From: Pavel Koshevoy <pkoshevoy@gmail.com>
To: FFmpeg development discussions and patches <ffmpeg-devel@ffmpeg.org>
Subject: Re: [FFmpeg-devel] [PATCH] avfilter/af_atempo: switch to rdft from lavu/tx
Date: Sun, 6 Feb 2022 11:04:09 -0700
Message-ID: <CAJgjuoxT2vVD=QS8DiS_1Twgcx7JVHoyXfSiG+yLs+iJ5vF+iA@mail.gmail.com> (raw)
In-Reply-To: <CAPYw7P5wykpQjnszfGHePmH45LUb9WzA=PLThvp+Fu3mkSnmJg@mail.gmail.com>
On Sun, Feb 6, 2022 at 10:24 AM Paul B Mahol <onemda@gmail.com> wrote:
> On Sun, Feb 6, 2022 at 6:16 PM Pavel Koshevoy <pkoshevoy@gmail.com> wrote:
>
> > On Sun, Feb 6, 2022 at 4:24 AM Paul B Mahol <onemda@gmail.com> wrote:
> >
> > > Signed-off-by: Paul B Mahol <onemda@gmail.com>
> > > ---
> > > configure | 3 -
> > > libavfilter/af_atempo.c | 126 ++++++++++++++++++++--------------------
> > > 2 files changed, 64 insertions(+), 65 deletions(-)
> > >
> > > diff --git a/configure b/configure
> > > index 5a8b52c77d..6ec25dd622 100755
> > > --- a/configure
> > > +++ b/configure
> > > @@ -3610,8 +3610,6 @@ amovie_filter_deps="avcodec avformat"
> > > aresample_filter_deps="swresample"
> > > asr_filter_deps="pocketsphinx"
> > > ass_filter_deps="libass"
> > > -atempo_filter_deps="avcodec"
> > > -atempo_filter_select="rdft"
> > > avgblur_opencl_filter_deps="opencl"
> > > avgblur_vulkan_filter_deps="vulkan spirv_compiler"
> > > azmq_filter_deps="libzmq"
> > > @@ -7387,7 +7385,6 @@ enabled zlib && add_cppflags -DZLIB_CONST
> > > # conditional library dependencies, in any order
> > > enabled amovie_filter && prepend avfilter_deps "avformat
> avcodec"
> > > enabled aresample_filter && prepend avfilter_deps "swresample"
> > > -enabled atempo_filter && prepend avfilter_deps "avcodec"
> > > enabled bm3d_filter && prepend avfilter_deps "avcodec"
> > > enabled cover_rect_filter && prepend avfilter_deps "avformat
> avcodec"
> > > enabled ebur128_filter && enabled swresample && prepend avfilter_deps
> > > "swresample"
> > > diff --git a/libavfilter/af_atempo.c b/libavfilter/af_atempo.c
> > > index e9a6da7970..27f2f6daa0 100644
> > > --- a/libavfilter/af_atempo.c
> > > +++ b/libavfilter/af_atempo.c
> > > @@ -39,13 +39,13 @@
> > > */
> > >
> > > #include <float.h>
> > > -#include "libavcodec/avfft.h"
> > > #include "libavutil/avassert.h"
> > > #include "libavutil/avstring.h"
> > > #include "libavutil/channel_layout.h"
> > > #include "libavutil/eval.h"
> > > #include "libavutil/opt.h"
> > > #include "libavutil/samplefmt.h"
> > > +#include "libavutil/tx.h"
> > > #include "avfilter.h"
> > > #include "audio.h"
> > > #include "internal.h"
> > > @@ -67,7 +67,8 @@ typedef struct AudioFragment {
> > >
> > > // rDFT transform of the down-mixed mono fragment, used for
> > > // fast waveform alignment via correlation in frequency domain:
> > > - FFTSample *xdat;
> > > + float *xdat_in;
> > > + float *xdat;
> > > } AudioFragment;
> > >
> > >
> > Is the old API being removed or deprecated?
> > Just wondering why this change is necessary.
> >
>
> New api is faster.
>
> >
> >
> >
> >
> > > /**
> > > @@ -140,9 +141,11 @@ typedef struct ATempoContext {
> > > FilterState state;
> > >
> > > // for fast correlation calculation in frequency domain:
> > > - RDFTContext *real_to_complex;
> > > - RDFTContext *complex_to_real;
> > > - FFTSample *correlation;
> > > + AVTXContext *real_to_complex;
> > > + AVTXContext *complex_to_real;
> > > + av_tx_fn r2c_fn, c2r_fn;
> > > + float *correlation_in;
> > > + float *correlation;
> > >
> > > // for managing AVFilterPad.request_frame and
> > AVFilterPad.filter_frame
> > > AVFrame *dst_buffer;
> > > @@ -228,18 +231,18 @@ static void yae_release_buffers(ATempoContext
> > > *atempo)
> > >
> > > av_freep(&atempo->frag[0].data);
> > > av_freep(&atempo->frag[1].data);
> > > + av_freep(&atempo->frag[0].xdat_in);
> > > + av_freep(&atempo->frag[1].xdat_in);
> > > av_freep(&atempo->frag[0].xdat);
> > > av_freep(&atempo->frag[1].xdat);
> > >
> > > av_freep(&atempo->buffer);
> > > av_freep(&atempo->hann);
> > > + av_freep(&atempo->correlation_in);
> > > av_freep(&atempo->correlation);
> > >
> > > - av_rdft_end(atempo->real_to_complex);
> > > - atempo->real_to_complex = NULL;
> > > -
> > > - av_rdft_end(atempo->complex_to_real);
> > > - atempo->complex_to_real = NULL;
> > > + av_tx_uninit(&atempo->real_to_complex);
> > > + av_tx_uninit(&atempo->complex_to_real);
> > > }
> > >
> > > /* av_realloc is not aligned enough; fortunately, the data does not
> need
> > > to
> > > @@ -247,7 +250,7 @@ static void yae_release_buffers(ATempoContext
> > *atempo)
> > > #define RE_MALLOC_OR_FAIL(field, field_size) \
> > > do { \
> > > av_freep(&field); \
> > > - field = av_malloc(field_size); \
> > > + field = av_calloc(field_size, 1); \
> > > if (!field) { \
> > > yae_release_buffers(atempo); \
> > > return AVERROR(ENOMEM); \
> > > @@ -265,6 +268,7 @@ static int yae_reset(ATempoContext *atempo,
> > > {
> > > const int sample_size = av_get_bytes_per_sample(format);
> > > uint32_t nlevels = 0;
> > > + float scale = 1.f, iscale = 1.f;
> > > uint32_t pot;
> > > int i;
> > >
> > > @@ -288,29 +292,29 @@ static int yae_reset(ATempoContext *atempo,
> > > // initialize audio fragment buffers:
> > > RE_MALLOC_OR_FAIL(atempo->frag[0].data, atempo->window *
> > > atempo->stride);
> > > RE_MALLOC_OR_FAIL(atempo->frag[1].data, atempo->window *
> > > atempo->stride);
> > > - RE_MALLOC_OR_FAIL(atempo->frag[0].xdat, atempo->window *
> > > sizeof(FFTComplex));
> > > - RE_MALLOC_OR_FAIL(atempo->frag[1].xdat, atempo->window *
> > > sizeof(FFTComplex));
> > > + RE_MALLOC_OR_FAIL(atempo->frag[0].xdat_in, (atempo->window + 1) *
> > > sizeof(AVComplexFloat));
> > > + RE_MALLOC_OR_FAIL(atempo->frag[1].xdat_in, (atempo->window + 1) *
> > > sizeof(AVComplexFloat));
> > > + RE_MALLOC_OR_FAIL(atempo->frag[0].xdat, (atempo->window + 1) *
> > > sizeof(AVComplexFloat));
> > > + RE_MALLOC_OR_FAIL(atempo->frag[1].xdat, (atempo->window + 1) *
> > > sizeof(AVComplexFloat));
> > >
> > > // initialize rDFT contexts:
> > > - av_rdft_end(atempo->real_to_complex);
> > > - atempo->real_to_complex = NULL;
> > > -
> > > - av_rdft_end(atempo->complex_to_real);
> > > - atempo->complex_to_real = NULL;
> > > + av_tx_uninit(&atempo->real_to_complex);
> > > + av_tx_uninit(&atempo->complex_to_real);
> > >
> > > - atempo->real_to_complex = av_rdft_init(nlevels + 1, DFT_R2C);
> > > + av_tx_init(&atempo->real_to_complex, &atempo->r2c_fn,
> > > AV_TX_FLOAT_RDFT, 0, 1 << (nlevels + 1), &scale, 0);
> > > if (!atempo->real_to_complex) {
> > > yae_release_buffers(atempo);
> > > return AVERROR(ENOMEM);
> > > }
> > >
> > > - atempo->complex_to_real = av_rdft_init(nlevels + 1, IDFT_C2R);
> > > + av_tx_init(&atempo->complex_to_real, &atempo->c2r_fn,
> > > AV_TX_FLOAT_RDFT, 1, 1 << (nlevels + 1), &iscale, 0);
> > > if (!atempo->complex_to_real) {
> > > yae_release_buffers(atempo);
> > > return AVERROR(ENOMEM);
> > > }
> > >
> > > - RE_MALLOC_OR_FAIL(atempo->correlation, atempo->window *
> > > sizeof(FFTComplex));
> > > + RE_MALLOC_OR_FAIL(atempo->correlation_in, (atempo->window + 1) *
> > > sizeof(AVComplexFloat));
> > > + RE_MALLOC_OR_FAIL(atempo->correlation, atempo->window *
> > > sizeof(AVComplexFloat));
> > >
> > > atempo->ring = atempo->window * 3;
> > > RE_MALLOC_OR_FAIL(atempo->buffer, atempo->ring * atempo->stride);
> > > @@ -348,7 +352,7 @@ static int yae_update(AVFilterContext *ctx)
> > > const uint8_t *src_end = src +
> > \
> > > frag->nsamples * atempo->channels * sizeof(scalar_type);
> > \
> > >
> > \
> > > - FFTSample *xdat = frag->xdat;
> > \
> > > + float *xdat = frag->xdat_in;
> > \
> > > scalar_type tmp;
> > \
> > >
> > \
> > > if (atempo->channels == 1) {
> > \
> > > @@ -356,27 +360,27 @@ static int yae_update(AVFilterContext *ctx)
> > > tmp = *(const scalar_type *)src;
> > \
> > > src += sizeof(scalar_type);
> > \
> > >
> > \
> > > - *xdat = (FFTSample)tmp;
> > \
> > > + *xdat = (float)tmp;
> > \
> > > }
> > \
> > > } else {
> > \
> > > - FFTSample s, max, ti, si;
> > \
> > > + float s, max, ti, si;
> > \
> > > int i;
> > \
> > >
> > \
> > > for (; src < src_end; xdat++) {
> > \
> > > tmp = *(const scalar_type *)src;
> > \
> > > src += sizeof(scalar_type);
> > \
> > >
> > \
> > > - max = (FFTSample)tmp;
> > \
> > > - s = FFMIN((FFTSample)scalar_max,
> > \
> > > - (FFTSample)fabsf(max));
> > \
> > > + max = (float)tmp;
> > \
> > > + s = FFMIN((float)scalar_max,
> > \
> > > + (float)fabsf(max));
> > \
> > >
> > \
> > > for (i = 1; i < atempo->channels; i++) {
> > \
> > > tmp = *(const scalar_type *)src;
> > \
> > > src += sizeof(scalar_type);
> > \
> > >
> > \
> > > - ti = (FFTSample)tmp;
> > \
> > > - si = FFMIN((FFTSample)scalar_max,
> > \
> > > - (FFTSample)fabsf(ti));
> > \
> > > + ti = (float)tmp;
> > \
> > > + si = FFMIN((float)scalar_max,
> > \
> > > + (float)fabsf(ti));
> > \
> > >
> > \
> > > if (s < si) {
> > \
> > > s = si;
> > \
> > > @@ -399,7 +403,7 @@ static void yae_downmix(ATempoContext *atempo,
> > > AudioFragment *frag)
> > > const uint8_t *src = frag->data;
> > >
> > > // init complex data buffer used for FFT and Correlation:
> > > - memset(frag->xdat, 0, sizeof(FFTComplex) * atempo->window);
> > > + memset(frag->xdat_in, 0, sizeof(AVComplexFloat) * atempo->window);
> > >
> > > if (atempo->format == AV_SAMPLE_FMT_U8) {
> > > yae_init_xdat(uint8_t, 127);
> > > @@ -598,32 +602,24 @@ static void
> yae_advance_to_next_frag(ATempoContext
> > > *atempo)
> > > * Multiply two vectors of complex numbers (result of real_to_complex
> > > rDFT)
> > > * and transform back via complex_to_real rDFT.
> > > */
> > > -static void yae_xcorr_via_rdft(FFTSample *xcorr,
> > > - RDFTContext *complex_to_real,
> > > - const FFTComplex *xa,
> > > - const FFTComplex *xb,
> > > +static void yae_xcorr_via_rdft(float *xcorr_in,
> > > + float *xcorr,
> > > + AVTXContext *complex_to_real,
> > > + av_tx_fn c2r_fn,
> > > + const AVComplexFloat *xa,
> > > + const AVComplexFloat *xb,
> > > const int window)
> > > {
> > > - FFTComplex *xc = (FFTComplex *)xcorr;
> > > + AVComplexFloat *xc = (AVComplexFloat *)xcorr_in;
> > > int i;
> > >
> > > - // NOTE: first element requires special care -- Given Y = rDFT(X),
> > > - // Im(Y[0]) and Im(Y[N/2]) are always zero, therefore av_rdft_calc
> > > - // stores Re(Y[N/2]) in place of Im(Y[0]).
> > > -
> > > - xc->re = xa->re * xb->re;
> > > - xc->im = xa->im * xb->im;
> > > - xa++;
> > > - xb++;
> > > - xc++;
> > > -
> > > - for (i = 1; i < window; i++, xa++, xb++, xc++) {
> > > + for (i = 0; i <= window; i++, xa++, xb++, xc++) {
> > >
> >
> > This used to iterate over [1, window - 1] elements.
> > Now it iterates over [0, window] elements.
> > Is this correct? That's 2 additional elements.
> >
>
> Yes, newer api does not use previous api hack of producing n/2 complex
> numbers, but n/2 +1.
>
cool, thanks ... lgtm if it still works
>
> >
> >
> >
> > > xc->re = (xa->re * xb->re + xa->im * xb->im);
> > > xc->im = (xa->im * xb->re - xa->re * xb->im);
> > > }
> > >
> > > // apply inverse rDFT:
> > > - av_rdft_calc(complex_to_real, xcorr);
> > > + c2r_fn(complex_to_real, xcorr, xcorr_in, sizeof(float));
> > > }
> > >
> > > /**
> > > @@ -637,21 +633,25 @@ static int yae_align(AudioFragment *frag,
> > > const int window,
> > > const int delta_max,
> > > const int drift,
> > > - FFTSample *correlation,
> > > - RDFTContext *complex_to_real)
> > > + float *correlation_in,
> > > + float *correlation,
> > > + AVTXContext *complex_to_real,
> > > + av_tx_fn c2r_fn)
> > > {
> > > int best_offset = -drift;
> > > - FFTSample best_metric = -FLT_MAX;
> > > - FFTSample *xcorr;
> > > + float best_metric = -FLT_MAX;
> > > + float *xcorr;
> > >
> > > int i0;
> > > int i1;
> > > int i;
> > >
> > > - yae_xcorr_via_rdft(correlation,
> > > + yae_xcorr_via_rdft(correlation_in,
> > > + correlation,
> > > complex_to_real,
> > > - (const FFTComplex *)prev->xdat,
> > > - (const FFTComplex *)frag->xdat,
> > > + c2r_fn,
> > > + (const AVComplexFloat *)prev->xdat,
> > > + (const AVComplexFloat *)frag->xdat,
> > > window);
> > >
> > > // identify search window boundaries:
> > > @@ -665,11 +665,11 @@ static int yae_align(AudioFragment *frag,
> > > xcorr = correlation + i0;
> > >
> > > for (i = i0; i < i1; i++, xcorr++) {
> > > - FFTSample metric = *xcorr;
> > > + float metric = *xcorr;
> > >
> > > // normalize:
> > > - FFTSample drifti = (FFTSample)(drift + i);
> > > - metric *= drifti * (FFTSample)(i - i0) * (FFTSample)(i1 - i);
> > > + float drifti = (float)(drift + i);
> > > + metric *= drifti * (float)(i - i0) * (float)(i1 - i);
> > >
> > > if (metric > best_metric) {
> > > best_metric = metric;
> > > @@ -706,8 +706,10 @@ static int yae_adjust_position(ATempoContext
> > *atempo)
> > > atempo->window,
> > > delta_max,
> > > drift,
> > > + atempo->correlation_in,
> > > atempo->correlation,
> > > - atempo->complex_to_real);
> > > + atempo->complex_to_real,
> > > + atempo->c2r_fn);
> > >
> > > if (correction) {
> > > // adjust fragment position:
> > > @@ -833,7 +835,7 @@ yae_apply(ATempoContext *atempo,
> > > yae_downmix(atempo, yae_curr_frag(atempo));
> > >
> > > // apply rDFT:
> > > - av_rdft_calc(atempo->real_to_complex,
> > > yae_curr_frag(atempo)->xdat);
> > > + atempo->r2c_fn(atempo->real_to_complex,
> > > yae_curr_frag(atempo)->xdat, yae_curr_frag(atempo)->xdat_in,
> > sizeof(float));
> > >
> > > // must load the second fragment before alignment can
> start:
> > > if (!atempo->nfrag) {
> > > @@ -865,7 +867,7 @@ yae_apply(ATempoContext *atempo,
> > > yae_downmix(atempo, yae_curr_frag(atempo));
> > >
> > > // apply rDFT:
> > > - av_rdft_calc(atempo->real_to_complex,
> > > yae_curr_frag(atempo)->xdat);
> > > + atempo->r2c_fn(atempo->real_to_complex,
> > > yae_curr_frag(atempo)->xdat, yae_curr_frag(atempo)->xdat_in,
> > sizeof(float));
> > >
> > > atempo->state = YAE_OUTPUT_OVERLAP_ADD;
> > > }
> > > @@ -929,7 +931,7 @@ static int yae_flush(ATempoContext *atempo,
> > > yae_downmix(atempo, frag);
> > >
> > > // apply rDFT:
> > > - av_rdft_calc(atempo->real_to_complex, frag->xdat);
> > > + atempo->r2c_fn(atempo->real_to_complex, frag->xdat,
> > > frag->xdat_in, sizeof(float));
> > >
> > > // align current fragment to previous fragment:
> > > if (yae_adjust_position(atempo)) {
> > > --
> > > 2.33.0
> > >
> > > _______________________________________________
> > > ffmpeg-devel mailing list
> > > ffmpeg-devel@ffmpeg.org
> > > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> > >
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> > > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
> > >
> > _______________________________________________
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> > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> >
> > To unsubscribe, visit link above, or email
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> >
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next prev parent reply other threads:[~2022-02-06 18:04 UTC|newest]
Thread overview: 5+ messages / expand[flat|nested] mbox.gz Atom feed top
2022-02-06 11:25 Paul B Mahol
2022-02-06 17:15 ` Pavel Koshevoy
2022-02-06 17:24 ` Paul B Mahol
2022-02-06 18:04 ` Pavel Koshevoy [this message]
2022-02-06 19:02 ` Lynne
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