[FFmpeg-devel] [PATCH] avfilter: add SOFAlizer audio filter
Paul B Mahol
onemda at gmail.com
Fri Dec 11 00:03:34 CET 2015
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
Lite version of one sent to VLC mailing list with only slow but high quality mode present.
To use you need recent netCDF library, SOFA file(s), multichannel audio and headphones.
---
configure | 4 +
libavfilter/Makefile | 1 +
libavfilter/af_sofalizer.c | 932 +++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/formats.c | 11 +
libavfilter/formats.h | 3 +
6 files changed, 952 insertions(+)
create mode 100644 libavfilter/af_sofalizer.c
diff --git a/configure b/configure
index afac1bc..71b2d36 100755
--- a/configure
+++ b/configure
@@ -279,6 +279,7 @@ External library support:
--disable-lzma disable lzma [autodetect]
--enable-decklink enable Blackmagic DeckLink I/O support [no]
--enable-mmal enable decoding via MMAL [no]
+ --enable-netcdf enable NetCDF, needed for sofalizer filter [no]
--enable-nvenc enable NVIDIA NVENC support [no]
--enable-openal enable OpenAL 1.1 capture support [no]
--enable-opencl enable OpenCL code
@@ -1504,6 +1505,7 @@ EXTERNAL_LIBRARY_LIST="
libzvbi
lzma
mmal
+ netcdf
nvenc
openal
opencl
@@ -2891,6 +2893,7 @@ showfreqs_filter_deps="avcodec"
showfreqs_filter_select="fft"
showspectrum_filter_deps="avcodec"
showspectrum_filter_select="rdft"
+sofalizer_filter_deps="netcdf"
spp_filter_deps="gpl avcodec"
spp_filter_select="fft idctdsp fdctdsp me_cmp pixblockdsp"
stereo3d_filter_deps="gpl"
@@ -5502,6 +5505,7 @@ enabled mmal && { check_lib interface/mmal/mmal.h mmal_port_connect
check_lib interface/mmal/mmal.h mmal_port_connect ; }
check_lib interface/mmal/mmal.h mmal_port_connect ; } ||
die "ERROR: mmal not found"; }
+enabled netcdf && require_pkg_config netcdf netcdf.h nc_inq_libvers
enabled nvenc && { check_header nvEncodeAPI.h || die "ERROR: nvEncodeAPI.h not found."; } &&
{ check_cpp_condition nvEncodeAPI.h "NVENCAPI_MAJOR_VERSION >= 5" ||
die "ERROR: NVENC API version 4 or older is not supported"; } &&
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 8884d1d..d7a3f61 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -87,6 +87,7 @@ OBJS-$(CONFIG_SIDECHAINCOMPRESS_FILTER) += af_sidechaincompress.o
OBJS-$(CONFIG_SIDECHAINGATE_FILTER) += af_agate.o
OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o
OBJS-$(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_TREBLE_FILTER) += af_biquads.o
diff --git a/libavfilter/af_sofalizer.c b/libavfilter/af_sofalizer.c
new file mode 100644
index 0000000..52cd09c
--- /dev/null
+++ b/libavfilter/af_sofalizer.c
@@ -0,0 +1,932 @@
+/*****************************************************************************
+ * sofalizer.c : SOFAlizer filter for virtual binaural acoustics
+ *****************************************************************************
+ * Copyright (C) 2013-2015 Andreas Fuchs, Wolfgang Hrauda,
+ * Acoustics Research Institute (ARI), Vienna, Austria
+ *
+ * Authors: Andreas Fuchs <andi.fuchs.mail at gmail.com>
+ * Wolfgang Hrauda <wolfgang.hrauda at gmx.at>
+ *
+ * SOFAlizer project coordinator at ARI, main developer of SOFA:
+ * Piotr Majdak <piotr at majdak.at>
+ *
+ * This program 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.
+ *
+ * This program 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 this program; if not, write to the Free Software Foundation,
+ * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
+ *****************************************************************************/
+
+#include <math.h>
+#include <netcdf.h>
+
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "internal.h"
+#include "audio.h"
+
+typedef struct NCSofa { /* contains data of one SOFA file */
+ int ncid; /* netCDF ID of the opened SOFA file */
+ int n_samples; /* length of one impulse response (IR) */
+ int m_dim; /* number of measurement positions */
+ int *data_delay; /* broadband delay of each IR */
+ /* all measurement positions for each receiver (i.e. ear): */
+ float *sp_a; /* azimuth angles */
+ float *sp_e; /* elevation angles */
+ float *sp_r; /* radii */
+ /* data at each measurement position for each receiver: */
+ float *data_ir; /* IRs (time-domain) */
+} NCSofa;
+
+typedef struct SOFAlizerContext {
+ const AVClass *class;
+
+ char *filename; /* name of SOFA file */
+ NCSofa sofa; /* contains data of the SOFA file */
+
+ const int8_t *reorder; /* reorder in SOFA channel order */
+ int sample_rate; /* sample rate from SOFA file */
+ float *speaker_pos; /* positions of the virtual loudspekaers */
+ float gain_lfe; /* gain applied to LFE channel */
+
+ int n_conv; /* number of channels to convolute */
+
+ /* buffer variables (for convolution) */
+ float *ringbuffer_l; /* buffers input samples, length of one buffer: */
+ float *ringbuffer_r; /* no. input ch. (incl. LFE) x buffer_length */
+ int write_l; /* current write position to ringbuffer */
+ int write_r; /* current write position to ringbuffer */
+ int buffer_length; /* is: longest IR plus max. delay in all SOFA files */
+ /* then choose next power of 2 */
+
+ /* netCDF variables */
+ int *delay_l; /* broadband delay for each channel/IR to be convolved */
+ int *delay_r;
+ float *data_ir_l; /* IRs for all channels to be convolved */
+ float *data_ir_r; /* (this excludes the LFE) */
+
+ /* control variables */
+ float gain; /* filter gain (in dB) */
+ float rotation; /* rotation of virtual loudspeakers (in degrees) */
+ float elevation; /* elevation of virtual loudspeakers (in deg.) */
+ float radius; /* distance virtual loudspeakers to listener (in metres) */
+
+ int lfe; /* whether or not the LFE channel is used */
+} SOFAlizerContext;
+
+static int close_sofa(struct NCSofa *sofa)
+{
+ av_freep(&sofa->data_delay);
+ av_freep(&sofa->sp_a);
+ av_freep(&sofa->sp_e);
+ av_freep(&sofa->sp_r);
+ av_freep(&sofa->data_ir);
+ nc_close(sofa->ncid);
+ sofa->ncid = 0;
+
+ return 0;
+}
+
+static int load_sofa(AVFilterContext *ctx, char *filename, int *samplingrate)
+{
+ struct SOFAlizerContext *s = ctx->priv;
+ /* variables associated with content of SOFA file: */
+ int ncid, i_n_dims, i_n_vars, i_n_gatts, i_n_unlim_dim_id, status;
+ char psz_data_delay_dim_name[NC_MAX_NAME];
+ float *sp_a, *sp_e, *sp_r, *data_ir;
+ char *psz_sofa_conventions;
+ char dim_name[NC_MAX_NAME]; /* names of netCDF dimensions */
+ size_t *dim_length; /* lengths of netCDF dimensions */
+ char *psz_conventions;
+ unsigned int sample_rate;
+ int data_delay_dim_id[2];
+ int samplingrate_id;
+ int data_delay_id;
+ int n_samples = 0;
+ int i_m_dim_id = -1;
+ int i_n_dim_id = -1;
+ int i_data_ir_id;
+ size_t i_att_len;
+ int m_dim = 0;
+ int *data_delay;
+ int i_sp_id;
+ int i, ret;
+
+ s->sofa.ncid = 0;
+ status = nc_open(filename, NC_NOWRITE, &ncid); /* open SOFA file read-only */
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Can't find SOFA-file '%s'\n", filename);
+ return AVERROR(EINVAL);
+ }
+
+ /* get number of dimensions, vars, global attributes and Id of unlimited dimensions: */
+ nc_inq(ncid, &i_n_dims, &i_n_vars, &i_n_gatts, &i_n_unlim_dim_id);
+
+ /* -- get number of measurements ("M") and length of one IR ("N") -- */
+ dim_length = av_malloc_array(i_n_dims, sizeof(*dim_length));
+ if (!dim_length) {
+ nc_close(ncid);
+ return AVERROR(ENOMEM);
+ }
+
+ for (i = 0; i < i_n_dims; i++) { /* go through all dimensions of file */
+ nc_inq_dim(ncid, i, (char *)&dim_name, &dim_length[i]); /* get dimensions */
+ if (!strncmp("M", (const char *)&dim_name, 1)) /* get ID of dimension "M" */
+ i_m_dim_id = i;
+ if (!strncmp("N", (const char *)&dim_name, 1)) /* get ID of dimension "N" */
+ i_n_dim_id = i;
+ }
+
+ if ((i_m_dim_id == -1) || (i_n_dim_id == -1)) { /* dimension "M" or "N" couldn't be found */
+ av_log(ctx, AV_LOG_ERROR, "Can't find required dimensions in SOFA file.\n");
+ av_freep(&dim_length);
+ nc_close(ncid);
+ return AVERROR(EINVAL);
+ }
+
+ n_samples = dim_length[i_n_dim_id]; /* get number of measurements */
+ m_dim = dim_length[i_m_dim_id]; /* get length of one IR */
+
+ av_freep(&dim_length);
+
+ /* -- check file type -- */
+ /* get length of attritube "Conventions" */
+ status = nc_inq_attlen(ncid, NC_GLOBAL, "Conventions", &i_att_len);
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Can't get length of attribute \"Conventions\".\n");
+ nc_close(ncid);
+ return AVERROR_INVALIDDATA;
+ }
+
+ /* check whether file is SOFA file */
+ psz_conventions = av_malloc(i_att_len + 1);
+ if (!psz_conventions) {
+ nc_close(ncid);
+ return AVERROR(ENOMEM);
+ }
+
+ nc_get_att_text(ncid, NC_GLOBAL, "Conventions", psz_conventions);
+ *(psz_conventions + i_att_len) = 0;
+ if (strncmp("SOFA", psz_conventions, 4)) {
+ av_log(ctx, AV_LOG_ERROR, "Not a SOFA file!\n");
+ av_freep(&psz_conventions);
+ nc_close(ncid);
+ return AVERROR(EINVAL);
+ }
+ av_freep(&psz_conventions);
+
+ status = nc_inq_attlen(ncid, NC_GLOBAL, "SOFAConventions", &i_att_len);
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Can't get length of attribute \"SOFAConventions\".\n");
+ nc_close(ncid);
+ return AVERROR_INVALIDDATA;
+ }
+
+ psz_sofa_conventions = av_malloc(i_att_len + 1);
+ if (!psz_sofa_conventions) {
+ nc_close(ncid);
+ return AVERROR(ENOMEM);
+ }
+
+ nc_get_att_text(ncid, NC_GLOBAL, "SOFAConventions", psz_sofa_conventions);
+ *(psz_sofa_conventions + i_att_len) = 0;
+ if (strncmp("SimpleFreeFieldHRIR", psz_sofa_conventions, i_att_len)) {
+ av_log(ctx, AV_LOG_ERROR, "Not a SimpleFreeFieldHRIR file!\n");
+ av_freep(&psz_sofa_conventions);
+ nc_close(ncid);
+ return AVERROR(EINVAL);
+ }
+ av_freep(&psz_sofa_conventions);
+
+ /* -- get sampling rate of HRTFs -- */
+ /* read ID, then value */
+ status = nc_inq_varid(ncid, "Data.SamplingRate", &samplingrate_id);
+ status += nc_get_var_uint(ncid, samplingrate_id, &sample_rate);
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.SamplingRate.\n");
+ nc_close(ncid);
+ return AVERROR(EINVAL);
+ }
+ *samplingrate = sample_rate; /* remember sampling rate */
+
+ /* -- allocate memory for one value for each measurement position: -- */
+ sp_a = s->sofa.sp_a = av_malloc(sizeof(float) * m_dim);
+ sp_e = s->sofa.sp_e = av_malloc(sizeof(float) * m_dim);
+ sp_r = s->sofa.sp_r = av_malloc(sizeof(float) * m_dim);
+ /* delay and IR values required for each ear and measurement position: */
+ data_delay = s->sofa.data_delay = av_calloc(m_dim * 2, sizeof(int));
+ data_ir = s->sofa.data_ir = av_malloc(sizeof(float) * 2 * m_dim * n_samples);
+
+ if (!data_delay || !sp_a || !sp_e || !sp_r || !data_ir) {
+ /* if memory could not be allocated */
+ close_sofa(&s->sofa);
+ return AVERROR(ENOMEM);
+ }
+
+ /* get impulse responses (HRTFs): */
+ /* get corresponding ID */
+ status = nc_inq_varid(ncid, "Data.IR", &i_data_ir_id);
+ status += nc_get_var_float(ncid, i_data_ir_id, data_ir); /* read and store IRs */
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.IR!\n");
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
+
+ /* get source positions of the HRTFs in the SOFA file: */
+ status = nc_inq_varid(ncid, "SourcePosition", &i_sp_id); /* get corresponding ID */
+ status += nc_get_vara_float(ncid, i_sp_id, (size_t[2]){ 0 , 0 } ,
+ (size_t[2]){ m_dim, 1}, sp_a); /* read & store azimuth angles */
+ status += nc_get_vara_float(ncid, i_sp_id, (size_t[2]){ 0 , 1 } ,
+ (size_t[2]){ m_dim, 1}, sp_e); /* read & store elevation angles */
+ status += nc_get_vara_float (ncid, i_sp_id, (size_t[2]){ 0 , 2 } ,
+ (size_t[2]){ m_dim, 1}, sp_r); /* read & store radii */
+ if (status != NC_NOERR) { /* if any source position variable coudn't be read */
+ av_log(ctx, AV_LOG_ERROR, "Couldn't read SourcePosition.\n");
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
+
+ /* read Data.Delay, check for errors and fit it to data_delay */
+ status = nc_inq_varid(ncid, "Data.Delay", &data_delay_id);
+ status += nc_inq_vardimid(ncid, data_delay_id, &data_delay_dim_id[0]);
+ status += nc_inq_dimname(ncid, data_delay_dim_id[0], psz_data_delay_dim_name);
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay.\n");
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
+
+ /* Data.Delay dimension check */
+ /* dimension of Data.Delay is [I R]: */
+ if (!strncmp(psz_data_delay_dim_name, "I", 2)) {
+ /* check 2 characters to assure string is 0-terminated after "I" */
+ int pi_Delay[2]; /* delays get from SOFA file: */
+ av_log(ctx, AV_LOG_DEBUG, "Data.Delay has dimension [I R]\n");
+ status = nc_get_var_int(ncid, data_delay_id, &pi_Delay[0]);
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay\n");
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
+ int *data_delay_r = data_delay + m_dim;
+ for (i = 0 ; i < m_dim ; i++) /* extend given dimension [I R] to [M R] */
+ { /* assign constant delay value for all measurements to data_delay fields */
+ *(data_delay + i) = pi_Delay[0];
+ *(data_delay_r + i) = pi_Delay[1];
+ }
+ /* dimension of Data.Delay is [M R] */
+ } else if (!strncmp(psz_data_delay_dim_name, "M", 2)) {
+ av_log(ctx, AV_LOG_ERROR, "Data.Delay in dimension [M R]\n");
+ /* get delays from SOFA file: */
+ status = nc_get_var_int(ncid, data_delay_id, data_delay);
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay\n");
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
+ } else { /* dimension of Data.Delay is neither [I R] nor [M R] */
+ av_log(ctx, AV_LOG_ERROR, "Data.Delay does not have the required dimensions [I R] or [M R].\n");
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
+
+ /* save information in SOFA struct: */
+ s->sofa.m_dim = m_dim; /* no. measurement positions */
+ s->sofa.n_samples = n_samples; /* length on one IR */
+ s->sofa.ncid = ncid; /* netCDF ID of SOFA file */
+ nc_close(ncid); /* close SOFA file */
+
+ return 0;
+
+error:
+ close_sofa(&s->sofa);
+ return ret;
+}
+
+const int8_t reorder[16][9] = {
+ { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
+ { 0, 1, 2, 3, -1, -1, -1, -1, -1 },
+ { 0, 1, 2, 3, -1, -1, -1, -1, -1 },
+ { 0, 1, 2, 3, -1, -1, -1, -1, -1 },
+ { 0, 1, 3, 4, 2, -1, -1, -1, -1 },
+ { 0, 1, 3, 4, 2, -1, -1, -1, -1 },
+ { 0, 1, 4, 5, 2, 3, -1, -1, -1 },
+ { 0, 1, 4, 5, 2, 3, -1, -1, -1 },
+ { 0, 1, 5, 6, 4, 2, 3, -1, -1 },
+ { 0, 1, 2, 3, 4, 5, 6, -1, -1 },
+ { 0, 1, 2, 3, 4, 5, 6, 7, -1 },
+ { 0, 1, 2, 3, 4, 5, 6, 7, 8 },
+ { 0, 1, 2, 3, 4, 5, 6, 7, 8 },
+};
+
+static int get_speaker_pos(AVFilterContext *ctx, float *speaker_pos)
+{
+ struct SOFAlizerContext *s = ctx->priv;
+ uint64_t channels_layout = ctx->inputs[0]->channel_layout;
+ float pos_temp[9];
+ int nb_input_channels = ctx->inputs[0]->channels; /* get no. input channels */
+ int n_conv = nb_input_channels;
+ if (channels_layout & AV_CH_LOW_FREQUENCY) { /* if LFE is used */
+ /* decrease number of channels to be convolved: */
+ n_conv = nb_input_channels - 1;
+ }
+
+ /* set speaker positions according to input channel configuration: */
+ switch (channels_layout) {
+ case AV_CH_LAYOUT_MONO:
+ pos_temp[0] = 0;
+ break;
+ case AV_CH_LAYOUT_STEREO:
+ case AV_CH_LAYOUT_2_1:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ break;
+ case AV_CH_LAYOUT_SURROUND:
+ case AV_CH_LAYOUT_3POINT1:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 0;
+ break;
+ case AV_CH_LAYOUT_QUAD:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 120;
+ pos_temp[3] = 240;
+ break;
+ case AV_CH_LAYOUT_4POINT0:
+ case AV_CH_LAYOUT_4POINT1:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 0;
+ pos_temp[3] = 180;
+ break;
+ case AV_CH_LAYOUT_5POINT0:
+ case AV_CH_LAYOUT_5POINT1:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 120;
+ pos_temp[3] = 240;
+ pos_temp[4] = 0;
+ break;
+ case AV_CH_LAYOUT_6POINT0:
+ case AV_CH_LAYOUT_6POINT1:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 90;
+ pos_temp[3] = 270;
+ pos_temp[4] = 150;
+ pos_temp[5] = 210;
+ break;
+ case AV_CH_LAYOUT_7POINT0:
+ case AV_CH_LAYOUT_7POINT1:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 90;
+ pos_temp[3] = 270;
+ pos_temp[4] = 150;
+ pos_temp[5] = 210;
+ pos_temp[6] = 0;
+ break;
+ //case AV_CH_LAYOUT_8POINT0:
+ //case AV_CH_LAYOUT_8POINT1:
+ // pos_temp[0] = 30;
+ // pos_temp[1] = 330;
+ // pos_temp[2] = 90;
+ // pos_temp[3] = 270;
+ // pos_temp[4] = 150;
+ // pos_temp[5] = 210;
+ // pos_temp[6] = 180;
+ // pos_temp[7] = 0;
+ // break;
+ default:
+ return -1;
+ }
+
+ switch (channels_layout) {
+ case AV_CH_LAYOUT_MONO:
+ s->reorder = reorder[0];
+ break;
+ case AV_CH_LAYOUT_STEREO:
+ s->reorder = reorder[1];
+ break;
+ case AV_CH_LAYOUT_2_1:
+ s->reorder = reorder[2];
+ break;
+ case AV_CH_LAYOUT_SURROUND:
+ s->reorder = reorder[3];
+ break;
+ case AV_CH_LAYOUT_3POINT1:
+ s->reorder = reorder[4];
+ break;
+ case AV_CH_LAYOUT_QUAD:
+ s->reorder = reorder[5];
+ break;
+ case AV_CH_LAYOUT_4POINT0:
+ s->reorder = reorder[6];
+ break;
+ case AV_CH_LAYOUT_4POINT1:
+ s->reorder = reorder[7];
+ break;
+ case AV_CH_LAYOUT_5POINT0:
+ s->reorder = reorder[8];
+ break;
+ case AV_CH_LAYOUT_5POINT1:
+ s->reorder = reorder[9];
+ break;
+ case AV_CH_LAYOUT_6POINT0:
+ s->reorder = reorder[10];
+ break;
+ case AV_CH_LAYOUT_6POINT1:
+ s->reorder = reorder[11];
+ break;
+ case AV_CH_LAYOUT_7POINT0:
+ s->reorder = reorder[12];
+ break;
+ case AV_CH_LAYOUT_7POINT1:
+ s->reorder = reorder[13];
+ break;
+ //case AV_CH_LAYOUT_8POINT0:
+ // s->reorder = reorder[14];
+ // break;
+ //case AV_CH_LAYOUT_8POINT1:
+ // s->reorder = reorder[15];
+ // break;
+ default:
+ return -1;
+ }
+
+ memcpy(speaker_pos, pos_temp, n_conv * sizeof(float));
+
+ return 0;
+
+}
+
+static int max_delay(struct NCSofa *sofa)
+{
+ int i, max = 0;
+
+ for (i = 0; i < sofa->m_dim * 2; i++) {
+ /* search maximum delay in given SOFA file */
+ if (*(sofa->data_delay + i) > max)
+ max = *(sofa->data_delay + i) ;
+ }
+
+ return max;
+}
+
+static int find_m(SOFAlizerContext *s, int azim, int elev, float radius)
+{
+ /* get source positions and M of currently selected SOFA file */
+ float *sp_a = s->sofa.sp_a; /* azimuth angle */
+ float *sp_e = s->sofa.sp_e; /* elevation angle */
+ float *sp_r = s->sofa.sp_r; /* radius */
+ int m_dim = s->sofa.m_dim; /* no. measurements */
+ int best_id = 0; /* index m currently closest to desired source pos. */
+ float delta = 1000; /* offset between desired and currently best pos. */
+ float current;
+ int i;
+
+ for (i = 0; i < m_dim; i++) {
+ /* search through all measurements in currently selected SOFA file */
+ /* distance of current to desired source position: */
+ current = fabs(*(sp_a++) - azim) +
+ fabs(*(sp_e++) - elev) +
+ fabs(*(sp_r++) - radius);
+ if (current <= delta) {
+ /* if current distance is smaller than smallest distance so far */
+ delta = current;
+ best_id = i; /* remember index */
+ }
+ }
+
+ return best_id;
+}
+
+static int compensate_volume(AVFilterContext *ctx)
+{
+ struct SOFAlizerContext *s = ctx->priv;
+ float compensate;
+ float energy = 0;
+ float *ir;
+ int m, j;
+
+ if (s->sofa.ncid) {
+ /* find IR at front center position in i-th SOFA file (IR closest to 0°,0°,1m) */
+ struct NCSofa *sofa = &s->sofa;
+ m = find_m(s, 0, 0, 1);
+ /* get energy of that IR and compensate volume */
+ ir = sofa->data_ir + 2 * m * sofa->n_samples;
+ for (j = 0 ; j < sofa->n_samples ; j++) {
+ energy += *(ir + j) * *(ir + j);
+ }
+ compensate = 256 / (sofa->n_samples * sqrt(energy));
+ av_log(ctx, AV_LOG_DEBUG, "Compensate-factor: %f\n", compensate);
+ ir = sofa->data_ir;
+ for (j = 0; j < sofa->n_samples * sofa->m_dim * 2; j++) {
+ *(ir + j) *= compensate; /* apply volume compensation to IRs */
+ }
+ }
+
+ return 0;
+}
+
+static void sofalizer_convolute(SOFAlizerContext *s, AVFrame *in, AVFrame *out, int offset,
+ int *write, int *delay, float *ir, int *i_n_clippings,
+ float *ringbuffer)
+{
+ int n_samples = s->sofa.n_samples; /* length of one IR */
+ const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */
+ float *temp_ir;
+ float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */
+ int in_channels = in->channels; /* number of input channels */
+ /* ring buffer length is: longest IR plus max. delay -> next power of 2 */
+ int buffer_length = s->buffer_length;
+ /* -1 for AND instead of MODULO (applied to powers of 2): */
+ uint32_t modulo = (uint32_t) buffer_length - 1;
+ float *buffer[10]; /* holds ringbuffer for each input channel */
+ int wr = *write;
+ int read;
+ int i, j, l;
+
+ dst += offset;
+ for (l = 0; l < in_channels; l++) {
+ /* get starting address of ringbuffer for each input channel */
+ buffer[l] = ringbuffer + l * buffer_length;
+ }
+
+ for (i = 0; i < in->nb_samples; i++) {
+ *(dst) = 0;
+ for (l = 0; l < in_channels; l++) {
+ /* write current input sample to ringbuffer (for each channel) */
+ *(buffer[l] + wr) = src[s->reorder[l]];
+ }
+ temp_ir = ir; /* using same set of IRs for each sample */
+ /* loop goes through all channels to be convolved (excl. LFE): */
+ for (l = 0; l < s->n_conv; l++) {
+ /* current read position in ringbuffer: input sample write position
+ * - delay for l-th ch. + diff. betw. IR length and buffer length
+ * (mod buffer length) */
+ read = (wr - *(delay + l) - (n_samples - 1) + buffer_length) & modulo;
+
+ for (j = 0; j < n_samples; j++) { /* go through samples of IR */
+ /* multiply signal and IR, and add up the results */
+ *dst += *(buffer[l] + ((read++) & modulo)) * *(temp_ir++);
+ }
+ }
+ if (s->lfe) { /* LFE */
+ /* apply gain to LFE signal and add to output buffer */
+ *dst += *(buffer[s->n_conv] + wr) * s->gain_lfe;
+ }
+ /* clippings counter */
+ if (fabs(*dst) > 1)
+ *i_n_clippings += 1;
+
+ /* move output buffer pointer by +2 to get to next sample of processed channel: */
+ dst += 2;
+ src += in_channels;
+ wr = (wr + 1) & modulo; /* update ringbuffer write position */
+ }
+
+ *write = wr; /* remember write position in ringbuffer for next call */
+
+ return;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ SOFAlizerContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ int n_clippings_l = 0;
+ int n_clippings_r = 0;
+ AVFrame *out;
+
+ out = ff_get_audio_buffer(outlink, in->nb_samples);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+
+ sofalizer_convolute(s, in, out, 0, &s->write_l,
+ s->delay_l, s->data_ir_l,
+ &n_clippings_l,
+ s->ringbuffer_l);
+ sofalizer_convolute(s, in, out, 1, &s->write_r,
+ s->delay_r, s->data_ir_r,
+ &n_clippings_r,
+ s->ringbuffer_r);
+
+ /* display error message if clipping occured */
+ if (n_clippings_l + n_clippings_r > 0) {
+ av_log(ctx, AV_LOG_WARNING, "%d of %d samples clipped. Please reduce gain.\n",
+ n_clippings_l + n_clippings_r, out->nb_samples * 2);
+ }
+
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ struct SOFAlizerContext *s = ctx->priv;
+ AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layouts = NULL;
+ static int sample_rates[] = { 48000, -1 };
+ static const uint64_t channel_layouts[] = { AV_CH_LAYOUT_MONO,
+ AV_CH_LAYOUT_STEREO,
+ AV_CH_LAYOUT_2_1,
+ AV_CH_LAYOUT_SURROUND,
+ AV_CH_LAYOUT_3POINT1,
+ AV_CH_LAYOUT_QUAD,
+ AV_CH_LAYOUT_4POINT0,
+ AV_CH_LAYOUT_4POINT1,
+ AV_CH_LAYOUT_5POINT0,
+ AV_CH_LAYOUT_5POINT1,
+ AV_CH_LAYOUT_6POINT0,
+ AV_CH_LAYOUT_6POINT1,
+ AV_CH_LAYOUT_7POINT0,
+ AV_CH_LAYOUT_7POINT1,
+ 0, };
+ int ret;
+
+ ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT);
+ if (ret)
+ return ret;
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret)
+ return ret;
+
+ layouts = ff_make_formatu64_list(channel_layouts);
+ if (!layouts)
+ return AVERROR(ENOMEM);
+
+ ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_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->outputs[0]->in_channel_layouts);
+ if (ret)
+ return ret;
+
+ sample_rates[0] = s->sample_rate;
+ formats = ff_make_format_list(sample_rates);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+static int load_data(AVFilterContext *ctx, int azim, int elev, float radius)
+{
+ struct SOFAlizerContext *s = ctx->priv;
+ const int n_samples = s->sofa.n_samples;
+ int n_conv = s->n_conv; /* no. channels to convolve (excl. LFE) */
+ int delay_l[10]; /* broadband delay for each IR */
+ int delay_r[10];
+ int nb_input_channels = ctx->inputs[0]->channels; /* no. input channels */
+ float f_gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10); /* gain - 3dB/channel */
+ float *data_ir_l = NULL;
+ float *data_ir_r = NULL;
+ int offset = 0; /* used for faster pointer arithmetics in for-loop */
+ int i_m[s->n_conv]; /* measurement index m of IR closest to required source positions */
+ int i, j, azim_orig = azim;
+
+ if (!s->sofa.ncid) { /* if an invalid SOFA file has been selected */
+ av_log(ctx, AV_LOG_ERROR, "Selected SOFA file is invalid. Please select valid SOFA file.\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ /* get temporary IR for L and R channel */
+ data_ir_l = av_malloc(sizeof(float) * n_conv * n_samples);
+ data_ir_r = av_malloc(sizeof(float) * n_conv * n_samples);
+ if (!data_ir_r || !data_ir_l)
+ return AVERROR(ENOMEM);
+
+ for (i = 0; i < s->n_conv; i++) {
+ /* load and store IRs and corresponding delays */
+ azim = (int)(s->speaker_pos[i] + azim_orig) % 360;
+ /* get id of IR closest to desired position */
+ i_m[i] = find_m(s, azim, elev, radius);
+
+ /* load the delays associated with the current IRs */
+ delay_l[i] = *(s->sofa.data_delay + 2 * i_m[i]);
+ delay_r[i] = *(s->sofa.data_delay + 2 * i_m[i] + 1);
+
+ offset = i * n_samples; /* no. samples already written */
+ for (j = 0; j < n_samples; j++) {
+ /* load reversed IRs of the specified source position
+ * sample-by-sample for left and right ear; and apply gain */
+ *(data_ir_l + offset + j) = /* left channel */
+ *(s->sofa.data_ir + 2 * i_m[i] * n_samples + n_samples - 1 - j) * f_gain_lin;
+ *(data_ir_r + offset + j) = /* right channel */
+ *(s->sofa.data_ir + 2 * i_m[i] * n_samples + n_samples - 1 - j + n_samples) * f_gain_lin;
+ }
+
+ av_log(ctx, AV_LOG_DEBUG, "Index: %d, Azimuth: %f, Elevation: %f, Radius: %f of SOFA file.\n",
+ i_m[i], *(s->sofa.sp_a + i_m[i]), *(s->sofa.sp_e + i_m[i]), *(s->sofa.sp_r + i_m[i]));
+ }
+
+ /* copy IRs and delays to allocated memory in the SOFAlizerContext struct: */
+ memcpy(s->data_ir_l, data_ir_l, sizeof(float) * n_conv * n_samples);
+ memcpy(s->data_ir_r, data_ir_r, sizeof(float) * n_conv * n_samples);
+
+ av_free(data_ir_l); /* free temporary IR memory */
+ av_free(data_ir_r);
+
+ memcpy(s->delay_l, &delay_l[0], sizeof(int) * s->n_conv);
+ memcpy(s->delay_r, &delay_r[0], sizeof(int) * s->n_conv);
+
+ return 0;
+}
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ SOFAlizerContext *s = ctx->priv;
+ int ret;
+
+ /* load SOFA file, */
+ /* initialize file IDs to 0 before attempting to load SOFA files,
+ * this assures that in case of error, only the memory of already
+ * loaded files is free'd */
+ s->sofa.ncid = 0;
+ ret = load_sofa(ctx, s->filename, &s->sample_rate);
+ if (ret) {
+ /* file loading error */
+ av_log(ctx, AV_LOG_ERROR, "Error while loading SOFA file: '%s'\n", s->filename);
+ } else { /* no file loading error, resampling not required */
+ av_log(ctx, AV_LOG_DEBUG, "File '%s' loaded.\n", s->filename);
+ }
+
+ if (ret) {
+ av_log(ctx, AV_LOG_ERROR, "No valid SOFA file could be loaded. Please specify valid SOFA file.\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static inline unsigned clz(unsigned x)
+{
+ unsigned i = sizeof(x) * 8;
+
+ while (x) {
+ x >>= 1;
+ i--;
+ }
+
+ return i;
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ SOFAlizerContext *s = ctx->priv;
+ int nb_input_channels = inlink->channels; /* no. input channels */
+ int n_max_ir = 0;
+ int n_current;
+ int n_max = 0;
+ int ret;
+
+ /* gain -3 dB per channel, -6 dB to get LFE on a similar level */
+ s->gain_lfe = expf((s->gain - 3 * inlink->channels - 6) / 20 * M_LN10);
+
+ if (inlink->channel_layout & AV_CH_LOW_FREQUENCY) { /* if LFE is used */
+ s->lfe = 1;
+ s->n_conv = nb_input_channels - 1 ; /* LFE is an input channel but requires no convolution */
+ } else /* if LFE is not used */ {
+ s->lfe = 0;
+ s->n_conv = nb_input_channels ;
+ }
+
+ /* get size of ringbuffer (longest IR plus max. delay) */
+ /* then choose next power of 2 for performance optimization */
+ n_current = s->sofa.n_samples + max_delay(&s->sofa);
+ if (n_current > n_max) {
+ /* length of longest IR plus max. delay (in all SOFA files) */
+ n_max = n_current;
+ /* length of longest IR (without delay, in all SOFA files) */
+ n_max_ir = s->sofa.n_samples;
+ }
+ /* buffer length is longest IR plus max. delay -> next power of 2
+ (32 - count leading zeros gives required exponent) */
+ s->buffer_length = exp2(32 - clz((uint32_t)n_max));
+
+ /* Allocate memory for the impulse responses, delays and the ringbuffers */
+ /* size: (longest IR) * (number of channels to convolute), without LFE */
+ s->data_ir_l = av_malloc(sizeof(float) * n_max_ir * s->n_conv);
+ s->data_ir_r = av_malloc(sizeof(float) * n_max_ir * s->n_conv);
+ /* length: number of channels to convolute */
+ s->delay_l = av_malloc(sizeof(int) * s->n_conv);
+ s->delay_r = av_malloc(sizeof(int) * s->n_conv);
+ /* length: (buffer length) * (number of input channels),
+ * OR: buffer length (if frequency domain processing)
+ * calloc zero-initializes the buffer */
+ s->ringbuffer_l = av_calloc(s->buffer_length * nb_input_channels, sizeof(float));
+ s->ringbuffer_r = av_calloc(s->buffer_length * nb_input_channels, sizeof(float));
+ /* length: number of channels to convolute */
+ s->speaker_pos = av_malloc_array(s->n_conv, sizeof(*s->speaker_pos));
+
+ /* memory allocation failed: */
+ if (!s->data_ir_l || !s->data_ir_r || !s->delay_l ||
+ !s->delay_r || !s->ringbuffer_l || !s->ringbuffer_r ||
+ !s->speaker_pos)
+ return AVERROR(ENOMEM);
+
+ compensate_volume(ctx);
+
+ /* get speaker positions */
+ if ((ret = get_speaker_pos(ctx, s->speaker_pos)) < 0) {
+ av_log(ctx, AV_LOG_ERROR, "Couldn't get speaker positions. Input channel configuration not supported.\n");
+ return ret;
+ }
+ /* load IRs to data_ir_l and data_ir_r for required directions */
+ /* only load IRs if time-domain convolution is used. */
+ if ((ret = load_data(ctx, s->rotation, s->elevation, s->radius)) < 0)
+ return ret;
+
+ av_log(ctx, AV_LOG_DEBUG, "Samplerate: %d Channels to convolute: %d, Length of ringbuffer: %d x %d\n",
+ inlink->sample_rate, s->n_conv, nb_input_channels, (int)s->buffer_length);
+
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ SOFAlizerContext *s = ctx->priv;
+
+ if (s->sofa.ncid) {
+ av_freep(&s->sofa.sp_a);
+ av_freep(&s->sofa.sp_e);
+ av_freep(&s->sofa.sp_r);
+ av_freep(&s->sofa.data_delay);
+ av_freep(&s->sofa.data_ir);
+ }
+ av_freep(&s->delay_l);
+ av_freep(&s->delay_r);
+ av_freep(&s->data_ir_l);
+ av_freep(&s->data_ir_r);
+ av_freep(&s->ringbuffer_l);
+ av_freep(&s->ringbuffer_r);
+ av_freep(&s->speaker_pos);
+}
+
+#define OFFSET(x) offsetof(SOFAlizerContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption sofalizer_options[] = {
+ { "sofa", "sofa filename", OFFSET(filename), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS },
+ { "gain", "set gain in dB", OFFSET(gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20, 40, .flags = FLAGS },
+ { "rotation", "set rotation" , OFFSET(rotation), AV_OPT_TYPE_FLOAT, {.dbl=0}, -360, 360, .flags = FLAGS },
+ { "elevation", "set elevation", OFFSET(elevation), AV_OPT_TYPE_FLOAT, {.dbl=0}, -90, 90, .flags = FLAGS },
+ { "radius", "set radius", OFFSET(radius), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 2.1, .flags = FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(sofalizer);
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_input,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_sofalizer = {
+ .name = "sofalizer",
+ .description = NULL_IF_CONFIG_SMALL("SOFAlizer (Spatially Oriented Format for Acoustics)."),
+ .priv_size = sizeof(SOFAlizerContext),
+ .priv_class = &sofalizer_class,
+ .init = init,
+ .uninit = uninit,
+ .query_formats = query_formats,
+ .inputs = inputs,
+ .outputs = outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 0eeef53..131e067 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -109,6 +109,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(SIDECHAINGATE, sidechaingate, af);
REGISTER_FILTER(SILENCEDETECT, silencedetect, af);
REGISTER_FILTER(SILENCEREMOVE, silenceremove, af);
+ REGISTER_FILTER(SOFALIZER, sofalizer, af);
REGISTER_FILTER(STEREOTOOLS, stereotools, af);
REGISTER_FILTER(STEREOWIDEN, stereowiden, af);
REGISTER_FILTER(TREBLE, treble, af);
diff --git a/libavfilter/formats.c b/libavfilter/formats.c
index 2b13cbf..27916ba 100644
--- a/libavfilter/formats.c
+++ b/libavfilter/formats.c
@@ -289,6 +289,17 @@ AVFilterFormats *ff_make_format_list(const int *fmts)
return formats;
}
+AVFilterChannelLayouts *ff_make_formatu64_list(const uint64_t *fmts)
+{
+ MAKE_FORMAT_LIST(AVFilterChannelLayouts,
+ channel_layouts, nb_channel_layouts);
+ if (count)
+ memcpy(formats->channel_layouts, fmts,
+ sizeof(*formats->channel_layouts) * count);
+
+ return formats;
+}
+
AVFilterChannelLayouts *avfilter_make_format64_list(const int64_t *fmts)
{
MAKE_FORMAT_LIST(AVFilterChannelLayouts,
diff --git a/libavfilter/formats.h b/libavfilter/formats.h
index 3d730f3..ce09d3c 100644
--- a/libavfilter/formats.h
+++ b/libavfilter/formats.h
@@ -141,6 +141,9 @@ AVFilterChannelLayouts *ff_all_channel_counts(void);
av_warn_unused_result
AVFilterChannelLayouts *avfilter_make_format64_list(const int64_t *fmts);
+av_warn_unused_result
+AVFilterChannelLayouts *ff_make_formatu64_list(const uint64_t *fmts);
+
/**
* A helper for query_formats() which sets all links to the same list of channel
--
1.9.1
More information about the ffmpeg-devel
mailing list