[FFmpeg-devel] avfilter: add audio surround upmixer

Message ID	20170525144546.15459-1-onemda@gmail.com
State	New
Headers	show Delivered-To: ffmpegpatchwork@gmail.com Received-SPF: pass (google.com: domain of ffmpeg-devel-bounces@ffmpeg.org designates 79.124.17.100 as permitted sender) client-ip=79.124.17.100; From: Paul B Mahol <onemda@gmail.com> To: ffmpeg-devel@ffmpeg.org Date: Thu, 25 May 2017 16:45:46 +0200 Message-Id: <20170525144546.15459-1-onemda@gmail.com> Subject: [FFmpeg-devel] [PATCH] avfilter: add audio surround upmixer Precedence: list Reply-To: FFmpeg development discussions and patches <ffmpeg-devel@ffmpeg.org> MIME-Version: 1.0 Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: base64 Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" <ffmpeg-devel-bounces@ffmpeg.org>

diff --git a/doc/filters.texi b/doc/filters.texi index a0ab2fb..19e4bfe 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -3792,6 +3792,33 @@ channels. Default is 0.3. Set level of input signal of original channel. Default is 0.8. @end table +@section surround +Apply audio surround upmix filter. + +This filter allows to produce multichannel output from stereo audio stream. + +The filter accepts the following options: + +@table @option +@item chl_out +Set output channel layout. By default is @var{5.1}. + +@item level_in +Set input volume level. By default is @var{1}. + +@item level_out +Set output volume level. By default is @var{1}. + +@item lfe +Enable LFE channel output if output channel layout have it. By default is enabled. + +@item lfe_low +Set LFE low cut off frequency. By default is @var{128} Hz. + +@item lfe_high +Set LFE high cut off frequency. By default is @var{256} Hz. +@end table + @section treble Boost or cut treble (upper) frequencies of the audio using a two-pole diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 434a989..c88dfb3 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -108,6 +108,7 @@ OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o OBJS-$(CONFIG_SOFALIZER_FILTER) += af_sofalizer.o OBJS-$(CONFIG_STEREOTOOLS_FILTER) += af_stereotools.o OBJS-$(CONFIG_STEREOWIDEN_FILTER) += af_stereowiden.o +OBJS-$(CONFIG_SURROUND_FILTER) += af_surround.o OBJS-$(CONFIG_TREBLE_FILTER) += af_biquads.o OBJS-$(CONFIG_TREMOLO_FILTER) += af_tremolo.o OBJS-$(CONFIG_VIBRATO_FILTER) += af_vibrato.o generate_wave_table.o diff --git a/libavfilter/af_surround.c b/libavfilter/af_surround.c new file mode 100644 index 0000000..310b8d7 --- /dev/null +++ b/libavfilter/af_surround.c @@ -0,0 +1,853 @@ +/* + * 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/audio_fifo.h" +#include "libavutil/channel_layout.h" +#include "libavutil/opt.h" +#include "libavcodec/avfft.h" +#include "avfilter.h" +#include "audio.h" +#include "formats.h" + +typedef struct AudioSurroundContext { + const AVClass *class; + + char *out_channel_layout_str; + float level_in; + float level_out; + int output_lfe; + int lowcutf; + int highcutf; + + float lowcut; + float highcut; + + uint64_t out_channel_layout; + int nb_in_channels; + int nb_out_channels; + + AVFrame *input; + AVFrame *output; + AVFrame *overlap_buffer; + + int buf_size; + int hop_size; + AVAudioFifo *fifo; + RDFTContext **rdft, **irdft; + float *window_func_lut; + + int64_t pts; + + void (*upmix)(AVFilterContext *ctx, + float l_phase, + float r_phase, + float c_phase, + float mag_total, + float x, float y, + 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_channel_layout); + if (ret) + return ret; + + ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts); + if (ret) + return ret; + + layouts = NULL; + ret = ff_add_channel_layout(&layouts, AV_CH_LAYOUT_STEREO); + if (ret) + return ret; + + ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts); + if (ret) + return ret; + + formats = ff_all_samplerates(); + if (!formats) + return AVERROR(ENOMEM); + return ff_set_common_samplerates(ctx, formats); +} + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + AudioSurroundContext *s = ctx->priv; + int ch; + + s->rdft = av_calloc(inlink->channels, sizeof(*s->rdft)); + if (!s->rdft) + return AVERROR(ENOMEM); + + for (ch = 0; ch < inlink->channels; ch++) { + s->rdft[ch] = av_rdft_init(ff_log2(s->buf_size), DFT_R2C); + if (!s->rdft[ch]) + return AVERROR(ENOMEM); + } + s->nb_in_channels = inlink->channels; + + s->input = ff_get_audio_buffer(inlink, s->buf_size * 2); + if (!s->input) + return AVERROR(ENOMEM); + + s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->buf_size); + if (!s->fifo) + return AVERROR(ENOMEM); + + s->lowcut = 1.f * s->lowcutf / (inlink->sample_rate * 0.5) * (s->buf_size / 2); + s->highcut = 1.f * s->highcutf / (inlink->sample_rate * 0.5) * (s->buf_size / 2); + + return 0; +} + +static int config_output(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + AudioSurroundContext *s = ctx->priv; + int ch; + + s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft)); + if (!s->irdft) + return AVERROR(ENOMEM); + + for (ch = 0; ch < outlink->channels; ch++) { + s->irdft[ch] = av_rdft_init(ff_log2(s->buf_size), IDFT_C2R); + if (!s->irdft[ch]) + return AVERROR(ENOMEM); + } + s->nb_out_channels = outlink->channels; + + s->output = ff_get_audio_buffer(outlink, s->buf_size * 2); + s->overlap_buffer = ff_get_audio_buffer(outlink, s->buf_size * 2); + if (!s->overlap_buffer || !s->output) + return AVERROR(ENOMEM); + + return 0; +} + +static void stereo_position(float a, float p, float *x, float *y) +{ + *x = av_clipf(a+FFMAX(0, sinf(p-M_PI_2))*FFDIFFSIGN(a,0), -1, 1); + *y = av_clipf(cosf(a*M_PI_2+M_PI)*cosf(M_PI_2-p/M_PI)*M_LN10+1, -1, 1); +} + +static void upmix_1_0(AVFilterContext *ctx, + float l_phase, + float r_phase, + float c_phase, + float mag_total, + float x, float y, + int n) +{ + float mag, *dst; + AudioSurroundContext *s = ctx->priv; + + dst = (float *)s->output->extended_data[0]; + + mag = sqrtf(1 - fabsf(x)) * ((y + 1) * .5) * mag_total; + + dst[2 * n ] = mag * cosf(c_phase); + dst[2 * n + 1] = mag * sinf(c_phase); +} + +static void upmix_stereo(AVFilterContext *ctx, + float l_phase, + float r_phase, + float c_phase, + float mag_total, + float x, float y, + int n) +{ + float l_mag, r_mag, *dstl, *dstr; + AudioSurroundContext *s = ctx->priv; + + dstl = (float *)s->output->extended_data[0]; + dstr = (float *)s->output->extended_data[1]; + + l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total; + r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total; + + dstl[2 * n ] = l_mag * cosf(l_phase); + dstl[2 * n + 1] = l_mag * sinf(l_phase); + + dstr[2 * n ] = r_mag * cosf(r_phase); + dstr[2 * n + 1] = r_mag * sinf(r_phase); +} + +static void upmix_2_1(AVFilterContext *ctx, + float l_phase, + float r_phase, + float c_phase, + float mag_total, + float x, float y, + int n) +{ + AudioSurroundContext *s = ctx->priv; + float lfe_mag, l_mag, r_mag, *dstl, *dstr, *dstlfe; + + dstl = (float *)s->output->extended_data[0]; + dstr = (float *)s->output->extended_data[1]; + dstlfe = (float *)s->output->extended_data[2]; + + if (s->output_lfe && n < s->highcut) { + lfe_mag = n < s->lowcut ? 1.f : .5f*(1.f+cosf(M_PI*(s->lowcut-n)/(s->lowcut-s->highcut))); + lfe_mag *= mag_total; + mag_total -= lfe_mag; + } else { + lfe_mag = 0.f; + } + + l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total; + r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total; + + dstl[2 * n ] = l_mag * cosf(l_phase); + dstl[2 * n + 1] = l_mag * sinf(l_phase); + + dstr[2 * n ] = r_mag * cosf(r_phase); + dstr[2 * n + 1] = r_mag * sinf(r_phase); + + dstlfe[2 * n ] = lfe_mag * cosf(c_phase); + dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase); +} + +static void upmix_3_0(AVFilterContext *ctx, + float l_phase, + float r_phase, + float c_phase, + float mag_total, + float x, float y, + int n) +{ + AudioSurroundContext *s = ctx->priv; + float l_mag, r_mag, c_mag, *dstc, *dstl, *dstr; + + dstl = (float *)s->output->extended_data[0]; + dstr = (float *)s->output->extended_data[1]; + dstc = (float *)s->output->extended_data[2]; + + c_mag = sqrtf(1 - fabsf(x)) * ((y + 1) * .5) * mag_total; + l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total; + r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total; + + dstl[2 * n ] = l_mag * cosf(l_phase); + dstl[2 * n + 1] = l_mag * sinf(l_phase); + + dstr[2 * n ] = r_mag * cosf(r_phase); + dstr[2 * n + 1] = r_mag * sinf(r_phase); + + dstc[2 * n ] = c_mag * cosf(c_phase); + dstc[2 * n + 1] = c_mag * sinf(c_phase); +} + +static void upmix_3_1(AVFilterContext *ctx, + float l_phase, + float r_phase, + float c_phase, + float mag_total, + float x, float y, + int n) +{ + AudioSurroundContext *s = ctx->priv; + float lfe_mag, l_mag, r_mag, c_mag, *dstc, *dstl, *dstr, *dstlfe; + + 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]; + + if (s->output_lfe && n < s->highcut) { + lfe_mag = n < s->lowcut ? 1.f : .5f*(1.f+cosf(M_PI*(s->lowcut-n)/(s->lowcut-s->highcut))); + lfe_mag *= mag_total; + mag_total -= lfe_mag; + } else { + lfe_mag = 0.f; + } + + c_mag = sqrtf(1 - fabsf(x)) * ((y + 1) * .5) * mag_total; + l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total; + r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total; + + dstl[2 * n ] = l_mag * cosf(l_phase); + dstl[2 * n + 1] = l_mag * sinf(l_phase); + + dstr[2 * n ] = r_mag * cosf(r_phase); + dstr[2 * n + 1] = r_mag * sinf(r_phase); + + dstc[2 * n ] = c_mag * cosf(c_phase); + dstc[2 * n + 1] = c_mag * sinf(c_phase); + + dstlfe[2 * n ] = lfe_mag * cosf(c_phase); + dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase); +} + +static void upmix_4_0(AVFilterContext *ctx, + float l_phase, + float r_phase, + float c_phase, + float mag_total, + float x, float y, + int n) +{ + float b_mag, l_mag, r_mag, c_mag, *dstc, *dstl, *dstr, *dstb; + 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]; + dstb = (float *)s->output->extended_data[3]; + + c_mag = sqrtf(1 - fabsf(x)) * ((y + 1) * .5) * mag_total; + b_mag = sqrtf(1 - fabsf(x)) * ((1 - y) * .5) * mag_total; + l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total; + r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total; + + dstl[2 * n ] = l_mag * cosf(l_phase); + dstl[2 * n + 1] = l_mag * sinf(l_phase); + + dstr[2 * n ] = r_mag * cosf(r_phase); + dstr[2 * n + 1] = r_mag * sinf(r_phase); + + dstc[2 * n ] = c_mag * cosf(c_phase); + dstc[2 * n + 1] = c_mag * sinf(c_phase); + + dstb[2 * n ] = b_mag * cosf(c_phase); + dstb[2 * n + 1] = b_mag * sinf(c_phase); +} + +static void upmix_4_1(AVFilterContext *ctx, + float l_phase, + float r_phase, + float c_phase, + float mag_total, + float x, float y, + int n) +{ + float lfe_mag, b_mag, l_mag, r_mag, c_mag, *dstc, *dstl, *dstr, *dstb, *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]; + dstb = (float *)s->output->extended_data[4]; + + if (s->output_lfe && n < s->highcut) { + lfe_mag = n < s->lowcut ? 1.f : .5f*(1.f+cosf(M_PI*(s->lowcut-n)/(s->lowcut-s->highcut))); + lfe_mag *= mag_total; + mag_total -= lfe_mag; + } else { + lfe_mag = 0.f; + } + + dstlfe[2 * n ] = lfe_mag * cosf(c_phase); + dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase); + + c_mag = sqrtf(1 - fabsf(x)) * ((y + 1) * .5) * mag_total; + b_mag = sqrtf(1 - fabsf(x)) * ((1 - y) * .5) * mag_total; + l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total; + r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total; + + dstl[2 * n ] = l_mag * cosf(l_phase); + dstl[2 * n + 1] = l_mag * sinf(l_phase); + + dstr[2 * n ] = r_mag * cosf(r_phase); + dstr[2 * n + 1] = r_mag * sinf(r_phase); + + dstc[2 * n ] = c_mag * cosf(c_phase); + dstc[2 * n + 1] = c_mag * sinf(c_phase); + + dstb[2 * n ] = b_mag * cosf(c_phase); + dstb[2 * n + 1] = b_mag * sinf(c_phase); +} + +static void upmix_5_0_back(AVFilterContext *ctx, + float l_phase, + float r_phase, + float c_phase, + float mag_total, + float x, float y, + int n) +{ + float l_mag, r_mag, ls_mag, rs_mag, c_mag, *dstc, *dstl, *dstr, *dstls, *dstrs; + 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]; + dstls = (float *)s->output->extended_data[3]; + dstrs = (float *)s->output->extended_data[4]; + + c_mag = sqrtf(1 - fabsf(x)) * ((y + 1) * .5) * mag_total; + l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total; + r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total; + ls_mag = sqrtf(0.5 * ( x + 1)) * (1 - ((y + 1) * .5)) * mag_total; + rs_mag = sqrtf(0.5 * (-x + 1)) * (1 - ((y + 1) * .5)) * mag_total; + + dstl[2 * n ] = l_mag * cosf(l_phase); + dstl[2 * n + 1] = l_mag * sinf(l_phase); + + dstr[2 * n ] = r_mag * cosf(r_phase); + dstr[2 * n + 1] = r_mag * sinf(r_phase); + + dstc[2 * n ] = c_mag * cosf(c_phase); + dstc[2 * n + 1] = c_mag * sinf(c_phase); + + dstls[2 * n ] = ls_mag * cosf(l_phase); + dstls[2 * n + 1] = ls_mag * sinf(l_phase); + + dstrs[2 * n ] = rs_mag * cosf(r_phase); + dstrs[2 * n + 1] = rs_mag * sinf(r_phase); +} + +static void upmix_5_1_back(AVFilterContext *ctx, + float l_phase, + float r_phase, + float c_phase, + float mag_total, + float x, float y, + int n) +{ + float lfe_mag, l_mag, r_mag, ls_mag, rs_mag, c_mag, *dstc, *dstl, *dstr, *dstls, *dstrs, *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]; + dstls = (float *)s->output->extended_data[4]; + dstrs = (float *)s->output->extended_data[5]; + + if (s->output_lfe && n < s->highcut) { + lfe_mag = n < s->lowcut ? 1.f : .5f*(1.f+cosf(M_PI*(s->lowcut-n)/(s->lowcut-s->highcut))); + lfe_mag *= mag_total; + mag_total -= lfe_mag; + } else { + lfe_mag = 0.f; + } + + c_mag = sqrtf(1 - fabsf(x)) * ((y + 1) * .5) * mag_total; + l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total; + r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total; + ls_mag = sqrtf(0.5 * ( x + 1)) * (1 - ((y + 1) * .5)) * mag_total; + rs_mag = sqrtf(0.5 * (-x + 1)) * (1 - ((y + 1) * .5)) * mag_total; + + dstl[2 * n ] = l_mag * cosf(l_phase); + dstl[2 * n + 1] = l_mag * sinf(l_phase); + + dstr[2 * n ] = r_mag * cosf(r_phase); + dstr[2 * n + 1] = r_mag * sinf(r_phase); + + dstc[2 * n ] = c_mag * cosf(c_phase); + dstc[2 * n + 1] = c_mag * sinf(c_phase); + + dstlfe[2 * n ] = lfe_mag * cosf(c_phase); + dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase); + + dstls[2 * n ] = ls_mag * cosf(l_phase); + dstls[2 * n + 1] = ls_mag * sinf(l_phase); + + dstrs[2 * n ] = rs_mag * cosf(r_phase); + dstrs[2 * n + 1] = rs_mag * sinf(r_phase); +} + +static void upmix_7_0(AVFilterContext *ctx, + float l_phase, + float r_phase, + float c_phase, + float mag_total, + float x, float y, + int n) +{ + float l_mag, r_mag, ls_mag, rs_mag, c_mag, lb_mag, rb_mag; + float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb; + 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]; + dstlb = (float *)s->output->extended_data[3]; + dstrb = (float *)s->output->extended_data[4]; + dstls = (float *)s->output->extended_data[5]; + dstrs = (float *)s->output->extended_data[6]; + + c_mag = sqrtf(1 - fabsf(x)) * ((y + 1) * .5) * mag_total; + l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total; + r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total; + lb_mag = sqrtf(0.5 * ( x + 1)) * (1 - ((y + 1) * .5)) * mag_total; + rb_mag = sqrtf(0.5 * (-x + 1)) * (1 - ((y + 1) * .5)) * mag_total; + ls_mag = sqrtf(0.5 * ( x + 1)) * (1 - fabsf(y)) * mag_total; + rs_mag = sqrtf(0.5 * (-x + 1)) * (1 - fabsf(y)) * mag_total; + + dstl[2 * n ] = l_mag * cosf(l_phase); + dstl[2 * n + 1] = l_mag * sinf(l_phase); + + dstr[2 * n ] = r_mag * cosf(r_phase); + dstr[2 * n + 1] = r_mag * sinf(r_phase); + + dstc[2 * n ] = c_mag * cosf(c_phase); + dstc[2 * n + 1] = c_mag * sinf(c_phase); + + dstlb[2 * n ] = lb_mag * cosf(l_phase); + dstlb[2 * n + 1] = lb_mag * sinf(l_phase); + + dstrb[2 * n ] = rb_mag * cosf(r_phase); + dstrb[2 * n + 1] = rb_mag * sinf(r_phase); + + dstls[2 * n ] = ls_mag * cosf(l_phase); + dstls[2 * n + 1] = ls_mag * sinf(l_phase); + + dstrs[2 * n ] = rs_mag * cosf(r_phase); + dstrs[2 * n + 1] = rs_mag * sinf(r_phase); +} + +static void upmix_7_1(AVFilterContext *ctx, + float l_phase, + float r_phase, + float c_phase, + float mag_total, + float x, float y, + int n) +{ + float lfe_mag, l_mag, r_mag, ls_mag, rs_mag, c_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]; + + if (s->output_lfe && n < s->highcut) { + lfe_mag = n < s->lowcut ? 1.f : .5f*(1.f+cosf(M_PI*(s->lowcut-n)/(s->lowcut-s->highcut))); + lfe_mag *= mag_total; + mag_total -= lfe_mag; + } else { + lfe_mag = 0.f; + } + + c_mag = sqrtf(1 - fabsf(x)) * ((y + 1) * .5) * mag_total; + l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total; + r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total; + lb_mag = sqrtf(0.5 * ( x + 1)) * (1 - ((y + 1) * .5)) * mag_total; + rb_mag = sqrtf(0.5 * (-x + 1)) * (1 - ((y + 1) * .5)) * mag_total; + ls_mag = sqrtf(0.5 * ( x + 1)) * (1 - fabsf(y)) * mag_total; + rs_mag = sqrtf(0.5 * (-x + 1)) * (1 - fabsf(y)) * mag_total; + + dstl[2 * n ] = l_mag * cosf(l_phase); + dstl[2 * n + 1] = l_mag * sinf(l_phase); + + dstr[2 * n ] = r_mag * cosf(r_phase); + dstr[2 * n + 1] = r_mag * sinf(r_phase); + + dstc[2 * n ] = c_mag * cosf(c_phase); + dstc[2 * n + 1] = c_mag * sinf(c_phase); + + dstlfe[2 * n ] = lfe_mag * cosf(c_phase); + dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase); + + dstlb[2 * n ] = lb_mag * cosf(l_phase); + dstlb[2 * n + 1] = lb_mag * sinf(l_phase); + + dstrb[2 * n ] = rb_mag * cosf(r_phase); + dstrb[2 * n + 1] = rb_mag * sinf(r_phase); + + dstls[2 * n ] = ls_mag * cosf(l_phase); + dstls[2 * n + 1] = ls_mag * sinf(l_phase); + + dstrs[2 * n ] = rs_mag * cosf(r_phase); + dstrs[2 * n + 1] = rs_mag * sinf(r_phase); +} + +static int init(AVFilterContext *ctx) +{ + AudioSurroundContext *s = ctx->priv; + float overlap; + int i; + + if (!(s->out_channel_layout = av_get_channel_layout(s->out_channel_layout_str))) { + av_log(ctx, AV_LOG_ERROR, "Error parsing channel layout '%s'.\n", + s->out_channel_layout_str); + return AVERROR(EINVAL); + } + + 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); + } + + switch (s->out_channel_layout) { + case AV_CH_LAYOUT_MONO: + s->upmix = upmix_1_0; + break; + case AV_CH_LAYOUT_STEREO: + s->upmix = upmix_stereo; + break; + case AV_CH_LAYOUT_2POINT1: + s->upmix = upmix_2_1; + break; + case AV_CH_LAYOUT_SURROUND: + s->upmix = upmix_3_0; + break; + case AV_CH_LAYOUT_3POINT1: + s->upmix = upmix_3_1; + break; + case AV_CH_LAYOUT_4POINT0: + s->upmix = upmix_4_0; + break; + case AV_CH_LAYOUT_4POINT1: + s->upmix = upmix_4_1; + break; + case AV_CH_LAYOUT_5POINT0_BACK: + s->upmix = upmix_5_0_back; + break; + case AV_CH_LAYOUT_5POINT1_BACK: + s->upmix = upmix_5_1_back; + break; + case AV_CH_LAYOUT_7POINT0: + s->upmix = upmix_7_0; + break; + case AV_CH_LAYOUT_7POINT1: + s->upmix = upmix_7_1; + break; + default: + av_log(ctx, AV_LOG_ERROR, "Unsupported output channel layout '%s'.\n", + s->out_channel_layout_str); + return AVERROR(EINVAL); + } + + s->buf_size = 4096; + s->pts = AV_NOPTS_VALUE; + + s->window_func_lut = av_calloc(s->buf_size, sizeof(*s->window_func_lut)); + if (!s->window_func_lut) + return AVERROR(ENOMEM); + + for (i = 0; i < s->buf_size; i++) + s->window_func_lut[i] = sqrtf(0.5 * (1 - cosf(2 * M_PI * i / s->buf_size)) / s->buf_size); + overlap = .5; + s->hop_size = s->buf_size * (1. - overlap); + + return 0; +} + +static int fft_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) +{ + AudioSurroundContext *s = ctx->priv; + const float level_in = s->level_in; + float *dst; + int n; + + memset(s->input->extended_data[ch] + s->buf_size * sizeof(float), 0, s->buf_size * sizeof(float)); + + dst = (float *)s->input->extended_data[ch]; + for (n = 0; n < s->buf_size; n++) { + dst[n] *= s->window_func_lut[n] * level_in; + } + + av_rdft_calc(s->rdft[ch], (float *)s->input->extended_data[ch]); + + return 0; +} + +static int ifft_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) +{ + AudioSurroundContext *s = ctx->priv; + const float level_out = s->level_out; + AVFrame *out = arg; + float *dst, *ptr; + int n; + + av_rdft_calc(s->irdft[ch], (float *)s->output->extended_data[ch]); + + dst = (float *)s->output->extended_data[ch]; + ptr = (float *)s->overlap_buffer->extended_data[ch]; + + memmove(s->overlap_buffer->extended_data[ch], + s->overlap_buffer->extended_data[ch] + s->hop_size * sizeof(float), + s->buf_size * sizeof(float)); + memset(s->overlap_buffer->extended_data[ch] + s->buf_size * sizeof(float), + 0, s->hop_size * sizeof(float)); + + for (n = 0; n < s->buf_size; n++) { + ptr[n] += dst[n] * s->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 filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + AVFilterLink *outlink = ctx->outputs[0]; + AudioSurroundContext *s = ctx->priv; + + av_audio_fifo_write(s->fifo, (void **)in->extended_data, + in->nb_samples); + + if (s->pts == AV_NOPTS_VALUE) + s->pts = in->pts; + + av_frame_free(&in); + + while (av_audio_fifo_size(s->fifo) >= s->buf_size) { + float *srcl, *srcr; + AVFrame *out; + int n, ret; + + ret = av_audio_fifo_peek(s->fifo, (void **)s->input->extended_data, s->buf_size); + if (ret < 0) + return ret; + + ctx->internal->execute(ctx, fft_channel, NULL, NULL, inlink->channels); + + srcl = (float *)s->input->extended_data[0]; + srcr = (float *)s->input->extended_data[1]; + + for (n = 0; n < s->buf_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 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_dif = (l_mag - r_mag) / (l_mag + r_mag); + float mag_total = hypotf(l_mag, r_mag); + float x, y; + + if (phase_dif > M_PI) + phase_dif = 2 * M_PI - phase_dif; + + stereo_position(mag_dif, phase_dif, &x, &y); + + s->upmix(ctx, l_phase, r_phase, c_phase, mag_total, x, y, n); + } + + out = ff_get_audio_buffer(outlink, s->hop_size); + if (!out) + return AVERROR(ENOMEM); + + ctx->internal->execute(ctx, ifft_channel, out, NULL, outlink->channels); + + out->pts = s->pts; + if (s->pts != AV_NOPTS_VALUE) + s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); + av_audio_fifo_drain(s->fifo, s->hop_size); + ret = ff_filter_frame(outlink, out); + if (ret < 0) + return ret; + } + + return 0; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + AudioSurroundContext *s = ctx->priv; + int ch; + + av_frame_free(&s->input); + av_frame_free(&s->output); + av_frame_free(&s->overlap_buffer); + + for (ch = 0; ch < s->nb_in_channels; ch++) { + av_rdft_end(s->rdft[ch]); + } + for (ch = 0; ch < s->nb_out_channels; ch++) { + av_rdft_end(s->irdft[ch]); + } + av_freep(&s->rdft); + av_freep(&s->irdft); + av_audio_fifo_free(s->fifo); + av_freep(&s->window_func_lut); +} + +#define OFFSET(x) offsetof(AudioSurroundContext, x) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption surround_options[] = { + { "chl_out", "set output channel layout", OFFSET(out_channel_layout_str), AV_OPT_TYPE_STRING, {.str="5.1"}, 0, 0, FLAGS }, + { "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS }, + { "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS }, + { "lfe", "output LFE", OFFSET(output_lfe), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, FLAGS }, + { "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 }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(surround); + +static const AVFilterPad inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + .config_props = config_input, + }, + { NULL } +}; + +static const AVFilterPad outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_output, + }, + { NULL } +}; + +AVFilter ff_af_surround = { + .name = "surround", + .description = NULL_IF_CONFIG_SMALL("Apply audio surround upmix filter."), + .query_formats = query_formats, + .priv_size = sizeof(AudioSurroundContext), + .priv_class = &surround_class, + .init = init, + .uninit = uninit, + .inputs = inputs, + .outputs = outputs, + .flags = AVFILTER_FLAG_SLICE_THREADS, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index f8cd193..534c340 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -121,6 +121,7 @@ static void register_all(void) REGISTER_FILTER(SOFALIZER, sofalizer, af); REGISTER_FILTER(STEREOTOOLS, stereotools, af); REGISTER_FILTER(STEREOWIDEN, stereowiden, af); + REGISTER_FILTER(SURROUND, surround, af); REGISTER_FILTER(TREBLE, treble, af); REGISTER_FILTER(TREMOLO, tremolo, af); REGISTER_FILTER(VIBRATO, vibrato, af);

[FFmpeg-devel] avfilter: add audio surround upmixer

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