[FFmpeg-cvslog] avfilter/af_afir: switch to lavu/tx
Paul B Mahol
git at videolan.org
Sat Jan 29 12:58:25 EET 2022
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sat Jan 29 11:35:40 2022 +0100| [d388dc20b9dacb5775d701000f23bc78b7d21402] | committer: Paul B Mahol
avfilter/af_afir: switch to lavu/tx
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=d388dc20b9dacb5775d701000f23bc78b7d21402
---
libavfilter/af_afir.c | 98 +++++++++++++++++++++++++--------------------------
libavfilter/af_afir.h | 11 +++---
2 files changed, 56 insertions(+), 53 deletions(-)
diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c
index ace5087e90..7690218ff4 100644
--- a/libavfilter/af_afir.c
+++ b/libavfilter/af_afir.c
@@ -25,6 +25,7 @@
#include <float.h>
+#include "libavutil/tx.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
@@ -32,7 +33,6 @@
#include "libavutil/intreadwrite.h"
#include "libavutil/opt.h"
#include "libavutil/xga_font_data.h"
-#include "libavcodec/avfft.h"
#include "audio.h"
#include "avfilter.h"
@@ -58,7 +58,7 @@ static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t le
sum[2 * n] += t[2 * n] * c[2 * n];
}
-static void direct(const float *in, const FFTComplex *ir, int len, float *out)
+static void direct(const float *in, const AVComplexFloat *ir, int len, float *out)
{
for (int n = 0; n < len; n++)
for (int m = 0; m <= n; m++)
@@ -79,7 +79,7 @@ static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
{
AudioFIRContext *s = ctx->priv;
const float *in = (const float *)s->in->extended_data[ch] + offset;
- float *block, *buf, *ptr = (float *)out->extended_data[ch] + offset;
+ float *blockin, *blockout, *buf, *ptr = (float *)out->extended_data[ch] + offset;
const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset);
int n, i, j;
@@ -87,7 +87,8 @@ static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
AudioFIRSegment *seg = &s->seg[segment];
float *src = (float *)seg->input->extended_data[ch];
float *dst = (float *)seg->output->extended_data[ch];
- float *sum = (float *)seg->sum->extended_data[ch];
+ float *sumin = (float *)seg->sumin->extended_data[ch];
+ float *sumout = (float *)seg->sumout->extended_data[ch];
if (s->min_part_size >= 8) {
s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4));
@@ -115,7 +116,7 @@ static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
for (i = 0; i < seg->nb_partitions; i++) {
const int coffset = j * seg->coeff_size;
- const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
+ const AVComplexFloat *coeff = (const AVComplexFloat *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
direct(src, coeff, nb_samples, dst);
@@ -134,40 +135,38 @@ static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
continue;
}
- memset(sum, 0, sizeof(*sum) * seg->fft_length);
- block = (float *)seg->block->extended_data[ch] + seg->part_index[ch] * seg->block_size;
- memset(block + seg->part_size, 0, sizeof(*block) * (seg->fft_length - seg->part_size));
+ memset(sumin, 0, sizeof(*sumin) * seg->fft_length);
+ blockin = (float *)seg->blockin->extended_data[ch] + seg->part_index[ch] * seg->block_size;
+ blockout = (float *)seg->blockout->extended_data[ch] + seg->part_index[ch] * seg->block_size;
+ memset(blockin + seg->part_size, 0, sizeof(*blockin) * (seg->fft_length - seg->part_size));
- memcpy(block, src, sizeof(*src) * seg->part_size);
+ memcpy(blockin, src, sizeof(*src) * seg->part_size);
- av_rdft_calc(seg->rdft[ch], block);
- block[2 * seg->part_size] = block[1];
- block[1] = 0;
+ seg->tx_fn(seg->tx[ch], blockout, blockin, sizeof(float));
j = seg->part_index[ch];
for (i = 0; i < seg->nb_partitions; i++) {
const int coffset = j * seg->coeff_size;
- const float *block = (const float *)seg->block->extended_data[ch] + i * seg->block_size;
- const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
+ const float *blockout = (const float *)seg->blockout->extended_data[ch] + i * seg->block_size;
+ const AVComplexFloat *coeff = (const AVComplexFloat *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
- s->afirdsp.fcmul_add(sum, block, (const float *)coeff, seg->part_size);
+ s->afirdsp.fcmul_add(sumin, blockout, (const float *)coeff, seg->part_size);
if (j == 0)
j = seg->nb_partitions;
j--;
}
- sum[1] = sum[2 * seg->part_size];
- av_rdft_calc(seg->irdft[ch], sum);
+ seg->itx_fn(seg->itx[ch], sumout, sumin, sizeof(float));
buf = (float *)seg->buffer->extended_data[ch];
- fir_fadd(s, buf, sum, seg->part_size);
+ fir_fadd(s, buf, sumout, seg->part_size);
memcpy(dst, buf, seg->part_size * sizeof(*dst));
buf = (float *)seg->buffer->extended_data[ch];
- memcpy(buf, sum + seg->part_size, seg->part_size * sizeof(*buf));
+ memcpy(buf, sumout + seg->part_size, seg->part_size * sizeof(*buf));
seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
@@ -381,9 +380,9 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
{
AudioFIRContext *s = ctx->priv;
- seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));
- seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));
- if (!seg->rdft || !seg->irdft)
+ seg->tx = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->tx));
+ seg->itx = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->itx));
+ if (!seg->tx || !seg->itx)
return AVERROR(ENOMEM);
seg->fft_length = part_size * 2 + 1;
@@ -400,19 +399,22 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
return AVERROR(ENOMEM);
for (int ch = 0; ch < ctx->inputs[0]->channels && part_size >= 8; ch++) {
- seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C);
- seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R);
- if (!seg->rdft[ch] || !seg->irdft[ch])
+ float scale = 1.f, iscale = 1.f / part_size;
+ av_tx_init(&seg->tx[ch], &seg->tx_fn, AV_TX_FLOAT_RDFT, 0, 2 * part_size, &scale, 0);
+ av_tx_init(&seg->itx[ch], &seg->itx_fn, AV_TX_FLOAT_RDFT, 1, 2 * part_size, &iscale, 0);
+ if (!seg->tx[ch] || !seg->itx[ch])
return AVERROR(ENOMEM);
}
- seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
- seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
+ seg->sumin = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
+ seg->sumout = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
+ seg->blockin = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
+ seg->blockout = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
seg->coeff = ff_get_audio_buffer(ctx->inputs[1 + s->selir], seg->nb_partitions * seg->coeff_size * 2);
seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
- if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff || !seg->input || !seg->output)
+ if (!seg->buffer || !seg->sumin || !seg->sumout || !seg->blockin || !seg->blockout || !seg->coeff || !seg->input || !seg->output)
return AVERROR(ENOMEM);
return 0;
@@ -422,25 +424,27 @@ static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
{
AudioFIRContext *s = ctx->priv;
- if (seg->rdft) {
+ if (seg->tx) {
for (int ch = 0; ch < s->nb_channels; ch++) {
- av_rdft_end(seg->rdft[ch]);
+ av_tx_uninit(&seg->tx[ch]);
}
}
- av_freep(&seg->rdft);
+ av_freep(&seg->tx);
- if (seg->irdft) {
+ if (seg->itx) {
for (int ch = 0; ch < s->nb_channels; ch++) {
- av_rdft_end(seg->irdft[ch]);
+ av_tx_uninit(&seg->itx[ch]);
}
}
- av_freep(&seg->irdft);
+ av_freep(&seg->itx);
av_freep(&seg->output_offset);
av_freep(&seg->part_index);
- av_frame_free(&seg->block);
- av_frame_free(&seg->sum);
+ av_frame_free(&seg->blockin);
+ av_frame_free(&seg->blockout);
+ av_frame_free(&seg->sumin);
+ av_frame_free(&seg->sumout);
av_frame_free(&seg->buffer);
av_frame_free(&seg->coeff);
av_frame_free(&seg->input);
@@ -558,13 +562,13 @@ static int convert_coeffs(AVFilterContext *ctx)
for (int segment = 0; segment < s->nb_segments; segment++) {
AudioFIRSegment *seg = &s->seg[segment];
- float *block = (float *)seg->block->extended_data[ch];
- FFTComplex *coeff = (FFTComplex *)seg->coeff->extended_data[ch];
+ float *blockin = (float *)seg->blockin->extended_data[ch];
+ float *blockout = (float *)seg->blockout->extended_data[ch];
+ AVComplexFloat *coeff = (AVComplexFloat *)seg->coeff->extended_data[ch];
av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
for (i = 0; i < seg->nb_partitions; i++) {
- const float scale = 1.f / seg->part_size;
const int coffset = i * seg->coeff_size;
const int remaining = s->nb_taps - toffset;
const int size = remaining >= seg->part_size ? seg->part_size : remaining;
@@ -577,19 +581,15 @@ static int convert_coeffs(AVFilterContext *ctx)
continue;
}
- memset(block, 0, sizeof(*block) * seg->fft_length);
- memcpy(block, time + toffset, size * sizeof(*block));
+ memset(blockin, 0, sizeof(*blockin) * seg->fft_length);
+ memcpy(blockin, time + toffset, size * sizeof(*blockin));
- av_rdft_calc(seg->rdft[0], block);
+ seg->tx_fn(seg->tx[0], blockout, blockin, sizeof(float));
- coeff[coffset].re = block[0] * scale;
- coeff[coffset].im = 0;
- for (n = 1; n < seg->part_size; n++) {
- coeff[coffset + n].re = block[2 * n] * scale;
- coeff[coffset + n].im = block[2 * n + 1] * scale;
+ for (n = 0; n < seg->part_size + 1; n++) {
+ coeff[coffset + n].re = blockout[2 * n];
+ coeff[coffset + n].im = blockout[2 * n + 1];
}
- coeff[coffset + seg->part_size].re = block[1] * scale;
- coeff[coffset + seg->part_size].im = 0;
toffset += size;
}
diff --git a/libavfilter/af_afir.h b/libavfilter/af_afir.h
index 4f44675848..8f40c1b2f4 100644
--- a/libavfilter/af_afir.h
+++ b/libavfilter/af_afir.h
@@ -21,10 +21,10 @@
#ifndef AVFILTER_AFIR_H
#define AVFILTER_AFIR_H
+#include "libavutil/tx.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
-#include "libavcodec/avfft.h"
#include "audio.h"
#include "avfilter.h"
@@ -43,14 +43,17 @@ typedef struct AudioFIRSegment {
int *output_offset;
int *part_index;
- AVFrame *sum;
- AVFrame *block;
+ AVFrame *sumin;
+ AVFrame *sumout;
+ AVFrame *blockin;
+ AVFrame *blockout;
AVFrame *buffer;
AVFrame *coeff;
AVFrame *input;
AVFrame *output;
- RDFTContext **rdft, **irdft;
+ AVTXContext **tx, **itx;
+ av_tx_fn tx_fn, itx_fn;
} AudioFIRSegment;
typedef struct AudioFIRDSPContext {
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