[FFmpeg-cvslog] avfilter/af_afir: add support for switching impulse response streams at runtime
Paul B Mahol
git at videolan.org
Fri Jan 10 15:05:48 EET 2020
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Thu Jan 9 20:06:57 2020 +0100| [52bf43eb4959918f75a7e3c9678812521ef2efff] | committer: Paul B Mahol
avfilter/af_afir: add support for switching impulse response streams at runtime
Currently, switching is not free of artifacts, to be resolved later.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=52bf43eb4959918f75a7e3c9678812521ef2efff
---
doc/filters.texi | 17 +++-
libavfilter/af_afir.c | 271 +++++++++++++++++++++++++++++++-------------------
libavfilter/af_afir.h | 4 +-
3 files changed, 182 insertions(+), 110 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index 9d18880913..9ff7bc2814 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1183,7 +1183,7 @@ afftfilt="real='hypot(re,im)*cos((random(0)*2-1)*2*3.14)':imag='hypot(re,im)*sin
@anchor{afir}
@section afir
-Apply an arbitrary Frequency Impulse Response filter.
+Apply an arbitrary Finite Impulse Response filter.
This filter is designed for applying long FIR filters,
up to 60 seconds long.
@@ -1192,10 +1192,10 @@ It can be used as component for digital crossover filters,
room equalization, cross talk cancellation, wavefield synthesis,
auralization, ambiophonics, ambisonics and spatialization.
-This filter uses the second stream as FIR coefficients.
-If the second stream holds a single channel, it will be used
+This filter uses the streams higher than first one as FIR coefficients.
+If the non-first stream holds a single channel, it will be used
for all input channels in the first stream, otherwise
-the number of channels in the second stream must be same as
+the number of channels in the non-first stream must be same as
the number of channels in the first stream.
It accepts the following parameters:
@@ -1264,6 +1264,15 @@ Lower values decreases latency at cost of higher CPU usage.
Set maximal partition size used for convolution. Default is @var{8192}.
Allowed range is from @var{8} to @var{32768}.
Lower values may increase CPU usage.
+
+ at item nbirs
+Set number of input impulse responses streams which will be switchable at runtime.
+Allowed range is from @var{1} to @var{32}. Default is @var{1}.
+
+ at item ir
+Set IR stream which will be used for convolution, starting from @var{0}, should always be
+lower than supplied value by @code{nbirs} option. Default is @var{0}.
+This option can be changed at runtime via @ref{commands}.
@end table
@subsection Examples
diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c
index 2545039a9e..077f9c7962 100644
--- a/libavfilter/af_afir.c
+++ b/libavfilter/af_afir.c
@@ -25,6 +25,7 @@
#include <float.h>
+#include "libavutil/avstring.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/intreadwrite.h"
@@ -298,9 +299,9 @@ static void draw_response(AVFilterContext *ctx, AVFrame *out)
if (!mag || !phase || !delay)
goto end;
- channel = av_clip(s->ir_channel, 0, s->ir[0]->channels - 1);
+ channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->channels - 1);
for (i = 0; i < s->w; i++) {
- const float *src = (const float *)s->ir[0]->extended_data[channel];
+ const float *src = (const float *)s->ir[s->selir]->extended_data[channel];
double w = i * M_PI / (s->w - 1);
double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
@@ -403,7 +404,7 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
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);
+ 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)
@@ -412,79 +413,116 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
return 0;
}
+static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
+{
+ AudioFIRContext *s = ctx->priv;
+
+ if (seg->rdft) {
+ for (int ch = 0; ch < s->nb_channels; ch++) {
+ av_rdft_end(seg->rdft[ch]);
+ }
+ }
+ av_freep(&seg->rdft);
+
+ if (seg->irdft) {
+ for (int ch = 0; ch < s->nb_channels; ch++) {
+ av_rdft_end(seg->irdft[ch]);
+ }
+ }
+ 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 int convert_coeffs(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
- int left, offset = 0, part_size, max_part_size;
- int ret, i, ch, n;
+ int ret, i, ch, n, cur_nb_taps;
float power = 0;
- s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1]);
- if (s->nb_taps <= 0)
- return AVERROR(EINVAL);
+ if (!s->nb_taps) {
+ int part_size, max_part_size;
+ int left, offset = 0;
- if (s->minp > s->maxp) {
- s->maxp = s->minp;
- }
+ s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1 + s->selir]);
+ if (s->nb_taps <= 0)
+ return AVERROR(EINVAL);
+
+ if (s->minp > s->maxp) {
+ s->maxp = s->minp;
+ }
+
+ left = s->nb_taps;
+ part_size = 1 << av_log2(s->minp);
+ max_part_size = 1 << av_log2(s->maxp);
- 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;
- 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);
- 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);
+ }
+ }
- s->nb_segments = i + 1;
- ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size);
+ if (!s->ir[s->selir]) {
+ ret = ff_inlink_consume_samples(ctx->inputs[1 + s->selir], s->nb_taps, s->nb_taps, &s->ir[s->selir]);
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);
+ if (ret == 0)
+ return AVERROR_BUG;
}
- ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->ir[0]);
- if (ret < 0)
- return ret;
- if (ret == 0)
- return AVERROR_BUG;
-
if (s->response)
draw_response(ctx, s->video);
s->gain = 1;
+ cur_nb_taps = s->ir[s->selir]->nb_samples;
switch (s->gtype) {
case -1:
/* nothing to do */
break;
case 0:
- for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
- float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
+ for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
+ float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
- for (i = 0; i < s->nb_taps; i++)
+ for (i = 0; i < cur_nb_taps; i++)
power += FFABS(time[i]);
}
- s->gain = ctx->inputs[1]->channels / power;
+ s->gain = ctx->inputs[1 + s->selir]->channels / power;
break;
case 1:
- for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
- float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
+ for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
+ float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
- for (i = 0; i < s->nb_taps; i++)
+ for (i = 0; i < cur_nb_taps; i++)
power += time[i];
}
- s->gain = ctx->inputs[1]->channels / power;
+ s->gain = ctx->inputs[1 + s->selir]->channels / power;
break;
case 2:
- for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
- float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
+ for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
+ float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
- for (i = 0; i < s->nb_taps; i++)
+ for (i = 0; i < cur_nb_taps; i++)
power += time[i] * time[i];
}
s->gain = sqrtf(ch / power);
@@ -495,17 +533,17 @@ static int convert_coeffs(AVFilterContext *ctx)
s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
- for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
- float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
+ for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
+ float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
- s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4));
+ s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 4));
}
- av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
+ av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", cur_nb_taps);
av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
- for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
- float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
+ for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
+ float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
int toffset = 0;
for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
@@ -561,7 +599,6 @@ static int convert_coeffs(AVFilterContext *ctx)
}
}
- av_frame_free(&s->ir[0]);
s->have_coeffs = 1;
return 0;
@@ -594,26 +631,26 @@ static int activate(AVFilterContext *ctx)
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
if (s->response)
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
- if (!s->eof_coeffs) {
+ if (!s->eof_coeffs[s->selir]) {
AVFrame *ir = NULL;
- ret = check_ir(ctx->inputs[1], ir);
+ ret = check_ir(ctx->inputs[1 + s->selir], ir);
if (ret < 0)
return ret;
- if (ff_outlink_get_status(ctx->inputs[1]) == AVERROR_EOF)
- s->eof_coeffs = 1;
+ if (ff_outlink_get_status(ctx->inputs[1 + s->selir]) == AVERROR_EOF)
+ s->eof_coeffs[s->selir] = 1;
- if (!s->eof_coeffs) {
+ if (!s->eof_coeffs[s->selir]) {
if (ff_outlink_frame_wanted(ctx->outputs[0]))
- ff_inlink_request_frame(ctx->inputs[1]);
+ ff_inlink_request_frame(ctx->inputs[1 + s->selir]);
else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
- ff_inlink_request_frame(ctx->inputs[1]);
+ ff_inlink_request_frame(ctx->inputs[1 + s->selir]);
return 0;
}
}
- if (!s->have_coeffs && s->eof_coeffs) {
+ if (!s->have_coeffs && s->eof_coeffs[s->selir]) {
ret = convert_coeffs(ctx);
if (ret < 0)
return ret;
@@ -709,8 +746,10 @@ static int query_formats(AVFilterContext *ctx)
return ret;
if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
return ret;
- if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[1]->out_channel_layouts)) < 0)
- return ret;
+ for (int i = 1; i < ctx->nb_inputs; i++) {
+ if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[i]->out_channel_layouts)) < 0)
+ return ret;
+ }
}
formats = ff_make_format_list(sample_fmts);
@@ -726,49 +765,19 @@ static int config_output(AVFilterLink *outlink)
AVFilterContext *ctx = outlink->src;
AudioFIRContext *s = ctx->priv;
- s->one2many = ctx->inputs[1]->channels == 1;
+ s->one2many = ctx->inputs[1 + s->selir]->channels == 1;
outlink->sample_rate = ctx->inputs[0]->sample_rate;
outlink->time_base = ctx->inputs[0]->time_base;
outlink->channel_layout = ctx->inputs[0]->channel_layout;
outlink->channels = ctx->inputs[0]->channels;
s->nb_channels = outlink->channels;
- s->nb_coef_channels = ctx->inputs[1]->channels;
+ s->nb_coef_channels = ctx->inputs[1 + s->selir]->channels;
s->pts = AV_NOPTS_VALUE;
return 0;
}
-static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
-{
- AudioFIRContext *s = ctx->priv;
-
- if (seg->rdft) {
- for (int ch = 0; ch < s->nb_channels; ch++) {
- av_rdft_end(seg->rdft[ch]);
- }
- }
- av_freep(&seg->rdft);
-
- if (seg->irdft) {
- for (int ch = 0; ch < s->nb_channels; ch++) {
- av_rdft_end(seg->irdft[ch]);
- }
- }
- 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)
{
AudioFIRContext *s = ctx->priv;
@@ -778,7 +787,13 @@ static av_cold void uninit(AVFilterContext *ctx)
}
av_freep(&s->fdsp);
- av_frame_free(&s->ir[0]);
+
+ for (int i = 0; i < s->nb_irs; i++) {
+ av_frame_free(&s->ir[i]);
+ }
+
+ for (int i = 0; i < ctx->nb_inputs; i++)
+ av_freep(&ctx->input_pads[i].name);
for (int i = 0; i < ctx->nb_outputs; i++)
av_freep(&ctx->output_pads[i].name);
@@ -818,7 +833,37 @@ static av_cold int init(AVFilterContext *ctx)
AVFilterPad pad, vpad;
int ret;
- pad = (AVFilterPad){
+ pad = (AVFilterPad) {
+ .name = av_strdup("main"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ };
+
+ if (!pad.name)
+ return AVERROR(ENOMEM);
+
+ ret = ff_insert_inpad(ctx, 0, &pad);
+ if (ret < 0) {
+ av_freep(&pad.name);
+ return ret;
+ }
+
+ for (int n = 0; n < s->nb_irs; n++) {
+ pad = (AVFilterPad) {
+ .name = av_asprintf("ir%d", n),
+ .type = AVMEDIA_TYPE_AUDIO,
+ };
+
+ if (!pad.name)
+ return AVERROR(ENOMEM);
+
+ ret = ff_insert_inpad(ctx, n + 1, &pad);
+ if (ret < 0) {
+ av_freep(&pad.name);
+ return ret;
+ }
+ }
+
+ pad = (AVFilterPad) {
.name = av_strdup("default"),
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
@@ -860,18 +905,31 @@ static av_cold int init(AVFilterContext *ctx)
return 0;
}
-static const AVFilterPad afir_inputs[] = {
- {
- .name = "main",
- .type = AVMEDIA_TYPE_AUDIO,
- },{
- .name = "ir",
- .type = AVMEDIA_TYPE_AUDIO,
- },
- { NULL }
-};
+static int process_command(AVFilterContext *ctx,
+ const char *cmd,
+ const char *arg,
+ char *res,
+ int res_len,
+ int flags)
+{
+ AudioFIRContext *s = ctx->priv;
+ int prev_ir = s->selir;
+ int ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
+
+ if (ret < 0)
+ return ret;
+
+ s->selir = FFMIN(s->nb_irs - 1, s->selir);
+
+ if (prev_ir != s->selir) {
+ s->have_coeffs = 0;
+ }
+
+ return 0;
+}
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
#define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define OFFSET(x) offsetof(AudioFIRContext, x)
@@ -895,6 +953,8 @@ static const AVOption afir_options[] = {
{ "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=8192}, 1, 32768, AF },
{ "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 32768, AF },
+ { "nbirs", "set number of input IRs",OFFSET(nb_irs),AV_OPT_TYPE_INT, {.i64=1}, 1, 32, AF },
+ { "ir", "select IR", OFFSET(selir), AV_OPT_TYPE_INT, {.i64=0}, 0, 31, AFR },
{ NULL }
};
@@ -902,14 +962,15 @@ AVFILTER_DEFINE_CLASS(afir);
AVFilter ff_af_afir = {
.name = "afir",
- .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
+ .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in additional stream(s)."),
.priv_size = sizeof(AudioFIRContext),
.priv_class = &afir_class,
.query_formats = query_formats,
.init = init,
.activate = activate,
.uninit = uninit,
- .inputs = afir_inputs,
- .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
+ .process_command = process_command,
+ .flags = AVFILTER_FLAG_DYNAMIC_INPUTS |
+ AVFILTER_FLAG_DYNAMIC_OUTPUTS |
AVFILTER_FLAG_SLICE_THREADS,
};
diff --git a/libavfilter/af_afir.h b/libavfilter/af_afir.h
index 1b59d85175..4f44675848 100644
--- a/libavfilter/af_afir.h
+++ b/libavfilter/af_afir.h
@@ -74,10 +74,12 @@ typedef struct AudioFIRContext {
int ir_channel;
int minp;
int maxp;
+ int nb_irs;
+ int selir;
float gain;
- int eof_coeffs;
+ int eof_coeffs[32];
int have_coeffs;
int nb_taps;
int nb_channels;
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