[FFmpeg-cvslog] avfilter/af_aemphasis: add timeline/slice and commands support
Paul B Mahol
git at videolan.org
Fri Nov 27 23:41:20 EET 2020
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Fri Nov 27 22:32:00 2020 +0100| [7a95cf86ff6b681b2ef43660c769a9b3cfd82e6a] | committer: Paul B Mahol
avfilter/af_aemphasis: add timeline/slice and commands support
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7a95cf86ff6b681b2ef43660c769a9b3cfd82e6a
---
doc/filters.texi | 4 ++
libavfilter/af_aemphasis.c | 127 ++++++++++++++++++++++++++++-----------------
2 files changed, 83 insertions(+), 48 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index a3d344d558..8aefa560f2 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -911,6 +911,10 @@ select 75µs (FM-KF).
@end table
@end table
+ at subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section aeval
Modify an audio signal according to the specified expressions.
diff --git a/libavfilter/af_aemphasis.c b/libavfilter/af_aemphasis.c
index 29d6e46965..50999572c9 100644
--- a/libavfilter/af_aemphasis.c
+++ b/libavfilter/af_aemphasis.c
@@ -27,13 +27,9 @@ typedef struct BiquadCoeffs {
double a0, a1, a2, b1, b2;
} BiquadCoeffs;
-typedef struct BiquadD2 {
- double a0, a1, a2, b1, b2, w1, w2;
-} BiquadD2;
-
typedef struct RIAACurve {
- BiquadD2 r1;
- BiquadD2 brickw;
+ BiquadCoeffs r1;
+ BiquadCoeffs brickw;
int use_brickw;
} RIAACurve;
@@ -42,11 +38,13 @@ typedef struct AudioEmphasisContext {
int mode, type;
double level_in, level_out;
- RIAACurve *rc;
+ RIAACurve rc;
+
+ AVFrame *w;
} AudioEmphasisContext;
#define OFFSET(x) offsetof(AudioEmphasisContext, x)
-#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption aemphasis_options[] = {
{ "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
@@ -69,16 +67,16 @@ static const AVOption aemphasis_options[] = {
AVFILTER_DEFINE_CLASS(aemphasis);
-static inline void biquad_process(BiquadD2 *bq, double *dst, const double *src, int nb_samples,
- double level_in, double level_out)
+static inline void biquad_process(BiquadCoeffs *bq, double *dst, const double *src, int nb_samples,
+ double *w, double level_in, double level_out)
{
const double a0 = bq->a0;
const double a1 = bq->a1;
const double a2 = bq->a2;
const double b1 = bq->b1;
const double b2 = bq->b2;
- double w1 = bq->w1;
- double w2 = bq->w2;
+ double w1 = w[0];
+ double w2 = w[1];
for (int i = 0; i < nb_samples; i++) {
double n = src[i] * level_in;
@@ -91,17 +89,46 @@ static inline void biquad_process(BiquadD2 *bq, double *dst, const double *src,
dst[i] = out * level_out;
}
- bq->w1 = w1;
- bq->w2 = w2;
+ w[0] = w1;
+ w[1] = w2;
}
-static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+typedef struct ThreadData {
+ AVFrame *in, *out;
+} ThreadData;
+
+static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
- AVFilterContext *ctx = inlink->dst;
- AVFilterLink *outlink = ctx->outputs[0];
AudioEmphasisContext *s = ctx->priv;
const double level_out = s->level_out;
const double level_in = s->level_in;
+ ThreadData *td = arg;
+ AVFrame *out = td->out;
+ AVFrame *in = td->in;
+ const int start = (in->channels * jobnr) / nb_jobs;
+ const int end = (in->channels * (jobnr+1)) / nb_jobs;
+
+ for (int ch = start; ch < end; ch++) {
+ const double *src = (const double *)in->extended_data[ch];
+ double *w = (double *)s->w->extended_data[ch];
+ double *dst = (double *)out->extended_data[ch];
+
+ if (s->rc.use_brickw) {
+ biquad_process(&s->rc.brickw, dst, src, in->nb_samples, w + 2, level_in, 1.);
+ biquad_process(&s->rc.r1, dst, dst, in->nb_samples, w, 1., level_out);
+ } else {
+ biquad_process(&s->rc.r1, dst, src, in->nb_samples, w, level_in, level_out);
+ }
+ }
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ ThreadData td;
AVFrame *out;
if (av_frame_is_writable(in)) {
@@ -115,17 +142,9 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
av_frame_copy_props(out, in);
}
- for (int ch = 0; ch < inlink->channels; ch++) {
- const double *src = (const double *)in->extended_data[ch];
- double *dst = (double *)out->extended_data[ch];
-
- if (s->rc[ch].use_brickw) {
- biquad_process(&s->rc[ch].brickw, dst, src, in->nb_samples, level_in, 1.);
- biquad_process(&s->rc[ch].r1, dst, dst, in->nb_samples, 1., level_out);
- } else {
- biquad_process(&s->rc[ch].r1, dst, src, in->nb_samples, level_in, level_out);
- }
- }
+ td.in = in; td.out = out;
+ ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
+ ff_filter_get_nb_threads(ctx)));
if (in != out)
av_frame_free(&in);
@@ -162,7 +181,7 @@ static int query_formats(AVFilterContext *ctx)
return ff_set_common_samplerates(ctx, formats);
}
-static inline void set_highshelf_rbj(BiquadD2 *bq, double freq, double q, double peak, double sr)
+static inline void set_highshelf_rbj(BiquadCoeffs *bq, double freq, double q, double peak, double sr)
{
double A = sqrt(peak);
double w0 = freq * 2 * M_PI / sr;
@@ -186,7 +205,7 @@ static inline void set_highshelf_rbj(BiquadD2 *bq, double freq, double q, double
bq->a2 *= ib0;
}
-static inline void set_lp_rbj(BiquadD2 *bq, double fc, double q, double sr, double gain)
+static inline void set_lp_rbj(BiquadCoeffs *bq, double fc, double q, double sr, double gain)
{
double omega = 2.0 * M_PI * fc / sr;
double sn = sin(omega);
@@ -220,10 +239,10 @@ static int config_input(AVFilterLink *inlink)
AVFilterContext *ctx = inlink->dst;
AudioEmphasisContext *s = ctx->priv;
BiquadCoeffs coeffs;
- int ch;
- s->rc = av_calloc(inlink->channels, sizeof(*s->rc));
- if (!s->rc)
+ if (!s->w)
+ s->w = ff_get_audio_buffer(inlink, 4);
+ if (!s->w)
return AVERROR(ENOMEM);
switch (s->type) {
@@ -297,12 +316,12 @@ static int config_input(AVFilterLink *inlink)
if (s->type == 7)
q = pow((sr / 4750.0) + 19.5, -0.25);
if (s->mode == 0)
- set_highshelf_rbj(&s->rc[0].r1, cfreq, q, 1. / gain, sr);
+ set_highshelf_rbj(&s->rc.r1, cfreq, q, 1. / gain, sr);
else
- set_highshelf_rbj(&s->rc[0].r1, cfreq, q, gain, sr);
- s->rc[0].use_brickw = 0;
+ set_highshelf_rbj(&s->rc.r1, cfreq, q, gain, sr);
+ s->rc.use_brickw = 0;
} else {
- s->rc[0].use_brickw = 1;
+ s->rc.use_brickw = 1;
if (s->mode == 0) { // Reproduction
g = 1. / (4.+2.*i*t+2.*k*t+i*k*t*t);
a0 = (2.*t+j*t*t)*g;
@@ -331,27 +350,36 @@ static int config_input(AVFilterLink *inlink)
gain1kHz = freq_gain(&coeffs, 1000.0, sr);
// divide one filter's x[n-m] coefficients by that value
gc = 1.0 / gain1kHz;
- s->rc[0].r1.a0 = coeffs.a0 * gc;
- s->rc[0].r1.a1 = coeffs.a1 * gc;
- s->rc[0].r1.a2 = coeffs.a2 * gc;
- s->rc[0].r1.b1 = coeffs.b1;
- s->rc[0].r1.b2 = coeffs.b2;
+ s->rc.r1.a0 = coeffs.a0 * gc;
+ s->rc.r1.a1 = coeffs.a1 * gc;
+ s->rc.r1.a2 = coeffs.a2 * gc;
+ s->rc.r1.b1 = coeffs.b1;
+ s->rc.r1.b2 = coeffs.b2;
}
cutfreq = FFMIN(0.45 * sr, 21000.);
- set_lp_rbj(&s->rc[0].brickw, cutfreq, 0.707, sr, 1.);
-
- for (ch = 1; ch < inlink->channels; ch++) {
- memcpy(&s->rc[ch], &s->rc[0], sizeof(RIAACurve));
- }
+ set_lp_rbj(&s->rc.brickw, cutfreq, 0.707, sr, 1.);
return 0;
}
+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;
+
+ return config_input(ctx->inputs[0]);
+}
+
static av_cold void uninit(AVFilterContext *ctx)
{
AudioEmphasisContext *s = ctx->priv;
- av_freep(&s->rc);
+
+ av_frame_free(&s->w);
}
static const AVFilterPad avfilter_af_aemphasis_inputs[] = {
@@ -381,4 +409,7 @@ AVFilter ff_af_aemphasis = {
.query_formats = query_formats,
.inputs = avfilter_af_aemphasis_inputs,
.outputs = avfilter_af_aemphasis_outputs,
+ .process_command = process_command,
+ .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
+ AVFILTER_FLAG_SLICE_THREADS,
};
More information about the ffmpeg-cvslog
mailing list