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avfilter/af_anlmdn: support all options as commands

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This commit is contained in:
Paul B Mahol 2020-11-17 13:43:55 +01:00
parent 96f1b45b8c
commit bb7bb440c2
2 changed files with 77 additions and 37 deletions
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11
doc/filters.texi
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@ -1935,16 +1935,7 @@ Set smooth factor. Default value is @var{11}. Allowed range is from @var{1} to @
@subsection Commands
This filter supports the following commands:
@table @option
@item s
Change denoise strength. Argument is single float number.
Syntax for the command is : "@var{s}"
@item o
Change output mode.
Syntax for the command is : "i", "o" or "n" string.
@end table
This filter supports the all above options as @ref{commands}.
@section anlms
Apply Normalized Least-Mean-Squares algorithm to the first audio stream using the second audio stream.

103
libavfilter/af_anlmdn.c
View File

@ -72,13 +72,12 @@ enum OutModes {
};
#define OFFSET(x) offsetof(AudioNLMeansContext, x)
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define AFT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption anlmdn_options[] = {
{ "s", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=0.00001},0.00001, 10, AFT },
{ "p", "set patch duration", OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AF },
{ "r", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AF },
{ "p", "set patch duration", OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AFT },
{ "r", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AFT },
{ "o", "set output mode", OFFSET(om), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_MODES-1, AFT, "mode" },
{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AFT, "mode" },
{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AFT, "mode" },
@ -147,31 +146,72 @@ void ff_anlmdn_init(AudioNLMDNDSPContext *dsp)
ff_anlmdn_init_x86(dsp);
}
static int config_filter(AVFilterContext *ctx)
{
AudioNLMeansContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int newK, newS, newH, newN;
AVFrame *new_in, *new_cache;
newK = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
newS = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);
newH = newK * 2 + 1;
newN = newH + (newK + newS) * 2;
av_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", newK, newS, newH, newN);
if (!s->cache || s->cache->nb_samples < newS * 2) {
new_cache = ff_get_audio_buffer(outlink, newS * 2);
if (new_cache) {
av_frame_free(&s->cache);
s->cache = new_cache;
} else {
return AVERROR(ENOMEM);
}
}
if (!s->cache)
return AVERROR(ENOMEM);
s->pdiff_lut_scale = 1.f / s->m * WEIGHT_LUT_SIZE;
for (int i = 0; i < WEIGHT_LUT_SIZE; i++) {
float w = -i / s->pdiff_lut_scale;
s->weight_lut[i] = expf(w);
}
if (!s->in || s->in->nb_samples < newN) {
new_in = ff_get_audio_buffer(outlink, newN);
if (new_in) {
av_frame_free(&s->in);
s->in = new_in;
} else {
return AVERROR(ENOMEM);
}
}
if (!s->in)
return AVERROR(ENOMEM);
s->K = newK;
s->S = newS;
s->H = newH;
s->N = newN;
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioNLMeansContext *s = ctx->priv;
int ret;
s->K = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
s->S = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);
s->eof_left = -1;
s->pts = AV_NOPTS_VALUE;
s->H = s->K * 2 + 1;
s->N = s->H + (s->K + s->S) * 2;
av_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", s->K, s->S, s->H, s->N);
av_frame_free(&s->in);
av_frame_free(&s->cache);
s->in = ff_get_audio_buffer(outlink, s->N);
if (!s->in)
return AVERROR(ENOMEM);
s->cache = ff_get_audio_buffer(outlink, s->S * 2);
if (!s->cache)
return AVERROR(ENOMEM);
ret = config_filter(ctx);
if (ret < 0)
return ret;
s->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, s->N);
if (!s->fifo)
@ -181,13 +221,6 @@ static int config_output(AVFilterLink *outlink)
if (ret < 0)
return ret;
s->pdiff_lut_scale = 1.f / s->m * WEIGHT_LUT_SIZE;
for (int i = 0; i < WEIGHT_LUT_SIZE; i++) {
float w = -i / s->pdiff_lut_scale;
s->weight_lut[i] = expf(w);
}
ff_anlmdn_init(&s->dsp);
return 0;
@ -331,6 +364,22 @@ static int request_frame(AVFilterLink *outlink)
return ret;
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
int ret;
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
if (ret < 0)
return ret;
ret = config_filter(ctx);
if (ret < 0)
return ret;
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioNLMeansContext *s = ctx->priv;
@ -368,7 +417,7 @@ AVFilter ff_af_anlmdn = {
.uninit = uninit,
.inputs = inputs,
.outputs = outputs,
.process_command = ff_filter_process_command,
.process_command = process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
};