| /* |
| * 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 "internal.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 (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, |
| }; |