[FFmpeg-cvslog] avfilter/af_anlmdn: support all options as commands
Paul B Mahol
git at videolan.org
Tue Nov 17 14:50:16 EET 2020
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Tue Nov 17 13:43:55 2020 +0100| [bb7bb440c2010d707edf1e8f0b85d031ae299cbd] | committer: Paul B Mahol
avfilter/af_anlmdn: support all options as commands
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=bb7bb440c2010d707edf1e8f0b85d031ae299cbd
---
doc/filters.texi | 11 +-----
libavfilter/af_anlmdn.c | 103 +++++++++++++++++++++++++++++++++++-------------
2 files changed, 77 insertions(+), 37 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index 8c9eccb00e..15e4873dcf 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -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:
- at table @option
- at item s
-Change denoise strength. Argument is single float number.
-Syntax for the command is : "@var{s}"
-
- at item o
-Change output mode.
-Syntax for the command is : "i", "o" or "n" string.
- at 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.
diff --git a/libavfilter/af_anlmdn.c b/libavfilter/af_anlmdn.c
index ea473bdab8..e2661e102f 100644
--- a/libavfilter/af_anlmdn.c
+++ b/libavfilter/af_anlmdn.c
@@ -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" },
@@ -147,32 +146,73 @@ void ff_anlmdn_init(AudioNLMDNDSPContext *dsp)
ff_anlmdn_init_x86(dsp);
}
-static int config_output(AVFilterLink *outlink)
+static int config_filter(AVFilterContext *ctx)
{
- AVFilterContext *ctx = outlink->src;
AudioNLMeansContext *s = ctx->priv;
- int ret;
+ AVFilterLink *outlink = ctx->outputs[0];
+ int newK, newS, newH, newN;
+ AVFrame *new_in, *new_cache;
- s->K = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
- s->S = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);
+ newK = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
+ newS = 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;
+ newH = newK * 2 + 1;
+ newN = newH + (newK + newS) * 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_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", newK, newS, newH, newN);
- av_frame_free(&s->in);
- av_frame_free(&s->cache);
- s->in = ff_get_audio_buffer(outlink, s->N);
- if (!s->in)
+ 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->cache = ff_get_audio_buffer(outlink, s->S * 2);
- if (!s->cache)
+ 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->eof_left = -1;
+ s->pts = AV_NOPTS_VALUE;
+
+ ret = config_filter(ctx);
+ if (ret < 0)
+ return ret;
+
s->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, s->N);
if (!s->fifo)
return AVERROR(ENOMEM);
@@ -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,
};
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