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avfilter/af_afir: introduce AudioFIRSegment structure and use it

This commit is contained in:
Paul B Mahol 2018-12-28 17:57:24 +01:00
parent fccba32b4c
commit 7312e027d6
2 changed files with 86 additions and 103 deletions
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161
libavfilter/af_afir.c
View File

@ -60,41 +60,41 @@ static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
{
AudioFIRContext *s = ctx->priv;
const float *src = (const float *)s->in[0]->extended_data[ch];
float *sum = s->sum[ch];
float *sum = (float *)s->seg.sum->extended_data[ch];
AVFrame *out = arg;
float *block, *dst, *ptr;
int n, i, j;
memset(sum, 0, sizeof(*sum) * s->fft_length);
block = s->block[ch] + s->part_index * s->block_size;
memset(block, 0, sizeof(*block) * s->fft_length);
memset(sum, 0, sizeof(*sum) * s->seg.fft_length);
block = (float *)s->seg.block->extended_data[ch] + s->seg.part_index * s->seg.block_size;
memset(block, 0, sizeof(*block) * s->seg.fft_length);
s->fdsp->vector_fmul_scalar(block, src, s->dry_gain, FFALIGN(out->nb_samples, 4));
emms_c();
av_rdft_calc(s->rdft[ch], block);
block[2 * s->part_size] = block[1];
av_rdft_calc(s->seg.rdft[ch], block);
block[2 * s->seg.part_size] = block[1];
block[1] = 0;
j = s->part_index;
j = s->seg.part_index;
for (i = 0; i < s->nb_partitions; i++) {
const int coffset = i * s->coeff_size;
const FFTComplex *coeff = s->coeff[ch * !s->one2many] + coffset;
for (i = 0; i < s->seg.nb_partitions; i++) {
const int coffset = i * s->seg.coeff_size;
const float *block = (const float *)s->seg.block->extended_data[ch] + j * s->seg.block_size;
const FFTComplex *coeff = s->seg.coeff[ch * !s->one2many] + coffset;
block = s->block[ch] + j * s->block_size;
s->fcmul_add(sum, block, (const float *)coeff, s->part_size);
s->fcmul_add(sum, block, (const float *)coeff, s->seg.part_size);
if (j == 0)
j = s->nb_partitions;
j = s->seg.nb_partitions;
j--;
}
sum[1] = sum[2 * s->part_size];
av_rdft_calc(s->irdft[ch], sum);
sum[1] = sum[2 * s->seg.part_size];
av_rdft_calc(s->seg.irdft[ch], sum);
dst = (float *)s->buffer->extended_data[ch];
for (n = 0; n < s->part_size; n++) {
dst = (float *)s->seg.buffer->extended_data[ch];
for (n = 0; n < s->seg.part_size; n++) {
dst[n] += sum[n];
}
@ -102,8 +102,8 @@ static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
s->fdsp->vector_fmul_scalar(ptr, dst, s->wet_gain, FFALIGN(out->nb_samples, 4));
emms_c();
dst = (float *)s->buffer->extended_data[ch];
memcpy(dst, sum + s->part_size, s->part_size * sizeof(*dst));
dst = (float *)s->seg.buffer->extended_data[ch];
memcpy(dst, sum + s->seg.part_size, s->seg.part_size * sizeof(*dst));
return 0;
}
@ -124,7 +124,7 @@ static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
s->in[0] = in;
ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
s->part_index = (s->part_index + 1) % s->nb_partitions;
s->seg.part_index = (s->seg.part_index + 1) % s->seg.nb_partitions;
out->pts = s->pts;
if (s->pts != AV_NOPTS_VALUE)
@ -286,39 +286,29 @@ static int convert_coeffs(AVFilterContext *ctx)
for (n = av_log2(s->minp); (1 << n) < s->nb_taps; n++);
N = FFMIN(n, av_log2(s->maxp));
s->fft_length = (1 << (N + 1)) + 1;
s->part_size = 1 << (N - 1);
s->block_size = FFALIGN(s->fft_length, 32);
s->coeff_size = FFALIGN(s->part_size + 1, 32);
s->nb_partitions = (s->nb_taps + s->part_size - 1) / s->part_size;
for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
s->sum[ch] = av_calloc(s->fft_length, sizeof(**s->sum));
if (!s->sum[ch])
return AVERROR(ENOMEM);
}
s->seg.fft_length = (1 << (N + 1)) + 1;
s->seg.part_size = 1 << (N - 1);
s->seg.block_size = FFALIGN(s->seg.fft_length, 32);
s->seg.coeff_size = FFALIGN(s->seg.part_size + 1, 32);
s->seg.nb_partitions = (s->nb_taps + s->seg.part_size - 1) / s->seg.part_size;
for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
s->coeff[ch] = av_calloc(s->nb_partitions * s->coeff_size, sizeof(**s->coeff));
if (!s->coeff[ch])
s->seg.coeff[ch] = av_calloc(s->seg.nb_partitions * s->seg.coeff_size, sizeof(**s->seg.coeff));
if (!s->seg.coeff[ch])
return AVERROR(ENOMEM);
}
for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
s->block[ch] = av_calloc(s->nb_partitions * s->block_size, sizeof(**s->block));
if (!s->block[ch])
s->seg.rdft[ch] = av_rdft_init(N, DFT_R2C);
s->seg.irdft[ch] = av_rdft_init(N, IDFT_C2R);
if (!s->seg.rdft[ch] || !s->seg.irdft[ch])
return AVERROR(ENOMEM);
}
for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
s->rdft[ch] = av_rdft_init(N, DFT_R2C);
s->irdft[ch] = av_rdft_init(N, IDFT_C2R);
if (!s->rdft[ch] || !s->irdft[ch])
return AVERROR(ENOMEM);
}
s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size);
if (!s->buffer)
s->seg.sum = ff_get_audio_buffer(ctx->inputs[0], s->seg.fft_length);
s->seg.block = ff_get_audio_buffer(ctx->inputs[0], s->seg.nb_partitions * s->seg.block_size);
s->seg.buffer = ff_get_audio_buffer(ctx->inputs[0], s->seg.part_size);
if (!s->seg.buffer || !s->seg.sum || !s->seg.block)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]);
@ -377,40 +367,40 @@ static int convert_coeffs(AVFilterContext *ctx)
for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
float *block = s->block[ch];
FFTComplex *coeff = s->coeff[ch];
float *block = (float *)s->seg.block->extended_data[ch];
FFTComplex *coeff = s->seg.coeff[ch];
for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
time[i] = 0;
for (i = 0; i < s->nb_partitions; i++) {
const float scale = 1.f / s->part_size;
const int toffset = i * s->part_size;
const int coffset = i * s->coeff_size;
const int remaining = s->nb_taps - (i * s->part_size);
const int size = remaining >= s->part_size ? s->part_size : remaining;
for (i = 0; i < s->seg.nb_partitions; i++) {
const float scale = 1.f / s->seg.part_size;
const int toffset = i * s->seg.part_size;
const int coffset = i * s->seg.coeff_size;
const int remaining = s->nb_taps - (i * s->seg.part_size);
const int size = remaining >= s->seg.part_size ? s->seg.part_size : remaining;
memset(block, 0, sizeof(*block) * s->fft_length);
memset(block, 0, sizeof(*block) * s->seg.fft_length);
memcpy(block, time + toffset, size * sizeof(*block));
av_rdft_calc(s->rdft[0], block);
av_rdft_calc(s->seg.rdft[0], block);
coeff[coffset].re = block[0] * scale;
coeff[coffset].im = 0;
for (n = 1; n < s->part_size; n++) {
for (n = 1; n < s->seg.part_size; n++) {
coeff[coffset + n].re = block[2 * n] * scale;
coeff[coffset + n].im = block[2 * n + 1] * scale;
}
coeff[coffset + s->part_size].re = block[1] * scale;
coeff[coffset + s->part_size].im = 0;
coeff[coffset + s->seg.part_size].re = block[1] * scale;
coeff[coffset + s->seg.part_size].im = 0;
}
}
av_frame_free(&s->in[1]);
av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions);
av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->part_size);
av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", s->fft_length);
av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->seg.nb_partitions);
av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->seg.part_size);
av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", s->seg.fft_length);
s->have_coeffs = 1;
@ -469,7 +459,7 @@ static int activate(AVFilterContext *ctx)
return ret;
}
ret = ff_inlink_consume_samples(ctx->inputs[0], s->part_size, s->part_size, &in);
ret = ff_inlink_consume_samples(ctx->inputs[0], s->seg.part_size, s->seg.part_size, &in);
if (ret > 0)
ret = fir_frame(s, in, outlink);
@ -486,7 +476,7 @@ static int activate(AVFilterContext *ctx)
}
}
if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->part_size) {
if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->seg.part_size) {
ff_filter_set_ready(ctx, 10);
return 0;
}
@ -580,12 +570,10 @@ static int config_output(AVFilterLink *outlink)
outlink->channel_layout = ctx->inputs[0]->channel_layout;
outlink->channels = ctx->inputs[0]->channels;
s->sum = av_calloc(outlink->channels, sizeof(*s->sum));
s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff));
s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block));
s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft));
s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft));
if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft)
s->seg.coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->seg.coeff));
s->seg.rdft = av_calloc(outlink->channels, sizeof(*s->seg.rdft));
s->seg.irdft = av_calloc(outlink->channels, sizeof(*s->seg.irdft));
if (!s->seg.coeff || !s->seg.rdft || !s->seg.irdft)
return AVERROR(ENOMEM);
s->nb_channels = outlink->channels;
@ -600,43 +588,32 @@ static av_cold void uninit(AVFilterContext *ctx)
AudioFIRContext *s = ctx->priv;
int ch;
if (s->sum) {
for (ch = 0; ch < s->nb_channels; ch++) {
av_freep(&s->sum[ch]);
}
}
av_freep(&s->sum);
if (s->coeff) {
if (s->seg.coeff) {
for (ch = 0; ch < s->nb_coef_channels; ch++) {
av_freep(&s->coeff[ch]);
av_freep(&s->seg.coeff[ch]);
}
}
av_freep(&s->coeff);
av_freep(&s->seg.coeff);
if (s->block) {
if (s->seg.rdft) {
for (ch = 0; ch < s->nb_channels; ch++) {
av_freep(&s->block[ch]);
av_rdft_end(s->seg.rdft[ch]);
}
}
av_freep(&s->block);
av_freep(&s->seg.rdft);
if (s->rdft) {
if (s->seg.irdft) {
for (ch = 0; ch < s->nb_channels; ch++) {
av_rdft_end(s->rdft[ch]);
av_rdft_end(s->seg.irdft[ch]);
}
}
av_freep(&s->rdft);
if (s->irdft) {
for (ch = 0; ch < s->nb_channels; ch++) {
av_rdft_end(s->irdft[ch]);
}
}
av_freep(&s->irdft);
av_freep(&s->seg.irdft);
av_frame_free(&s->in[1]);
av_frame_free(&s->buffer);
av_frame_free(&s->seg.block);
av_frame_free(&s->seg.sum);
av_frame_free(&s->seg.buffer);
av_freep(&s->fdsp);

28
libavfilter/af_afir.h
View File

@ -31,6 +31,22 @@
#include "formats.h"
#include "internal.h"
typedef struct AudioFIRSegment {
int nb_partitions;
int part_index;
int part_size;
int block_size;
int fft_length;
int coeff_size;
AVFrame *sum;
AVFrame *block;
AVFrame *buffer;
RDFTContext **rdft, **irdft;
FFTComplex **coeff;
} AudioFIRSegment;
typedef struct AudioFIRContext {
const AVClass *class;
@ -53,23 +69,13 @@ typedef struct AudioFIRContext {
int eof_coeffs;
int have_coeffs;
int nb_taps;
int part_size;
int part_index;
int coeff_size;
int block_size;
int nb_partitions;
int nb_channels;
int fft_length;
int nb_coef_channels;
int one2many;
RDFTContext **rdft, **irdft;
float **sum;
float **block;
FFTComplex **coeff;
AudioFIRSegment seg;
AVFrame *in[2];
AVFrame *buffer;
AVFrame *video;
int64_t pts;