[FFmpeg-devel] [PATCH] lavfi: add aeval filter
Stefano Sabatini
stefasab at gmail.com
Mon Dec 2 17:59:05 CET 2013
---
doc/filters.texi | 60 ++++++++++++++++
libavfilter/Makefile | 1 +
libavfilter/allfilters.c | 1 +
libavfilter/asrc_aevalsrc.c | 162 ++++++++++++++++++++++++++++++++++++++++++++
4 files changed, 224 insertions(+)
diff --git a/doc/filters.texi b/doc/filters.texi
index 0804bb4..40dec9c 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -435,6 +435,66 @@ aecho=0.8:0.9:1000|1800:0.3|0.25
@end example
@end itemize
+ at section aeval
+
+Modify an audio signal according to the specified expression.
+
+This source accepts in input one or more expressions (one for each
+channel), which are evaluated and used to modify a corresponding audio
+signal.
+
+This filter accepts the following options:
+
+ at table @option
+ at item exprs
+Set the '|'-separated expressions list for each separate channel. If
+the number of input channels is greater than the number of
+expressions, the last specified expression is repeated for the other
+channels as well.
+
+Default value is "val".
+ at end table
+
+Each expression in @var{exprs} can contain the following constants:
+
+ at table @option
+ at item pts
+presentation timestamp of the evaluated sample expressed in seconds, starting from 0.
+This corresponds to the timestamp of the first frame sample.
+
+ at item n
+number of the evaluated sample, starting from 0
+
+ at item s
+sample rate
+
+ at item t
+time of the evaluated sample expressed in seconds, starting from 0.
+This corresponds to the time of the first frame sample.
+
+ at item val
+the value of the currently processed frame
+ at end table
+
+Note: this filter is slow. For faster processing you should use a
+dedicated filter.
+
+ at subsection Examples
+
+ at itemize
+ at item
+Half volume:
+ at example
+aeval=val/2
+ at end example
+
+ at item
+Invert phase of the second channel:
+ at example
+eval=val|-val
+ at end example
+ at end itemize
+
@section afade
Apply fade-in/out effect to input audio.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 3d8a763..a419a7b 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -53,6 +53,7 @@ OBJS-$(CONFIG_AVCODEC) += avcodec.o
OBJS-$(CONFIG_ACONVERT_FILTER) += af_aconvert.o
OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
+OBJS-$(CONFIG_AEVAL_FILTER) += asrc_aevalsrc.o
OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 04b4973..bd23b0a 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -50,6 +50,7 @@ void avfilter_register_all(void)
#endif
REGISTER_FILTER(ADELAY, adelay, af);
REGISTER_FILTER(AECHO, aecho, af);
+ REGISTER_FILTER(AEVAL, aeval, af);
REGISTER_FILTER(AFADE, afade, af);
REGISTER_FILTER(AFORMAT, aformat, af);
REGISTER_FILTER(AINTERLEAVE, ainterleave, af);
diff --git a/libavfilter/asrc_aevalsrc.c b/libavfilter/asrc_aevalsrc.c
index 41ad632..674176e 100644
--- a/libavfilter/asrc_aevalsrc.c
+++ b/libavfilter/asrc_aevalsrc.c
@@ -34,16 +34,20 @@
#include "internal.h"
static const char * const var_names[] = {
+ "pts", ///< presentation timestamp
"n", ///< number of frame
"t", ///< timestamp expressed in seconds
"s", ///< sample rate
+ "val", ///< currently processed value
NULL
};
enum var_name {
+ VAR_PTS,
VAR_N,
VAR_T,
VAR_S,
+ VAR_VAL,
VAR_VARS_NB
};
@@ -220,6 +224,7 @@ static int request_frame(AVFilterLink *outlink)
return ff_filter_frame(outlink, samplesref);
}
+#if CONFIG_AEVALSRC_FILTER
static const AVFilterPad aevalsrc_outputs[] = {
{
.name = "default",
@@ -241,3 +246,160 @@ AVFilter ff_asrc_aevalsrc = {
.outputs = aevalsrc_outputs,
.priv_class = &aevalsrc_class,
};
+
+#endif /* CONFIG_AEVALSRC_FILTER */
+
+#define OFFSET(x) offsetof(EvalContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption aeval_options[]= {
+ { "exprs", "set the '|'-separated list of channels expressions", OFFSET(exprs), AV_OPT_TYPE_STRING, {.str = "val"}, .flags = FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(aeval);
+
+static int aeval_query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layouts;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE
+ };
+
+ layouts = ff_all_channel_layouts();
+ if (!layouts)
+ return AVERROR(ENOMEM);
+ ff_set_common_channel_layouts(ctx, layouts);
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ff_set_common_formats(ctx, formats);
+
+ formats = ff_all_samplerates();
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ff_set_common_samplerates(ctx, formats);
+
+ return 0;
+}
+
+static int aeval_config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ EvalContext *eval = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
+ char *args1 = av_strdup(eval->exprs);
+ char *expr, *buf;
+ int ret, i, nb_channels = 0;
+
+ /* parse expressions */
+ buf = args1;
+ while (expr = av_strtok(buf, "|", &buf)) {
+ if (!av_dynarray2_add((void **)&eval->expr, &nb_channels, sizeof(*eval->expr), NULL)) {
+ ret = AVERROR(ENOMEM);
+ goto end;
+ }
+ eval->expr[nb_channels-1] = NULL;
+ ret = av_expr_parse(&eval->expr[nb_channels - 1], expr, var_names,
+ NULL, NULL, NULL, NULL, 0, ctx);
+ if (ret < 0)
+ goto end;
+ }
+
+ if (nb_channels != inlink->channels)
+ av_log(ctx, AV_LOG_WARNING, "Mismatch between number of processed channels (%d), "
+ "and number of parsed expressions (%d)\n", nb_channels, inlink->channels);
+
+ if (nb_channels < inlink->channels) {
+ for (i = nb_channels; i < inlink->channels; i++) {
+ ret = av_expr_parse(&eval->expr[i], expr, var_names,
+ NULL, NULL, NULL, NULL, 0, ctx);
+ if (ret < 0)
+ goto end;
+ }
+ }
+
+ eval->n = 0;
+ eval->var_values[VAR_S] = inlink->sample_rate;
+ eval->var_values[VAR_PTS] = NAN;
+ eval->var_values[VAR_T] = NAN;
+end:
+ av_free(args1);
+ return ret;
+}
+
+#define TS2D(ts) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts))
+#define TS2T(ts, tb) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts)*av_q2d(tb))
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ EvalContext *eval = inlink->dst->priv;
+ AVFilterLink *outlink = inlink->dst->outputs[0];
+ int nb_samples = in->nb_samples;
+ AVFrame *out;
+ int i, j;
+
+ /* do volume scaling in-place if input buffer is writable */
+ if (av_frame_is_writable(in)) {
+ out = in;
+ } else {
+ out = ff_get_audio_buffer(inlink, nb_samples);
+ if (!out)
+ return AVERROR(ENOMEM);
+ av_frame_copy_props(out, in);
+ }
+
+ eval->var_values[VAR_PTS] = TS2D(in->pts);
+ eval->var_values[VAR_T] = TS2T(in->pts, inlink->time_base);
+
+ /* evaluate expression for each single sample and for each channel */
+ for (i = 0; i < nb_samples; i++, eval->n++) {
+ eval->var_values[VAR_N] = eval->n;
+ for (j = 0; j < inlink->channels; j++) {
+ eval->var_values[VAR_VAL] = *((double *) in->extended_data[j] + i);
+ *((double *) out->extended_data[j] + i) =
+ av_expr_eval(eval->expr[j], eval->var_values, NULL);
+ }
+ }
+
+ if (in != out)
+ av_frame_free(&in);
+
+ return ff_filter_frame(outlink, out);
+}
+
+#if CONFIG_AEVAL_FILTER
+
+static const AVFilterPad aeval_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad aeval_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = aeval_config_output,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_aeval = {
+ .name = "aeval",
+ .description = NULL_IF_CONFIG_SMALL("Filter audio signal according to a specified expression."),
+ .query_formats = aeval_query_formats,
+ .uninit = uninit,
+ .priv_size = sizeof(EvalContext),
+ .inputs = aeval_inputs,
+ .outputs = aeval_outputs,
+ .priv_class = &aeval_class,
+ .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
+};
+
+#endif /* CONFIG_AEVAL_FILTER */
--
1.8.1.2
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