ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Wed Feb 16 19:31:31 2022 +0100| [698de27f25bea278ce7155fc387b3d59b478d46b] | committer: Paul B Mahol
avfilter/af_speechnorm: speed up filtering code
Reduce some asserts by default.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=698de27f25bea278ce7155fc387b3d59b478d46b
---
libavfilter/af_speechnorm.c | 133 ++++++++++++++++++++++++--------------------
1 file changed, 73 insertions(+), 60 deletions(-)
diff --git a/libavfilter/af_speechnorm.c b/libavfilter/af_speechnorm.c
index 212a926f36..7fa87f814a 100644
--- a/libavfilter/af_speechnorm.c
+++ b/libavfilter/af_speechnorm.c
@@ -128,7 +128,7 @@ static int get_pi_samples(PeriodItem *pi, int start, int end, int remain)
start = 0;
if (pi[start].type == 0)
break;
- av_assert0(pi[start].size > 0);
+ av_assert1(pi[start].size > 0);
sum += pi[start].size;
}
@@ -156,7 +156,7 @@ static void consume_pi(ChannelContext *cc, int nb_samples)
if (cc->pi_size >= nb_samples) {
cc->pi_size -= nb_samples;
} else {
- av_assert0(0);
+ av_assert1(0);
}
}
@@ -178,16 +178,16 @@ static double next_gain(AVFilterContext *ctx, double pi_max_peak, int bypass, do
static void next_pi(AVFilterContext *ctx, ChannelContext *cc, int bypass)
{
- av_assert0(cc->pi_size >= 0);
+ av_assert1(cc->pi_size >= 0);
if (cc->pi_size == 0) {
SpeechNormalizerContext *s = ctx->priv;
int start = cc->pi_start;
- av_assert0(cc->pi[start].size > 0);
+ av_assert1(cc->pi[start].size > 0);
av_assert0(cc->pi[start].type > 0 || s->eof);
cc->pi_size = cc->pi[start].size;
cc->pi_max_peak = cc->pi[start].max_peak;
- av_assert0(cc->pi_start != cc->pi_end || s->eof);
+ av_assert1(cc->pi_start != cc->pi_end || s->eof);
start++;
if (start >= MAX_ITEMS)
start = 0;
@@ -219,58 +219,71 @@ static double min_gain(AVFilterContext *ctx, ChannelContext *cc, int max_size)
return min_gain;
}
-#define ANALYZE_CHANNEL(name, ptype, zero, min_peak) \
-static void analyze_channel_## name (AVFilterContext *ctx, ChannelContext *cc, \
- const uint8_t *srcp, int nb_samples) \
-{ \
- SpeechNormalizerContext *s = ctx->priv; \
- const ptype *src = (const ptype *)srcp; \
- int n = 0; \
- \
- if (cc->state < 0) \
- cc->state = src[0] >= zero; \
- \
- while (n < nb_samples) { \
- if ((cc->state != (src[n] >= zero)) || \
- (cc->pi[cc->pi_end].size > s->max_period)) { \
- ptype max_peak = cc->pi[cc->pi_end].max_peak; \
- int state = cc->state; \
- cc->state = src[n] >= zero; \
- av_assert0(cc->pi[cc->pi_end].size > 0); \
- if (max_peak >= min_peak || \
- cc->pi[cc->pi_end].size > s->max_period) { \
- cc->pi[cc->pi_end].type = 1; \
- cc->pi_end++; \
- if (cc->pi_end >= MAX_ITEMS) \
- cc->pi_end = 0; \
- if (cc->state != state) \
- cc->pi[cc->pi_end].max_peak = DBL_MIN; \
- else \
- cc->pi[cc->pi_end].max_peak = max_peak; \
- cc->pi[cc->pi_end].type = 0; \
- cc->pi[cc->pi_end].size = 0; \
- av_assert0(cc->pi_end != cc->pi_start); \
- } \
- } \
- \
- if (cc->state) { \
- while (src[n] >= zero) { \
- cc->pi[cc->pi_end].max_peak = FFMAX(cc->pi[cc->pi_end].max_peak, src[n]); \
- cc->pi[cc->pi_end].size++; \
- n++; \
- if (n >= nb_samples) \
- break; \
- } \
- } else { \
- while (src[n] < zero) { \
- cc->pi[cc->pi_end].max_peak = FFMAX(cc->pi[cc->pi_end].max_peak, -src[n]); \
- cc->pi[cc->pi_end].size++; \
- n++; \
- if (n >= nb_samples) \
- break; \
- } \
- } \
- } \
+#define ANALYZE_CHANNEL(name, ptype, zero, min_peak) \
+static void analyze_channel_## name (AVFilterContext *ctx, ChannelContext *cc, \
+ const uint8_t *srcp, int nb_samples) \
+{ \
+ SpeechNormalizerContext *s = ctx->priv; \
+ const ptype *src = (const ptype *)srcp; \
+ const int max_period = s->max_period; \
+ PeriodItem *pi = (PeriodItem *)&cc->pi; \
+ int pi_end = cc->pi_end; \
+ int n = 0; \
+ \
+ if (cc->state < 0) \
+ cc->state = src[0] >= zero; \
+ \
+ while (n < nb_samples) { \
+ ptype new_max_peak; \
+ int new_size; \
+ \
+ if ((cc->state != (src[n] >= zero)) || \
+ (pi[pi_end].size > max_period)) { \
+ ptype max_peak = pi[pi_end].max_peak; \
+ int state = cc->state; \
+ \
+ cc->state = src[n] >= zero; \
+ av_assert1(pi[pi_end].size > 0); \
+ if (max_peak >= min_peak || \
+ pi[pi_end].size > max_period) { \
+ pi[pi_end].type = 1; \
+ pi_end++; \
+ if (pi_end >= MAX_ITEMS) \
+ pi_end = 0; \
+ if (cc->state != state) \
+ pi[pi_end].max_peak = DBL_MIN; \
+ else \
+ pi[pi_end].max_peak = max_peak; \
+ pi[pi_end].type = 0; \
+ pi[pi_end].size = 0; \
+ av_assert1(pi_end != cc->pi_start); \
+ } \
+ } \
+ \
+ new_max_peak = pi[pi_end].max_peak; \
+ new_size = pi[pi_end].size; \
+ if (cc->state) { \
+ while (src[n] >= zero) { \
+ new_max_peak = FFMAX(new_max_peak, src[n]); \
+ new_size++; \
+ n++; \
+ if (n >= nb_samples) \
+ break; \
+ } \
+ } else { \
+ while (src[n] < zero) { \
+ new_max_peak = FFMAX(new_max_peak, -src[n]); \
+ new_size++; \
+ n++; \
+ if (n >= nb_samples) \
+ break; \
+ } \
+ } \
+ \
+ pi[pi_end].max_peak = new_max_peak; \
+ pi[pi_end].size = new_size; \
+ } \
+ cc->pi_end = pi_end; \
}
ANALYZE_CHANNEL(dbl, double, 0.0, MIN_PEAK)
@@ -296,7 +309,7 @@ static void filter_channels_## name (AVFilterContext *ctx,
\
next_pi(ctx, cc, bypass); \
size = FFMIN(nb_samples - n, cc->pi_size); \
- av_assert0(size > 0); \
+ av_assert1(size > 0); \
gain = cc->gain_state; \
consume_pi(cc, size); \
for (int i = n; !ctx->is_disabled && i < n + size; i++) \
@@ -343,7 +356,7 @@ static void filter_link_channels_## name (AVFilterContext *ctx,
max_size = FFMAX(max_size, cc->pi_size); \
} \
\
- av_assert0(min_size > 0); \
+ av_assert1(min_size > 0); \
for (int ch = 0; ch < inlink->channels; ch++) { \
ChannelContext *cc = &s->cc[ch]; \
\
@@ -509,7 +522,7 @@ static int config_input(AVFilterLink *inlink)
s->filter_channels[1] = filter_link_channels_dbl;
break;
default:
- av_assert0(0);
+ av_assert1(0);
}
return 0;