sum < MIN_MAG_SUM ? FFDIFFSIGN(fl_mag, sl_mag) : (fl_mag - sl_mag) / magl_sum; float mag_difr = magr_sum < MIN_MAG_SUM ? FFDIFFSIGN(fr_mag, sr_mag) : (fr_mag - sr_mag) / magr_sum; float mag_totall = hypotf(fl_mag, sl_mag); float mag_totalr = hypotf(fr_mag, sr_mag); float bl_phase = atan2f(fl_im + sl_im, fl_re + sl_re); float br_phase = atan2f(fr_im + sr_im, fr_re + sr_re); float xl, yl; float xr, yr; if (phase_difl > M_PIf) phase_difl = 2.f * M_PIf - phase_difl; if (phase_difr > M_PIf) phase_difr = 2.f * M_PIf - phase_difr; stereo_position(mag_difl, phase_difl, &xl, &yl); stereo_position(mag_difr, phase_difr, &xr, &yr); s->upmix_5_1(ctx, c_re, c_im, lfe_re, lfe_im, mag_totall, mag_totalr, fl_phase, fr_phase, bl_phase, br_phase, sl_phase, sr_phase, xl, yl, xr, yr, n); } } static void filter_5_1_back(AVFilterContext *ctx) { AudioSurroundContext *s = ctx->priv; const int rdft_size = s->rdft_size; float *srcl, *srcr, *srcc, *srclfe, *srcbl, *srcbr; int n; srcl = (float *)s->input->extended_data[0]; srcr = (float *)s->input->extended_data[1]; srcc = (float *)s->input->extended_data[2]; srclfe = (float *)s->input->extended_data[3]; srcbl = (float *)s->input->extended_data[4]; srcbr = (float *)s->input->extended_data[5]; for (n = 0; n < rdft_size; n++) { float fl_re = srcl[2 * n], fr_re = srcr[2 * n]; float fl_im = srcl[2 * n + 1], fr_im = srcr[2 * n + 1]; float c_re = srcc[2 * n], c_im = srcc[2 * n + 1]; float lfe_re = srclfe[2 * n], lfe_im = srclfe[2 * n + 1]; float bl_re = srcbl[2 * n], bl_im = srcbl[2 * n + 1]; float br_re = srcbr[2 * n], br_im = srcbr[2 * n + 1]; float fl_mag = hypotf(fl_re, fl_im); float fr_mag = hypotf(fr_re, fr_im); float fl_phase = atan2f(fl_im, fl_re); float fr_phase = atan2f(fr_im, fr_re); float bl_mag = hypotf(bl_re, bl_im); float br_mag = hypotf(br_re, br_im); float bl_phase = atan2f(bl_im, bl_re); float br_phase = atan2f(br_im, br_re); float phase_difl = fabsf(fl_phase - bl_phase); float phase_difr = fabsf(fr_phase - br_phase); float magl_sum = fl_mag + bl_mag; float magr_sum = fr_mag + br_mag; float mag_difl = magl_sum < MIN_MAG_SUM ? FFDIFFSIGN(fl_mag, bl_mag) : (fl_mag - bl_mag) / magl_sum; float mag_difr = magr_sum < MIN_MAG_SUM ? FFDIFFSIGN(fr_mag, br_mag) : (fr_mag - br_mag) / magr_sum; float mag_totall = hypotf(fl_mag, bl_mag); float mag_totalr = hypotf(fr_mag, br_mag); float sl_phase = atan2f(fl_im + bl_im, fl_re + bl_re); float sr_phase = atan2f(fr_im + br_im, fr_re + br_re); float xl, yl; float xr, yr; if (phase_difl > M_PIf) phase_difl = 2.f * M_PIf - phase_difl; if (phase_difr > M_PIf) phase_difr = 2.f * M_PIf - phase_difr; stereo_position(mag_difl, phase_difl, &xl, &yl); stereo_position(mag_difr, phase_difr, &xr, &yr); s->upmix_5_1(ctx, c_re, c_im, lfe_re, lfe_im, mag_totall, mag_totalr, fl_phase, fr_phase, bl_phase, br_phase, sl_phase, sr_phase, xl, yl, xr, yr, n); } } static void allchannels_spread(AVFilterContext *ctx) { AudioSurroundContext *s = ctx->priv; if (s->all_x >= 0.f) for (int n = 0; n < SC_NB; n++) s->f_x[n] = s->all_x; s->all_x = -1.f; if (s->all_y >= 0.f) for (int n = 0; n < SC_NB; n++) s->f_y[n] = s->all_y; s->all_y = -1.f; } static av_cold int init(AVFilterContext *ctx) { AudioSurroundContext *s = ctx->priv; int64_t in_channel_layout, out_channel_layout; char in_name[128], out_name[128]; float overlap; if (s->lowcutf >= s->highcutf) { av_log(ctx, AV_LOG_ERROR, "Low cut-off '%d' should be less than high cut-off '%d'.\n", s->lowcutf, s->highcutf); return AVERROR(EINVAL); } in_channel_layout = s->in_ch_layout.order == AV_CHANNEL_ORDER_NATIVE ? s->in_ch_layout.u.mask : 0; out_channel_layout = s->out_ch_layout.order == AV_CHANNEL_ORDER_NATIVE ? s->out_ch_layout.u.mask : 0; s->create_lfe = av_channel_layout_index_from_channel(&s->out_ch_layout, AV_CHAN_LOW_FREQUENCY) >= 0; switch (out_channel_layout) { case AV_CH_LAYOUT_MONO: case AV_CH_LAYOUT_STEREO: case AV_CH_LAYOUT_2POINT1: case AV_CH_LAYOUT_2_1: case AV_CH_LAYOUT_2_2: case AV_CH_LAYOUT_SURROUND: case AV_CH_LAYOUT_3POINT1: case AV_CH_LAYOUT_QUAD: case AV_CH_LAYOUT_4POINT0: case AV_CH_LAYOUT_4POINT1: case AV_CH_LAYOUT_5POINT0: case AV_CH_LAYOUT_5POINT1: case AV_CH_LAYOUT_5POINT0_BACK: case AV_CH_LAYOUT_5POINT1_BACK: case AV_CH_LAYOUT_6POINT0: case AV_CH_LAYOUT_6POINT1: case AV_CH_LAYOUT_7POINT0: case AV_CH_LAYOUT_7POINT1: case AV_CH_LAYOUT_OCTAGONAL: break; default: goto fail; } switch (in_channel_layout) { case AV_CH_LAYOUT_STEREO: s->filter = filter_stereo; s->upmix = stereo_upmix; break; case AV_CH_LAYOUT_2POINT1: s->filter = filter_2_1; s->upmix = l2_1_upmix; break; case AV_CH_LAYOUT_SURROUND: s->filter = filter_surround; s->upmix = surround_upmix; break; case AV_CH_LAYOUT_5POINT0: s->filter = filter_5_0_side; switch (out_channel_layout) { case AV_CH_LAYOUT_7POINT1: s->upmix_5_0 = upmix_7_1_5_0_side; break; default: goto fail; } break; case AV_CH_LAYOUT_5POINT1: s->filter = filter_5_1_side; switch (out_channel_layout) { case AV_CH_LAYOUT_7POINT1: s->upmix_5_1 = upmix_7_1_5_1; break; default: goto fail; } break; case AV_CH_LAYOUT_5POINT1_BACK: s->filter = filter_5_1_back; switch (out_channel_layout) { case AV_CH_LAYOUT_7POINT1: s->upmix_5_1 = upmix_7_1_5_1; break; default: goto fail; } break; default: fail: av_channel_layout_describe(&s->out_ch_layout, out_name, sizeof(out_name)); av_channel_layout_describe(&s->in_ch_layout, in_name, sizeof(in_name)); av_log(ctx, AV_LOG_ERROR, "Unsupported upmix: '%s' -> '%s'.\n", in_name, out_name); return AVERROR(EINVAL); } s->window_func_lut = av_calloc(s->win_size, sizeof(*s->window_func_lut)); if (!s->window_func_lut) return AVERROR(ENOMEM); generate_window_func(s->window_func_lut, s->win_size, s->win_func, &overlap); if (s->overlap == 1) s->overlap = overlap; for (int i = 0; i < s->win_size; i++) s->window_func_lut[i] = sqrtf(s->window_func_lut[i] / s->win_size); s->hop_size = FFMAX(1, s->win_size * (1. - s->overlap)); { float max = 0.f, *temp_lut = av_calloc(s->win_size, sizeof(*temp_lut)); if (!temp_lut) return AVERROR(ENOMEM); for (int j = 0; j < s->win_size; j += s->hop_size) { for (int i = 0; i < s->win_size; i++) temp_lut[(i + j) % s->win_size] += s->window_func_lut[i]; } for (int i = 0; i < s->win_size; i++) max = fmaxf(temp_lut[i], max); av_freep(&temp_lut); s->win_gain = 1.f / (max * sqrtf(s->win_size)); } allchannels_spread(ctx); return 0; } static int fft_channel(AVFilterContext *ctx, AVFrame *in, int ch) { AudioSurroundContext *s = ctx->priv; float *src = (float *)s->input_in->extended_data[ch]; float *win = (float *)s->window->extended_data[ch]; const float *window_func_lut = s->window_func_lut; const int offset = s->win_size - s->hop_size; const float level_in = s->input_levels[ch]; const int win_size = s->win_size; memmove(src, &src[s->hop_size], offset * sizeof(float)); memcpy(&src[offset], in->extended_data[ch], in->nb_samples * sizeof(float)); memset(&src[offset + in->nb_samples], 0, (s->hop_size - in->nb_samples) * sizeof(float)); for (int n = 0; n < win_size; n++) win[n] = src[n] * window_func_lut[n] * level_in; s->tx_fn(s->rdft[ch], (float *)s->input->extended_data[ch], win, sizeof(float)); return 0; } static int fft_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { AVFrame *in = arg; const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs; const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs; for (int ch = start; ch < end; ch++) fft_channel(ctx, in, ch); return 0; } static int ifft_channel(AVFilterContext *ctx, AVFrame *out, int ch) { AudioSurroundContext *s = ctx->priv; const float level_out = s->output_levels[ch] * s->win_gain; const float *window_func_lut = s->window_func_lut; const int win_size = s->win_size; float *dst, *ptr; dst = (float *)s->output_out->extended_data[ch]; ptr = (float *)s->overlap_buffer->extended_data[ch]; s->itx_fn(s->irdft[ch], dst, (float *)s->output->extended_data[ch], sizeof(AVComplexFloat)); memmove(s->overlap_buffer->extended_data[ch], s->overlap_buffer->extended_data[ch] + s->hop_size * sizeof(float), s->win_size * sizeof(float)); memset(s->overlap_buffer->extended_data[ch] + s->win_size * sizeof(float), 0, s->hop_size * sizeof(float)); for (int n = 0; n < win_size; n++) ptr[n] += dst[n] * window_func_lut[n] * level_out; ptr = (float *)s->overlap_buffer->extended_data[ch]; dst = (float *)out->extended_data[ch]; memcpy(dst, ptr, s->hop_size * sizeof(float)); return 0; } static int ifft_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { AudioSurroundContext *s = ctx->priv; AVFrame *out = arg; const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs; const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs; for (int ch = start; ch < end; ch++) { if (s->upmix) s->upmix(ctx, ch); ifft_channel(ctx, out, ch); } return 0; } static int filter_frame(AVFilterLink *inlink, AVFrame *in) { AVFilterContext *ctx = inlink->dst; AVFilterLink *outlink = ctx->outputs[0]; AudioSurroundContext *s = ctx->priv; AVFrame *out; ff_filter_execute(ctx, fft_channels, in, NULL, FFMIN(inlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx))); s->filter(ctx); out = ff_get_audio_buffer(outlink, s->hop_size); if (!out) return AVERROR(ENOMEM); ff_filter_execute(ctx, ifft_channels, out, NULL, FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx))); av_frame_copy_props(out, in); out->nb_samples = in->nb_samples; av_frame_free(&in); return ff_filter_frame(outlink, out); } static int activate(AVFilterContext *ctx) { AVFilterLink *inlink = ctx->inputs[0]; AVFilterLink *outlink = ctx->outputs[0]; AudioSurroundContext *s = ctx->priv; AVFrame *in = NULL; int ret = 0, status; int64_t pts; FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); ret = ff_inlink_consume_samples(inlink, s->hop_size, s->hop_size, &in); if (ret < 0) return ret; if (ret > 0) ret = filter_frame(inlink, in); if (ret < 0) return ret; if (ff_inlink_queued_samples(inlink) >= s->hop_size) { ff_filter_set_ready(ctx, 10); return 0; } if (ff_inlink_acknowledge_status(inlink, &status, &pts)) { ff_outlink_set_status(outlink, status, pts); return 0; } FF_FILTER_FORWARD_WANTED(outlink, inlink); return FFERROR_NOT_READY; } static av_cold void uninit(AVFilterContext *ctx) { AudioSurroundContext *s = ctx->priv; av_frame_free(&s->factors); av_frame_free(&s->sfactors); av_frame_free(&s->window); av_frame_free(&s->input_in); av_frame_free(&s->input); av_frame_free(&s->output); av_frame_free(&s->output_ph); av_frame_free(&s->output_mag); av_frame_free(&s->output_out); av_frame_free(&s->overlap_buffer); for (int ch = 0; ch < s->nb_in_channels; ch++) av_tx_uninit(&s->rdft[ch]); for (int ch = 0; ch < s->nb_out_channels; ch++) av_tx_uninit(&s->irdft[ch]); av_freep(&s->input_levels); av_freep(&s->output_levels); av_freep(&s->rdft); av_freep(&s->irdft); av_freep(&s->window_func_lut); av_freep(&s->x_pos); av_freep(&s->y_pos); av_freep(&s->l_phase); av_freep(&s->r_phase); av_freep(&s->c_mag); av_freep(&s->c_phase); av_freep(&s->mag_total); av_freep(&s->lfe_mag); av_freep(&s->lfe_phase); } static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags) { AudioSurroundContext *s = ctx->priv; int ret; ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags); if (ret < 0) return ret; s->hop_size = FFMAX(1, s->win_size * (1. - s->overlap)); allchannels_spread(ctx); set_input_levels(ctx); set_output_levels(ctx); return 0; } #define OFFSET(x) offsetof(AudioSurroundContext, x) #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM #define TFLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM static const AVOption surround_options[] = { { "chl_out", "set output channel layout", OFFSET(out_ch_layout), AV_OPT_TYPE_CHLAYOUT, {.str="5.1"}, 0, 0, FLAGS }, { "chl_in", "set input channel layout", OFFSET(in_ch_layout), AV_OPT_TYPE_CHLAYOUT, {.str="stereo"},0, 0, FLAGS }, { "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "lfe", "output LFE", OFFSET(output_lfe), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, TFLAGS }, { "lfe_low", "LFE low cut off", OFFSET(lowcutf), AV_OPT_TYPE_INT, {.i64=128}, 0, 256, FLAGS }, { "lfe_high", "LFE high cut off", OFFSET(highcutf), AV_OPT_TYPE_INT, {.i64=256}, 0, 512, FLAGS }, { "lfe_mode", "set LFE channel mode", OFFSET(lfe_mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, TFLAGS, .unit = "lfe_mode" }, { "add", "just add LFE channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 1, TFLAGS, .unit = "lfe_mode" }, { "sub", "subtract LFE channel with others", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 1, TFLAGS, .unit = "lfe_mode" }, { "smooth", "set temporal smoothness strength", OFFSET(smooth), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, TFLAGS }, { "angle", "set soundfield transform angle", OFFSET(angle), AV_OPT_TYPE_FLOAT, {.dbl=90}, 0, 360, TFLAGS }, { "focus", "set soundfield transform focus", OFFSET(focus), AV_OPT_TYPE_FLOAT, {.dbl=0}, -1, 1, TFLAGS }, { "fc_in", "set front center channel input level", OFFSET(f_i[SC_FC]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "fc_out", "set front center channel output level", OFFSET(f_o[SC_FC]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "fl_in", "set front left channel input level", OFFSET(f_i[SC_FL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "fl_out", "set front left channel output level", OFFSET(f_o[SC_FL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "fr_in", "set front right channel input level", OFFSET(f_i[SC_FR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "fr_out", "set front right channel output level", OFFSET(f_o[SC_FR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "sl_in", "set side left channel input level", OFFSET(f_i[SC_SL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "sl_out", "set side left channel output level", OFFSET(f_o[SC_SL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "sr_in", "set side right channel input level", OFFSET(f_i[SC_SR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "sr_out", "set side right channel output level", OFFSET(f_o[SC_SR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "bl_in", "set back left channel input level", OFFSET(f_i[SC_BL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "bl_out", "set back left channel output level", OFFSET(f_o[SC_BL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "br_in", "set back right channel input level", OFFSET(f_i[SC_BR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "br_out", "set back right channel output level", OFFSET(f_o[SC_BR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "bc_in", "set back center channel input level", OFFSET(f_i[SC_BC]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "bc_out", "set back center channel output level", OFFSET(f_o[SC_BC]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "lfe_in", "set lfe channel input level", OFFSET(f_i[SC_LF]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "lfe_out", "set lfe channel output level", OFFSET(f_o[SC_LF]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS }, { "allx", "set all channel's x spread", OFFSET(all_x), AV_OPT_TYPE_FLOAT, {.dbl=-1}, -1, 15, TFLAGS }, { "ally", "set all channel's y spread", OFFSET(all_y), AV_OPT_TYPE_FLOAT, {.dbl=-1}, -1, 15, TFLAGS }, { "fcx", "set front center channel x spread", OFFSET(f_x[SC_FC]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "flx", "set front left channel x spread", OFFSET(f_x[SC_FL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "frx", "set front right channel x spread", OFFSET(f_x[SC_FR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "blx", "set back left channel x spread", OFFSET(f_x[SC_BL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "brx", "set back right channel x spread", OFFSET(f_x[SC_BR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "slx", "set side left channel x spread", OFFSET(f_x[SC_SL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "srx", "set side right channel x spread", OFFSET(f_x[SC_SR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "bcx", "set back center channel x spread", OFFSET(f_x[SC_BC]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "fcy", "set front center channel y spread", OFFSET(f_y[SC_FC]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "fly", "set front left channel y spread", OFFSET(f_y[SC_FL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "fry", "set front right channel y spread", OFFSET(f_y[SC_FR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "bly", "set back left channel y spread", OFFSET(f_y[SC_BL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "bry", "set back right channel y spread", OFFSET(f_y[SC_BR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "sly", "set side left channel y spread", OFFSET(f_y[SC_SL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "sry", "set side right channel y spread", OFFSET(f_y[SC_SR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "bcy", "set back center channel y spread", OFFSET(f_y[SC_BC]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS }, { "win_size", "set window size", OFFSET(win_size), AV_OPT_TYPE_INT, {.i64=4096},1024,65536,FLAGS }, WIN_FUNC_OPTION("win_func", OFFSET(win_func), FLAGS, WFUNC_HANNING), { "overlap", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, 0, 1, TFLAGS }, { NULL } }; AVFILTER_DEFINE_CLASS(surround); static const AVFilterPad inputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_input, }, }; static const AVFilterPad outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_output, }, }; const AVFilter ff_af_surround = { .name = "surround", .description = NULL_IF_CONFIG_SMALL("Apply audio surround upmix filter."), .priv_size = sizeof(AudioSurroundContext), .priv_class = &surround_class, .init = init, .uninit = uninit, .activate = activate, FILTER_INPUTS(inputs), FILTER_OUTPUTS(outputs), FILTER_QUERY_FUNC(query_formats), .flags = AVFILTER_FLAG_SLICE_THREADS, .process_command = process_command, };