[FFmpeg-cvslog] avfilter: add audio surround upmixer
Paul B Mahol
git at videolan.org
Thu Jun 1 22:26:35 EEST 2017
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Fri May 19 20:12:04 2017 +0200| [dc72d1dde914c16d85673e80bbe3d21967e47deb] | committer: Paul B Mahol
avfilter: add audio surround upmixer
Signed-off-by: Paul B Mahol <onemda at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=dc72d1dde914c16d85673e80bbe3d21967e47deb
---
Changelog | 1 +
doc/filters.texi | 30 ++
libavfilter/Makefile | 1 +
libavfilter/af_surround.c | 835 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 869 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 1949ec7846..3533bdc682 100644
--- a/Changelog
+++ b/Changelog
@@ -16,6 +16,7 @@ version <next>:
- spec compliant VP9 muxing support in MP4
- remove the libnut muxer/demuxer wrappers
- remove the libschroedinger encoder/decoder wrappers
+- surround audio filter
version 3.3:
- CrystalHD decoder moved to new decode API
diff --git a/doc/filters.texi b/doc/filters.texi
index 107fe61447..51fb6cdcee 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -3792,6 +3792,36 @@ channels. Default is 0.3.
Set level of input signal of original channel. Default is 0.8.
@end table
+ at section surround
+Apply audio surround upmix filter.
+
+This filter allows to produce multichannel output from stereo audio stream.
+
+The filter accepts the following options:
+
+ at table @option
+ at item chl_out
+Set output channel layout. By default, this is @var{5.1}.
+
+See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
+for the required syntax.
+
+ at item level_in
+Set input volume level. By default, this is @var{1}.
+
+ at item level_out
+Set output volume level. By default, this is @var{1}.
+
+ at item lfe
+Enable LFE channel output if output channel layout has it. By default, this is enabled.
+
+ at item lfe_low
+Set LFE low cut off frequency. By default, this is @var{128} Hz.
+
+ at item lfe_high
+Set LFE high cut off frequency. By default, this is @var{256} Hz.
+ at 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 434a989244..c88dfb3264 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 0000000000..c7d86a50b8
--- /dev/null
+++ b/libavfilter/af_surround.c
@@ -0,0 +1,835 @@
+/*
+ * 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 inline void get_lfe(int output_lfe, int n, float lowcut, float highcut,
+ float *lfe_mag, float *mag_total)
+{
+ if (output_lfe && n < highcut) {
+ *lfe_mag = n < lowcut ? 1.f : .5f*(1.f+cosf(M_PI*(lowcut-n)/(lowcut-highcut)));
+ *lfe_mag *= *mag_total;
+ *mag_total -= *lfe_mag;
+ } else {
+ *lfe_mag = 0.f;
+ }
+}
+
+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)
+{
+ AudioSurroundContext *s = ctx->priv;
+ float mag, *dst;
+
+ dst = (float *)s->output->extended_data[0];
+
+ mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * 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)
+{
+ AudioSurroundContext *s = ctx->priv;
+ float l_mag, r_mag, *dstl, *dstr;
+
+ dstl = (float *)s->output->extended_data[0];
+ dstr = (float *)s->output->extended_data[1];
+
+ l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
+ r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * 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];
+
+ get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total);
+
+ l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
+ r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * 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.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
+ l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
+ r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * 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];
+
+ get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total);
+
+ c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
+ l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
+ r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * 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.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
+ b_mag = sqrtf(1.f - fabsf(x)) * ((1.f - y) * .5f) * mag_total;
+ l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
+ r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * 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];
+
+ get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total);
+
+ dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
+ dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
+
+ c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
+ b_mag = sqrtf(1.f - fabsf(x)) * ((1.f - y) * .5f) * mag_total;
+ l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
+ r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * 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.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
+ l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
+ r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
+ ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
+ rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * 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];
+
+ get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total);
+
+ c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
+ l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
+ r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
+ ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
+ rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * 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.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
+ l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
+ r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
+ lb_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
+ rb_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
+ ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - fabsf(y)) * mag_total;
+ rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - 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];
+
+ get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total);
+
+ c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
+ l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
+ r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
+ lb_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
+ rb_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
+ ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - fabsf(y)) * mag_total;
+ rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - 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 f8cd193dbe..534c340fa9 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);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 55cf5d0fc2..11cfe514b8 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 6
-#define LIBAVFILTER_VERSION_MINOR 90
+#define LIBAVFILTER_VERSION_MINOR 91
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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