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https://github.com/FFmpeg/FFmpeg.git
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avfilter/af_speechnorm: speed up filtering code
Reduce some asserts by default.
This commit is contained in:
Paul B Mahol
parent
52218c0322
commit
698de27f25
@ -128,7 +128,7 @@ static int get_pi_samples(PeriodItem *pi, int start, int end, int remain)
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start = 0;
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if (pi[start].type == 0)
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break;
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av_assert0(pi[start].size > 0);
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av_assert1(pi[start].size > 0);
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sum += pi[start].size;
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}
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@ -156,7 +156,7 @@ static void consume_pi(ChannelContext *cc, int nb_samples)
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if (cc->pi_size >= nb_samples) {
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cc->pi_size -= nb_samples;
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} else {
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av_assert0(0);
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av_assert1(0);
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}
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}
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@ -178,16 +178,16 @@ static double next_gain(AVFilterContext *ctx, double pi_max_peak, int bypass, do
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static void next_pi(AVFilterContext *ctx, ChannelContext *cc, int bypass)
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{
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av_assert0(cc->pi_size >= 0);
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av_assert1(cc->pi_size >= 0);
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if (cc->pi_size == 0) {
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SpeechNormalizerContext *s = ctx->priv;
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int start = cc->pi_start;
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av_assert0(cc->pi[start].size > 0);
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av_assert1(cc->pi[start].size > 0);
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av_assert0(cc->pi[start].type > 0 || s->eof);
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cc->pi_size = cc->pi[start].size;
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cc->pi_max_peak = cc->pi[start].max_peak;
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av_assert0(cc->pi_start != cc->pi_end || s->eof);
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av_assert1(cc->pi_start != cc->pi_end || s->eof);
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start++;
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if (start >= MAX_ITEMS)
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start = 0;
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@ -219,58 +219,71 @@ static double min_gain(AVFilterContext *ctx, ChannelContext *cc, int max_size)
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return min_gain;
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}
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#define ANALYZE_CHANNEL(name, ptype, zero, min_peak) \
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static void analyze_channel_## name (AVFilterContext *ctx, ChannelContext *cc, \
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const uint8_t *srcp, int nb_samples) \
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{ \
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SpeechNormalizerContext *s = ctx->priv; \
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const ptype *src = (const ptype *)srcp; \
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int n = 0; \
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\
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if (cc->state < 0) \
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cc->state = src[0] >= zero; \
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\
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while (n < nb_samples) { \
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if ((cc->state != (src[n] >= zero)) || \
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(cc->pi[cc->pi_end].size > s->max_period)) { \
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ptype max_peak = cc->pi[cc->pi_end].max_peak; \
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int state = cc->state; \
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cc->state = src[n] >= zero; \
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av_assert0(cc->pi[cc->pi_end].size > 0); \
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if (max_peak >= min_peak || \
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cc->pi[cc->pi_end].size > s->max_period) { \
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cc->pi[cc->pi_end].type = 1; \
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cc->pi_end++; \
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if (cc->pi_end >= MAX_ITEMS) \
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cc->pi_end = 0; \
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if (cc->state != state) \
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cc->pi[cc->pi_end].max_peak = DBL_MIN; \
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else \
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cc->pi[cc->pi_end].max_peak = max_peak; \
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cc->pi[cc->pi_end].type = 0; \
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cc->pi[cc->pi_end].size = 0; \
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av_assert0(cc->pi_end != cc->pi_start); \
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} \
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} \
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\
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if (cc->state) { \
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while (src[n] >= zero) { \
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cc->pi[cc->pi_end].max_peak = FFMAX(cc->pi[cc->pi_end].max_peak, src[n]); \
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cc->pi[cc->pi_end].size++; \
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n++; \
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if (n >= nb_samples) \
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break; \
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} \
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} else { \
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while (src[n] < zero) { \
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cc->pi[cc->pi_end].max_peak = FFMAX(cc->pi[cc->pi_end].max_peak, -src[n]); \
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cc->pi[cc->pi_end].size++; \
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n++; \
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if (n >= nb_samples) \
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break; \
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} \
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} \
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} \
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#define ANALYZE_CHANNEL(name, ptype, zero, min_peak) \
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static void analyze_channel_## name (AVFilterContext *ctx, ChannelContext *cc, \
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const uint8_t *srcp, int nb_samples) \
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{ \
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SpeechNormalizerContext *s = ctx->priv; \
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const ptype *src = (const ptype *)srcp; \
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const int max_period = s->max_period; \
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PeriodItem *pi = (PeriodItem *)&cc->pi; \
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int pi_end = cc->pi_end; \
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int n = 0; \
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\
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if (cc->state < 0) \
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cc->state = src[0] >= zero; \
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\
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while (n < nb_samples) { \
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ptype new_max_peak; \
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int new_size; \
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\
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if ((cc->state != (src[n] >= zero)) || \
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(pi[pi_end].size > max_period)) { \
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ptype max_peak = pi[pi_end].max_peak; \
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int state = cc->state; \
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\
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cc->state = src[n] >= zero; \
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av_assert1(pi[pi_end].size > 0); \
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if (max_peak >= min_peak || \
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pi[pi_end].size > max_period) { \
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pi[pi_end].type = 1; \
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pi_end++; \
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if (pi_end >= MAX_ITEMS) \
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pi_end = 0; \
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if (cc->state != state) \
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pi[pi_end].max_peak = DBL_MIN; \
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else \
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pi[pi_end].max_peak = max_peak; \
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pi[pi_end].type = 0; \
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pi[pi_end].size = 0; \
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av_assert1(pi_end != cc->pi_start); \
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} \
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} \
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\
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new_max_peak = pi[pi_end].max_peak; \
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new_size = pi[pi_end].size; \
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if (cc->state) { \
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while (src[n] >= zero) { \
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new_max_peak = FFMAX(new_max_peak, src[n]); \
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new_size++; \
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n++; \
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if (n >= nb_samples) \
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break; \
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} \
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} else { \
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while (src[n] < zero) { \
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new_max_peak = FFMAX(new_max_peak, -src[n]); \
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new_size++; \
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n++; \
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if (n >= nb_samples) \
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break; \
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} \
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} \
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\
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pi[pi_end].max_peak = new_max_peak; \
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pi[pi_end].size = new_size; \
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} \
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cc->pi_end = pi_end; \
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}
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ANALYZE_CHANNEL(dbl, double, 0.0, MIN_PEAK)
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@ -296,7 +309,7 @@ static void filter_channels_## name (AVFilterContext *ctx,
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\
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next_pi(ctx, cc, bypass); \
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size = FFMIN(nb_samples - n, cc->pi_size); \
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av_assert0(size > 0); \
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av_assert1(size > 0); \
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gain = cc->gain_state; \
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consume_pi(cc, size); \
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for (int i = n; !ctx->is_disabled && i < n + size; i++) \
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@ -343,7 +356,7 @@ static void filter_link_channels_## name (AVFilterContext *ctx,
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max_size = FFMAX(max_size, cc->pi_size); \
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} \
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\
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av_assert0(min_size > 0); \
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av_assert1(min_size > 0); \
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for (int ch = 0; ch < inlink->channels; ch++) { \
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ChannelContext *cc = &s->cc[ch]; \
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\
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@ -509,7 +522,7 @@ static int config_input(AVFilterLink *inlink)
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s->filter_channels[1] = filter_link_channels_dbl;
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break;
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default:
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av_assert0(0);
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av_assert1(0);
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}
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return 0;
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