[FFmpeg-cvslog] avfilter/af_afir: implement non-uniform partitioned convolution
Paul B Mahol
git at videolan.org
Sun Dec 30 14:43:02 EET 2018
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sat Dec 29 11:40:13 2018 +0100| [a53a9f1c8d1ffad80956027ffee5f506c98f28ff] | committer: Paul B Mahol
avfilter/af_afir: implement non-uniform partitioned convolution
Using multiple frequency delay lines.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=a53a9f1c8d1ffad80956027ffee5f506c98f28ff
---
doc/filters.texi | 4 +-
libavfilter/af_afir.c | 180 +++++++++++++++++++++++++++++++++-----------------
libavfilter/af_afir.h | 7 +-
3 files changed, 129 insertions(+), 62 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index 45d90f6165..b3e2081bcf 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1231,14 +1231,14 @@ Set video stream size. This option is used only when @var{response} is enabled.
Set video stream frame rate. This option is used only when @var{response} is enabled.
@item minp
-Set minimal partition size used for convolution. Default is @var{16}.
+Set minimal partition size used for convolution. Default is @var{8192}.
Allowed range is from @var{16} to @var{32768}.
Lower values decreases latency at cost of higher CPU usage.
@item maxp
Set maximal partition size used for convolution. Default is @var{8192}.
Allowed range is from @var{16} to @var{32768}.
-Lower values decreases latency at cost of higher CPU usage.
+Lower values may increase CPU usage.
@end table
@subsection Examples
diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c
index 3dda875e35..8271c2cfce 100644
--- a/libavfilter/af_afir.c
+++ b/libavfilter/af_afir.c
@@ -59,54 +59,84 @@ 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)
{
AudioFIRContext *s = ctx->priv;
- AudioFIRSegment *seg = &s->seg[0];
- const float *src = (const float *)s->in[0]->extended_data[ch];
- float *sum = (float *)seg->sum->extended_data[ch];
+ const float *in = (const float *)s->in[0]->extended_data[ch];
AVFrame *out = arg;
- float *block, *dst, *ptr;
+ float *block, *buf, *ptr = (float *)out->extended_data[ch];
int n, i, j;
- memset(sum, 0, sizeof(*sum) * seg->fft_length);
- block = (float *)seg->block->extended_data[ch] + seg->part_index[ch] * seg->block_size;
- memset(block, 0, sizeof(*block) * seg->fft_length);
+ for (int segment = 0; segment < s->nb_segments; segment++) {
+ 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];
+
+ s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(out->nb_samples, 4));
+ emms_c();
+
+ seg->output_offset[ch] += s->min_part_size;
+ if (seg->output_offset[ch] == seg->part_size) {
+ seg->output_offset[ch] = 0;
+ memset(dst, 0, sizeof(*dst) * seg->part_size);
+ } else {
+ memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
+
+ dst += seg->output_offset[ch];
+ for (n = 0; n < out->nb_samples; n++) {
+ ptr[n] += dst[n];
+ }
+ continue;
+ }
- s->fdsp->vector_fmul_scalar(block, src, s->dry_gain, FFALIGN(out->nb_samples, 4));
- emms_c();
+ 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));
- av_rdft_calc(seg->rdft[ch], block);
- block[2 * seg->part_size] = block[1];
- block[1] = 0;
+ memcpy(block, src, sizeof(*src) * seg->part_size);
- j = seg->part_index[ch];
+ av_rdft_calc(seg->rdft[ch], block);
+ block[2 * seg->part_size] = block[1];
+ block[1] = 0;
- for (i = 0; i < seg->nb_partitions; i++) {
- const int coffset = i * seg->coeff_size;
- const float *block = (const float *)seg->block->extended_data[ch] + j * seg->block_size;
- const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
+ j = seg->part_index[ch];
- s->fcmul_add(sum, block, (const float *)coeff, seg->part_size);
+ 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;
- if (j == 0)
- j = seg->nb_partitions;
- j--;
- }
+ s->fcmul_add(sum, block, (const float *)coeff, seg->part_size);
- sum[1] = sum[2 * seg->part_size];
- av_rdft_calc(seg->irdft[ch], sum);
+ if (j == 0)
+ j = seg->nb_partitions;
+ j--;
+ }
- dst = (float *)seg->buffer->extended_data[ch];
- for (n = 0; n < seg->part_size; n++) {
- dst[n] += sum[n];
- }
+ sum[1] = sum[2 * seg->part_size];
+ av_rdft_calc(seg->irdft[ch], sum);
- ptr = (float *)out->extended_data[ch];
- s->fdsp->vector_fmul_scalar(ptr, dst, s->wet_gain, FFALIGN(out->nb_samples, 4));
- emms_c();
+ buf = (float *)seg->buffer->extended_data[ch];
+ for (n = 0; n < seg->part_size; n++) {
+ buf[n] += sum[n];
+ }
- dst = (float *)seg->buffer->extended_data[ch];
- memcpy(dst, sum + seg->part_size, seg->part_size * sizeof(*dst));
+ for (n = 0; n < seg->part_size; n++) {
+ dst[n] += buf[n];
+ }
+
+ buf = (float *)seg->buffer->extended_data[ch];
+ memcpy(buf, sum + seg->part_size, seg->part_size * sizeof(*buf));
+
+ seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
+
+ memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
- seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
+ for (n = 0; n < out->nb_samples; n++) {
+ ptr[n] += dst[n];
+ }
+ }
+
+ s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(out->nb_samples, 4));
+ emms_c();
return 0;
}
@@ -275,19 +305,28 @@ end:
av_free(mag);
}
-static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int nb_partitions, int part_size)
+static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
+ int offset, int nb_partitions, int part_size)
{
+ 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)
return AVERROR(ENOMEM);
- seg->fft_length = part_size * 4 + 1;
- seg->part_size = part_size;
- seg->block_size = FFALIGN(seg->fft_length, 32);
- seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
+ seg->fft_length = part_size * 4 + 1;
+ seg->part_size = part_size;
+ seg->block_size = FFALIGN(seg->fft_length, 32);
+ seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
seg->nb_partitions = nb_partitions;
- seg->segment_size = part_size * nb_partitions;
+ seg->input_size = offset + s->min_part_size;
+ seg->input_offset = offset;
+
+ seg->part_index = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index));
+ seg->output_offset = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->output_offset));
+ if (!seg->part_index || !seg->output_offset)
+ return AVERROR(ENOMEM);
for (int ch = 0; ch < ctx->inputs[0]->channels; ch++) {
seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C);
@@ -296,15 +335,13 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int nb_parti
return AVERROR(ENOMEM);
}
- seg->part_index = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index));
- if (!seg->part_index)
- 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->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
seg->coeff = ff_get_audio_buffer(ctx->inputs[1], seg->nb_partitions * seg->coeff_size * 2);
- if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff)
+ 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)
return AVERROR(ENOMEM);
return 0;
@@ -313,20 +350,37 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int nb_parti
static int convert_coeffs(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
- int ret, i, ch, n, N;
+ int left, offset = 0, part_size, max_part_size;
+ int ret, i, ch, n;
float power = 0;
s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1]);
if (s->nb_taps <= 0)
return AVERROR(EINVAL);
- for (n = av_log2(s->minp); (1 << n) < s->nb_taps; n++);
- N = FFMIN(n, av_log2(s->maxp));
+ if (s->minp > s->maxp) {
+ s->maxp = s->minp;
+ }
- s->nb_segments = 1;
- ret = init_segment(ctx, &s->seg[0], (s->nb_taps + (1 << N) - 1) / (1 << N), 1 << N);
- if (ret < 0)
- return ret;
+ left = s->nb_taps;
+ part_size = 1 << av_log2(s->minp);
+ max_part_size = 1 << av_log2(s->maxp);
+
+ s->min_part_size = part_size;
+
+ for (i = 0; left > 0; i++) {
+ int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0);
+ int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
+
+ s->nb_segments = i + 1;
+ ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size);
+ if (ret < 0)
+ return ret;
+ offset += nb_partitions * part_size;
+ left -= nb_partitions * part_size;
+ part_size *= 2;
+ part_size = FFMIN(part_size, max_part_size);
+ }
ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]);
if (ret < 0)
@@ -426,7 +480,11 @@ static int convert_coeffs(AVFilterContext *ctx)
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, "block size: %d\n", seg->block_size);
av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
+ av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
+ av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
+ av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
}
}
@@ -488,7 +546,7 @@ static int activate(AVFilterContext *ctx)
return ret;
}
- ret = ff_inlink_consume_samples(ctx->inputs[0], s->seg[0].part_size, s->seg[0].part_size, &in);
+ ret = ff_inlink_consume_samples(ctx->inputs[0], s->min_part_size, s->min_part_size, &in);
if (ret > 0)
ret = fir_frame(s, in, outlink);
@@ -505,7 +563,7 @@ static int activate(AVFilterContext *ctx)
}
}
- if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->seg[0].part_size) {
+ if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
ff_filter_set_ready(ctx, 10);
return 0;
}
@@ -624,12 +682,16 @@ static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
}
av_freep(&seg->irdft);
+ av_freep(&seg->output_offset);
av_freep(&seg->part_index);
av_frame_free(&seg->block);
av_frame_free(&seg->sum);
av_frame_free(&seg->buffer);
av_frame_free(&seg->coeff);
+ av_frame_free(&seg->input);
+ av_frame_free(&seg->output);
+ seg->input_size = 0;
}
static av_cold void uninit(AVFilterContext *ctx)
@@ -720,11 +782,11 @@ static av_cold int init(AVFilterContext *ctx)
static const AVFilterPad afir_inputs[] = {
{
- .name = "main",
- .type = AVMEDIA_TYPE_AUDIO,
+ .name = "main",
+ .type = AVMEDIA_TYPE_AUDIO,
},{
- .name = "ir",
- .type = AVMEDIA_TYPE_AUDIO,
+ .name = "ir",
+ .type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
@@ -751,7 +813,7 @@ static const AVOption afir_options[] = {
{ "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
{ "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
{ "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
- { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=16}, 16, 32768, AF },
+ { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 16, 32768, AF },
{ "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 16, 32768, AF },
{ NULL }
};
diff --git a/libavfilter/af_afir.h b/libavfilter/af_afir.h
index f71ff34a0e..f9bec54b8c 100644
--- a/libavfilter/af_afir.h
+++ b/libavfilter/af_afir.h
@@ -37,14 +37,18 @@ typedef struct AudioFIRSegment {
int block_size;
int fft_length;
int coeff_size;
- int segment_size;
+ int input_size;
+ int input_offset;
+ int *output_offset;
int *part_index;
AVFrame *sum;
AVFrame *block;
AVFrame *buffer;
AVFrame *coeff;
+ AVFrame *input;
+ AVFrame *output;
RDFTContext **rdft, **irdft;
} AudioFIRSegment;
@@ -80,6 +84,7 @@ typedef struct AudioFIRContext {
AVFrame *in[2];
AVFrame *video;
+ int min_part_size;
int64_t pts;
AVFloatDSPContext *fdsp;
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