ffmpeg | branch: master | Paul B Mahol <[email protected]> | 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; _______________________________________________ ffmpeg-cvslog mailing list [email protected] https://ffmpeg.org/mailman/listinfo/ffmpeg-cvslog To unsubscribe, visit link above, or email [email protected] with subject "unsubscribe".
