From 2f12172d670996ff8f18b80ebdee7d0a8c230ac3 Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Sun, 13 Dec 2015 23:05:09 +0100 Subject: [PATCH] avfilter/af_sofalizer: add frequency domain processing and use it by default Code ported from SOFAlizer patch for VLC. Signed-off-by: Paul B Mahol --- configure | 3 +- doc/filters.texi | 6 + libavfilter/af_sofalizer.c | 297 +++++++++++++++++++++++++++++++++---- 3 files changed, 277 insertions(+), 29 deletions(-) diff --git a/configure b/configure index 43fa9a6632..04deb2ac80 100755 --- a/configure +++ b/configure @@ -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" diff --git a/doc/filters.texi b/doc/filters.texi index ba2ffc44ff..78fbd47767 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -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 diff --git a/libavfilter/af_sofalizer.c b/libavfilter/af_sofalizer.c index bcb35190ff..0aaae4baa8 100644 --- a/libavfilter/af_sofalizer.c +++ b/libavfilter/af_sofalizer.c @@ -28,12 +28,16 @@ #include #include +#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 } };