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avfilter/af_afir: make number of segments extendable

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This commit is contained in:
Paul B Mahol 2018-12-29 10:39:19 +01:00
parent 6095356d5b
commit 31c9d693aa
2 changed files with 48 additions and 30 deletions
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75
libavfilter/af_afir.c
View File

@ -59,7 +59,7 @@ static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t le
static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
{ {
AudioFIRContext *s = ctx->priv; AudioFIRContext *s = ctx->priv;
AudioFIRSegment *seg = &s->seg; AudioFIRSegment *seg = &s->seg[0];
const float *src = (const float *)s->in[0]->extended_data[ch]; const float *src = (const float *)s->in[0]->extended_data[ch];
float *sum = (float *)seg->sum->extended_data[ch]; float *sum = (float *)seg->sum->extended_data[ch];
AVFrame *out = arg; AVFrame *out = arg;
@ -125,7 +125,11 @@ static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
s->in[0] = in; s->in[0] = in;
ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels); ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
s->seg.part_index = (s->seg.part_index + 1) % s->seg.nb_partitions; for (int segment = 0; segment < s->nb_segments; segment++) {
AudioFIRSegment *seg = &s->seg[segment];
seg->part_index = (seg->part_index + 1) % seg->nb_partitions;
}
out->pts = s->pts; out->pts = s->pts;
if (s->pts != AV_NOPTS_VALUE) if (s->pts != AV_NOPTS_VALUE)
@ -319,7 +323,8 @@ static int convert_coeffs(AVFilterContext *ctx)
for (n = av_log2(s->minp); (1 << n) < s->nb_taps; n++); for (n = av_log2(s->minp); (1 << n) < s->nb_taps; n++);
N = FFMIN(n, av_log2(s->maxp)); N = FFMIN(n, av_log2(s->maxp));
ret = init_segment(ctx, &s->seg, (s->nb_taps + (1 << N) - 1) / (1 << N), 1 << N); s->nb_segments = 1;
ret = init_segment(ctx, &s->seg[0], (s->nb_taps + (1 << N) - 1) / (1 << N), 1 << N);
if (ret < 0) if (ret < 0)
return ret; return ret;
@ -377,43 +382,53 @@ static int convert_coeffs(AVFilterContext *ctx)
s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4)); s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4));
} }
av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
float *time = (float *)s->in[1]->extended_data[!s->one2many * ch]; float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
float *block = (float *)s->seg.block->extended_data[ch];
FFTComplex *coeff = (FFTComplex *)s->seg.coeff->extended_data[ch];
for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++) for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
time[i] = 0; time[i] = 0;
for (i = 0; i < s->seg.nb_partitions; i++) { av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
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->seg.fft_length); for (int segment = 0; segment < s->nb_segments; segment++) {
memcpy(block, time + toffset, size * sizeof(*block)); AudioFIRSegment *seg = &s->seg[segment];
float *block = (float *)seg->block->extended_data[ch];
FFTComplex *coeff = (FFTComplex *)seg->coeff->extended_data[ch];
av_rdft_calc(s->seg.rdft[0], block); av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
coeff[coffset].re = block[0] * scale; for (i = 0; i < seg->nb_partitions; i++) {
coeff[coffset].im = 0; const float scale = 1.f / seg->part_size;
for (n = 1; n < s->seg.part_size; n++) { const int toffset = i * seg->part_size;
coeff[coffset + n].re = block[2 * n] * scale; const int coffset = i * seg->coeff_size;
coeff[coffset + n].im = block[2 * n + 1] * scale; const int remaining = s->nb_taps - (i * seg->part_size);
const int size = remaining >= seg->part_size ? seg->part_size : remaining;
memset(block, 0, sizeof(*block) * seg->fft_length);
memcpy(block, time + toffset, size * sizeof(*block));
av_rdft_calc(seg->rdft[0], block);
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;
}
coeff[coffset + seg->part_size].re = block[1] * scale;
coeff[coffset + seg->part_size].im = 0;
} }
coeff[coffset + s->seg.part_size].re = block[1] * scale;
coeff[coffset + s->seg.part_size].im = 0; av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
} }
} }
av_frame_free(&s->in[1]); 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->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; s->have_coeffs = 1;
return 0; return 0;
@ -471,7 +486,7 @@ static int activate(AVFilterContext *ctx)
return ret; return ret;
} }
ret = ff_inlink_consume_samples(ctx->inputs[0], s->seg.part_size, s->seg.part_size, &in); ret = ff_inlink_consume_samples(ctx->inputs[0], s->seg[0].part_size, s->seg[0].part_size, &in);
if (ret > 0) if (ret > 0)
ret = fir_frame(s, in, outlink); ret = fir_frame(s, in, outlink);
@ -488,7 +503,7 @@ static int activate(AVFilterContext *ctx)
} }
} }
if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->seg.part_size) { if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->seg[0].part_size) {
ff_filter_set_ready(ctx, 10); ff_filter_set_ready(ctx, 10);
return 0; return 0;
} }
@ -617,7 +632,9 @@ static av_cold void uninit(AVFilterContext *ctx)
{ {
AudioFIRContext *s = ctx->priv; AudioFIRContext *s = ctx->priv;
uninit_segment(ctx, &s->seg); for (int i = 0; i < s->nb_segments; i++) {
uninit_segment(ctx, &s->seg[i]);
}
av_freep(&s->fdsp); av_freep(&s->fdsp);
av_frame_free(&s->in[1]); av_frame_free(&s->in[1]);

3
libavfilter/af_afir.h
View File

@ -74,7 +74,8 @@ typedef struct AudioFIRContext {
int nb_coef_channels; int nb_coef_channels;
int one2many; int one2many;
AudioFIRSegment seg; AudioFIRSegment seg[1024];
int nb_segments;
AVFrame *in[2]; AVFrame *in[2];
AVFrame *video; AVFrame *video;