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avfilter/af_sofalizer: add frequency domain processing and use it by default

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Code ported from SOFAlizer patch for VLC.

Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2015-12-13 23:05:09 +01:00
parent 30fe3fd527
commit 2f12172d67
3 changed files with 277 additions and 29 deletions
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3
configure vendored
View File

@ -2892,7 +2892,8 @@ showfreqs_filter_deps="avcodec"
showfreqs_filter_select="fft"
showspectrum_filter_deps="avcodec"
showspectrum_filter_select="rdft"
sofalizer_filter_deps="netcdf"
sofalizer_filter_deps="netcdf avcodec"
sofalizer_filter_select="fft"
spp_filter_deps="gpl avcodec"
spp_filter_select="fft idctdsp fdctdsp me_cmp pixblockdsp"
stereo3d_filter_deps="gpl"

6
doc/filters.texi
View File

@ -2916,6 +2916,12 @@ Set elevation of virtual speakers in deg. Default is 0.
@item radius
Set distance in meters between loudspeakers and the listener with near-field
HRTFs. Default is 1.
@item type
Set processing type. Can be @var{time} or @var{freq}. @var{time} is
processing audio in time domain which is slow but gives high quality output.
@var{freq} is processing audio in frequency domain which is fast but gives
mediocre output. Default is @var{freq}.
@end table
@section stereotools

297
libavfilter/af_sofalizer.c
View File

@ -28,12 +28,16 @@
#include <math.h>
#include <netcdf.h>
#include "libavcodec/avfft.h"
#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "internal.h"
#include "audio.h"
#define TIME_DOMAIN 0
#define FREQUENCY_DOMAIN 1
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) */
@ -67,6 +71,7 @@ typedef struct SOFAlizerContext {
int write[2]; /* current write position to ringbuffer */
int buffer_length; /* is: longest IR plus max. delay in all SOFA files */
/* then choose next power of 2 */
int n_fft; /* number of samples in one FFT block */
/* netCDF variables */
int *delay[2]; /* broadband delay for each channel/IR to be convolved */
@ -74,12 +79,17 @@ typedef struct SOFAlizerContext {
float *data_ir[2]; /* IRs for all channels to be convolved */
/* (this excludes the LFE) */
float *temp_src[2];
FFTComplex *temp_fft[2];
/* 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 type; /* processing type */
FFTContext *fft[2], *ifft[2];
FFTComplex *data_hrtf[2];
AVFloatDSPContext *fdsp;
} SOFAlizerContext;
@ -259,11 +269,8 @@ static int load_sofa(AVFilterContext *ctx, char *filename, int *samplingrate)
/* 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_array(m_dim * n_samples, sizeof(float) * 2);
s->temp_src[0] = av_calloc(FFALIGN(n_samples, 16), sizeof(float));
s->temp_src[1] = av_calloc(FFALIGN(n_samples, 16), sizeof(float));
if (!data_delay || !sp_a || !sp_e || !sp_r || !data_ir ||
!s->temp_src[0] || !s->temp_src[1]) {
if (!data_delay || !sp_a || !sp_e || !sp_r || !data_ir) {
/* if memory could not be allocated */
close_sofa(&s->sofa);
return AVERROR(ENOMEM);
@ -590,6 +597,7 @@ typedef struct ThreadData {
int *n_clippings;
float **ringbuffer;
float **temp_src;
FFTComplex **temp_fft;
} ThreadData;
static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
@ -678,6 +686,120 @@ static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
return 0;
}
static int sofalizer_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
SOFAlizerContext *s = ctx->priv;
ThreadData *td = arg;
AVFrame *in = td->in, *out = td->out;
int offset = jobnr;
int *write = &td->write[jobnr];
FFTComplex *hrtf = s->data_hrtf[jobnr]; /* get pointers to current HRTF data */
int *n_clippings = &td->n_clippings[jobnr];
float *ringbuffer = td->ringbuffer[jobnr];
const 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 *dst = (float *)out->data[0]; /* get pointer to audio output buffer */
const int in_channels = s->n_conv; /* number of input channels */
/* ring buffer length is: longest IR plus max. delay -> next power of 2 */
const int buffer_length = s->buffer_length;
/* -1 for AND instead of MODULO (applied to powers of 2): */
const uint32_t modulo = (uint32_t)buffer_length - 1;
FFTComplex *fft_in = s->temp_fft[jobnr]; /* temporary array for FFT input/output data */
FFTContext *ifft = s->ifft[jobnr];
FFTContext *fft = s->fft[jobnr];
const int n_conv = s->n_conv;
const int n_fft = s->n_fft;
int wr = *write;
int n_read;
int i, j;
dst += offset;
/* find minimum between number of samples and output buffer length:
* (important, if one IR is longer than the output buffer) */
n_read = FFMIN(s->sofa.n_samples, in->nb_samples);
for (j = 0; j < n_read; j++) {
/* initialize output buf with saved signal from overflow buf */
dst[2 * j] = ringbuffer[wr];
ringbuffer[wr] = 0.0; /* re-set read samples to zero */
/* update ringbuffer read/write position */
wr = (wr + 1) & modulo;
}
/* initialize rest of output buffer with 0 */
for (j = n_read; j < in->nb_samples; j++) {
dst[2 * j] = 0;
}
for (i = 0; i < n_conv; i++) {
if (i == s->lfe_channel) { /* LFE */
for (j = 0; j < in->nb_samples; j++) {
/* apply gain to LFE signal and add to output buffer */
dst[2 * j] += src[i + j * in_channels] * s->gain_lfe;
}
continue;
}
/* outer loop: go through all input channels to be convolved */
offset = i * n_fft; /* no. samples already processed */
/* fill FFT input with 0 (we want to zero-pad) */
memset(fft_in, 0, sizeof(FFTComplex) * n_fft);
for (j = 0; j < in->nb_samples; j++) {
/* prepare input for FFT */
/* write all samples of current input channel to FFT input array */
fft_in[j].re = src[j * in_channels + i];
}
/* transform input signal of current channel to frequency domain */
av_fft_permute(fft, fft_in);
av_fft_calc(fft, fft_in);
for (j = 0; j < n_fft; j++) {
const float re = fft_in[j].re;
const float im = fft_in[j].im;
/* complex multiplication of input signal and HRTFs */
/* output channel (real): */
fft_in[j].re = re * (hrtf + offset + j)->re - im * (hrtf + offset + j)->im;
/* output channel (imag): */
fft_in[j].im = re * (hrtf + offset + j)->im + im * (hrtf + offset + j)->re;
}
/* transform output signal of current channel back to time domain */
av_fft_permute(ifft, fft_in);
av_fft_calc(ifft, fft_in);
for (j = 0; j < in->nb_samples; j++) {
/* write output signal of current channel to output buffer */
dst[2 * j] += fft_in[j].re / (float)n_fft;
}
for (j = 0; j < n_samples - 1; j++) { /* overflow length is IR length - 1 */
/* write the rest of output signal to overflow buffer */
int write_pos = (wr + j) & modulo;
*(ringbuffer + write_pos) += fft_in[in->nb_samples + j].re / (float)n_fft;
}
}
/* go through all samples of current output buffer: count clippings */
for (i = 0; i < out->nb_samples; i++) {
/* clippings counter */
if (fabs(*dst) > 1) { /* if current output sample > 1 */
*n_clippings = *n_clippings + 1;
}
/* move output buffer pointer by +2 to get to next sample of processed channel: */
dst += 2;
}
/* remember read/write position in ringbuffer for next call */
*write = wr;
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
@ -697,8 +819,13 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
td.in = in; td.out = out; td.write = s->write;
td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings;
td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src;
td.temp_fft = s->temp_fft;
ctx->internal->execute(ctx, sofalizer_convolute, &td, NULL, 2);
if (s->type == TIME_DOMAIN) {
ctx->internal->execute(ctx, sofalizer_convolute, &td, NULL, 2);
} else {
ctx->internal->execute(ctx, sofalizer_fast_convolute, &td, NULL, 2);
}
emms_c();
/* display error message if clipping occured */
@ -776,10 +903,15 @@ 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 */
int n_fft = s->n_fft;
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 gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10); /* gain - 3dB/channel */
FFTComplex *data_hrtf_l = NULL;
FFTComplex *data_hrtf_r = NULL;
FFTComplex *fft_in_l = NULL;
FFTComplex *fft_in_r = NULL;
float *data_ir_l = NULL;
float *data_ir_r = NULL;
int offset = 0; /* used for faster pointer arithmetics in for-loop */
@ -791,13 +923,27 @@ static int load_data(AVFilterContext *ctx, int azim, int elev, float radius)
return AVERROR_INVALIDDATA;
}
/* get temporary IR for L and R channel */
data_ir_l = av_malloc_array(n_conv * n_samples, sizeof(*data_ir_l));
data_ir_r = av_malloc_array(n_conv * n_samples, sizeof(*data_ir_r));
if (!data_ir_r || !data_ir_l) {
av_free(data_ir_l);
av_free(data_ir_r);
return AVERROR(ENOMEM);
if (s->type == TIME_DOMAIN) {
s->temp_src[0] = av_calloc(FFALIGN(n_samples, 16), sizeof(float));
s->temp_src[1] = av_calloc(FFALIGN(n_samples, 16), sizeof(float));
/* get temporary IR for L and R channel */
data_ir_l = av_malloc_array(n_conv * n_samples, sizeof(*data_ir_l));
data_ir_r = av_malloc_array(n_conv * n_samples, sizeof(*data_ir_r));
if (!data_ir_r || !data_ir_l || !s->temp_src[0] || !s->temp_src[1]) {
av_free(data_ir_l);
av_free(data_ir_r);
return AVERROR(ENOMEM);
}
} else {
/* get temporary HRTF memory for L and R channel */
data_hrtf_l = av_malloc_array(n_fft, sizeof(*data_hrtf_l) * n_conv);
data_hrtf_r = av_malloc_array(n_fft, sizeof(*data_hrtf_r) * n_conv);
if (!data_hrtf_r || !data_hrtf_l) {
av_free(data_hrtf_l);
av_free(data_hrtf_r);
return AVERROR(ENOMEM);
}
}
for (i = 0; i < s->n_conv; i++) {
@ -811,26 +957,81 @@ static int load_data(AVFilterContext *ctx, int azim, int elev, float radius)
delay_l[i] = *(s->sofa.data_delay + 2 * m[i]);
delay_r[i] = *(s->sofa.data_delay + 2 * 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 * m[i] * n_samples + n_samples - 1 - j) * gain_lin;
*(data_ir_r + offset + j) = /* right channel */
*(s->sofa.data_ir + 2 * m[i] * n_samples + n_samples - 1 - j + n_samples) * gain_lin;
if (s->type == TIME_DOMAIN) {
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 * m[i] * n_samples + n_samples - 1 - j) * gain_lin;
*(data_ir_r + offset + j) = /* right channel */
*(s->sofa.data_ir + 2 * m[i] * n_samples + n_samples - 1 - j + n_samples) * gain_lin;
}
} else {
fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l));
fft_in_r = av_calloc(n_fft, sizeof(*fft_in_r));
if (!fft_in_l || !fft_in_r) {
av_free(data_hrtf_l);
av_free(data_hrtf_r);
av_free(fft_in_l);
av_free(fft_in_r);
return AVERROR(ENOMEM);
}
offset = i * n_fft; /* no. samples already written */
for (j = 0; j < n_samples; j++) {
/* load non-reversed IRs of the specified source position
* sample-by-sample and apply gain,
* L channel is loaded to real part, R channel to imag part,
* IRs ared shifted by L and R delay */
fft_in_l[delay_l[i] + j].re = /* left channel */
*(s->sofa.data_ir + 2 * m[i] * n_samples + j) * gain_lin;
fft_in_r[delay_r[i] + j].re = /* right channel */
*(s->sofa.data_ir + (2 * m[i] + 1) * n_samples + j) * gain_lin;
}
/* actually transform to frequency domain (IRs -> HRTFs) */
av_fft_permute(s->fft[0], fft_in_l);
av_fft_calc(s->fft[0], fft_in_l);
memcpy(data_hrtf_l + offset, fft_in_l, n_fft * sizeof(*fft_in_l));
av_fft_permute(s->fft[0], fft_in_r);
av_fft_calc(s->fft[0], fft_in_r);
memcpy(data_hrtf_r + offset, fft_in_r, n_fft * sizeof(*fft_in_r));
}
av_log(ctx, AV_LOG_DEBUG, "Index: %d, Azimuth: %f, Elevation: %f, Radius: %f of SOFA file.\n",
m[i], *(s->sofa.sp_a + m[i]), *(s->sofa.sp_e + m[i]), *(s->sofa.sp_r + m[i]));
}
/* copy IRs and delays to allocated memory in the SOFAlizerContext struct: */
memcpy(s->data_ir[0], data_ir_l, sizeof(float) * n_conv * n_samples);
memcpy(s->data_ir[1], data_ir_r, sizeof(float) * n_conv * n_samples);
if (s->type == TIME_DOMAIN) {
/* copy IRs and delays to allocated memory in the SOFAlizerContext struct: */
memcpy(s->data_ir[0], data_ir_l, sizeof(float) * n_conv * n_samples);
memcpy(s->data_ir[1], data_ir_r, sizeof(float) * n_conv * n_samples);
av_free(data_ir_l); /* free temporary IR memory */
av_free(data_ir_r);
av_freep(&data_ir_l); /* free temporary IR memory */
av_freep(&data_ir_r);
} else {
s->data_hrtf[0] = av_malloc_array(n_fft * s->n_conv, sizeof(FFTComplex));
s->data_hrtf[1] = av_malloc_array(n_fft * s->n_conv, sizeof(FFTComplex));
if (!s->data_hrtf[0] || !s->data_hrtf[1]) {
av_freep(&data_hrtf_l);
av_freep(&data_hrtf_r);
av_freep(&fft_in_l);
av_freep(&fft_in_r);
return AVERROR(ENOMEM); /* memory allocation failed */
}
memcpy(s->data_hrtf[0], data_hrtf_l, /* copy HRTF data to */
sizeof(FFTComplex) * n_conv * n_fft); /* filter struct */
memcpy(s->data_hrtf[1], data_hrtf_r,
sizeof(FFTComplex) * n_conv * n_fft);
av_freep(&data_hrtf_l); /* free temporary HRTF memory */
av_freep(&data_hrtf_r);
av_freep(&fft_in_l); /* free temporary FFT memory */
av_freep(&fft_in_r);
}
memcpy(s->delay[0], &delay_l[0], sizeof(int) * s->n_conv);
memcpy(s->delay[1], &delay_r[0], sizeof(int) * s->n_conv);
@ -890,6 +1091,12 @@ static int config_input(AVFilterLink *inlink)
int n_max = 0;
int ret;
if (s->type == FREQUENCY_DOMAIN) {
inlink->partial_buf_size =
inlink->min_samples =
inlink->max_samples = inlink->sample_rate;
}
/* 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);
@ -907,6 +1114,18 @@ static int config_input(AVFilterLink *inlink)
/* 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));
s->n_fft = exp2(32 - clz((uint32_t)(n_max + inlink->sample_rate)));
if (s->type == FREQUENCY_DOMAIN) {
av_fft_end(s->fft[0]);
av_fft_end(s->fft[1]);
s->fft[0] = av_fft_init(log2(s->n_fft), 0);
s->fft[1] = av_fft_init(log2(s->n_fft), 0);
av_fft_end(s->ifft[0]);
av_fft_end(s->ifft[1]);
s->ifft[0] = av_fft_init(log2(s->n_fft), 1);
s->ifft[1] = av_fft_init(log2(s->n_fft), 1);
}
/* Allocate memory for the impulse responses, delays and the ringbuffers */
/* size: (longest IR) * (number of channels to convolute) */
@ -918,8 +1137,19 @@ static int config_input(AVFilterLink *inlink)
/* length: (buffer length) * (number of input channels),
* OR: buffer length (if frequency domain processing)
* calloc zero-initializes the buffer */
s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
if (s->type == TIME_DOMAIN) {
s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
} else {
s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float));
s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float));
s->temp_fft[0] = av_malloc_array(s->n_fft, sizeof(FFTComplex));
s->temp_fft[1] = av_malloc_array(s->n_fft, sizeof(FFTComplex));
if (!s->temp_fft[0] || !s->temp_fft[1])
return AVERROR(ENOMEM);
}
/* length: number of channels to convolute */
s->speaker_azim = av_calloc(s->n_conv, sizeof(*s->speaker_azim));
s->speaker_elev = av_calloc(s->n_conv, sizeof(*s->speaker_elev));
@ -937,8 +1167,8 @@ static int config_input(AVFilterLink *inlink)
av_log(ctx, AV_LOG_ERROR, "Couldn't get speaker positions. Input channel configuration not supported.\n");
return ret;
}
/* load IRs to data_ir[0] and data_ir[1] 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;
@ -959,6 +1189,10 @@ static av_cold void uninit(AVFilterContext *ctx)
av_freep(&s->sofa.data_delay);
av_freep(&s->sofa.data_ir);
}
av_fft_end(s->ifft[0]);
av_fft_end(s->ifft[1]);
av_fft_end(s->fft[0]);
av_fft_end(s->fft[1]);
av_freep(&s->delay[0]);
av_freep(&s->delay[1]);
av_freep(&s->data_ir[0]);
@ -969,6 +1203,10 @@ static av_cold void uninit(AVFilterContext *ctx)
av_freep(&s->speaker_elev);
av_freep(&s->temp_src[0]);
av_freep(&s->temp_src[1]);
av_freep(&s->temp_fft[0]);
av_freep(&s->temp_fft[1]);
av_freep(&s->data_hrtf[0]);
av_freep(&s->data_hrtf[1]);
av_freep(&s->fdsp);
}
@ -981,6 +1219,9 @@ static const AVOption sofalizer_options[] = {
{ "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, 3, .flags = FLAGS },
{ "type", "set processing", OFFSET(type), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, .flags = FLAGS, "type" },
{ "time", "time domain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, .flags = FLAGS, "type" },
{ "freq", "frequency domain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, .flags = FLAGS, "type" },
{ NULL }
};