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avfilter/af_axcorrelate: add another algorithm for calculation
Rewrite EOF logic while here.
This commit is contained in:
Paul B Mahol
parent
f032234953
commit
c183f840fa
@ -3762,8 +3762,8 @@ Set size of segment over which cross-correlation is calculated.
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Default is 256. Allowed range is from 2 to 131072.
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@item algo
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Set algorithm for cross-correlation. Can be @code{slow} or @code{fast}.
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Default is @code{slow}. Fast algorithm assumes mean values over any given segment
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Set algorithm for cross-correlation. Can be @code{slow} or @code{fast} or @code{best}.
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Default is @code{best}. Fast algorithm assumes mean values over any given segment
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are always zero and thus need much less calculations to make.
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This is generally not true, but is valid for typical audio streams.
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@end table
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@ -110,7 +110,7 @@ static int xcorrelate_slow_##suffix(AVFilterContext *ctx, \
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AVFrame *out, int available) \
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{ \
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AudioXCorrelateContext *s = ctx->priv; \
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const int size = FFMIN(available, s->size); \
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const int size = s->size; \
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int used; \
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\
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for (int ch = 0; ch < out->ch_layout.nb_channels; ch++) { \
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@ -127,13 +127,13 @@ static int xcorrelate_slow_##suffix(AVFilterContext *ctx, \
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used = 1; \
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} \
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\
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for (int n = 0; n < out->nb_samples; n++) { \
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const int idx = available <= s->size ? out->nb_samples - n - 1 : n + size; \
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\
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dst[n] = xcorrelate_##suffix(x + n, y + n, \
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sumx[0], sumy[0], \
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size); \
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\
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for (int n = 0; n < out->nb_samples; n++) { \
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const int idx = n + size; \
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\
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dst[n] = xcorrelate_##suffix(x + n, y + n, \
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sumx[0], sumy[0],\
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size); \
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\
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sumx[0] -= x[n]; \
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sumx[0] += x[idx]; \
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sumy[0] -= y[n]; \
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@ -147,12 +147,15 @@ static int xcorrelate_slow_##suffix(AVFilterContext *ctx, \
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XCORRELATE_SLOW(f, float)
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XCORRELATE_SLOW(d, double)
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#define XCORRELATE_FAST(suffix, type, zero, small, sqrtfun) \
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#define clipf(x) (av_clipf(x, -1.f, 1.f))
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#define clipd(x) (av_clipd(x, -1.0, 1.0))
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#define XCORRELATE_FAST(suffix, type, zero, small, sqrtfun, CLIP) \
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static int xcorrelate_fast_##suffix(AVFilterContext *ctx, AVFrame *out, \
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int available) \
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{ \
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AudioXCorrelateContext *s = ctx->priv; \
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const int size = FFMIN(available, s->size); \
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const int size = s->size; \
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int used; \
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\
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for (int ch = 0; ch < out->ch_layout.nb_channels; ch++) { \
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@ -171,14 +174,14 @@ static int xcorrelate_fast_##suffix(AVFilterContext *ctx, AVFrame *out, \
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used = 1; \
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} \
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\
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for (int n = 0; n < out->nb_samples; n++) { \
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const int idx = available <= s->size ? out->nb_samples - n - 1 : n + size; \
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type num, den; \
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for (int n = 0; n < out->nb_samples; n++) { \
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const int idx = n + size; \
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type num, den; \
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\
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num = num_sum[0] / size; \
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den = sqrtfun((den_sumx[0] * den_sumy[0]) / size / size); \
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\
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dst[n] = den <= small ? zero : num / den; \
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dst[n] = den <= small ? zero : CLIP(num / den); \
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\
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num_sum[0] -= x[n] * y[n]; \
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num_sum[0] += x[idx] * y[idx]; \
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@ -194,20 +197,82 @@ static int xcorrelate_fast_##suffix(AVFilterContext *ctx, AVFrame *out, \
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return used; \
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}
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XCORRELATE_FAST(f, float, 0.f, 1e-6f, sqrtf)
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XCORRELATE_FAST(d, double, 0.0, 1e-9, sqrt)
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XCORRELATE_FAST(f, float, 0.f, 1e-6f, sqrtf, clipf)
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XCORRELATE_FAST(d, double, 0.0, 1e-9, sqrt, clipd)
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#define XCORRELATE_BEST(suffix, type, zero, small, sqrtfun, FMAX, CLIP) \
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static int xcorrelate_best_##suffix(AVFilterContext *ctx, AVFrame *out, \
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int available) \
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{ \
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AudioXCorrelateContext *s = ctx->priv; \
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const int size = s->size; \
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int used; \
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\
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for (int ch = 0; ch < out->ch_layout.nb_channels; ch++) { \
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const type *x = (const type *)s->cache[0]->extended_data[ch]; \
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const type *y = (const type *)s->cache[1]->extended_data[ch]; \
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type *mean_sumx = (type *)s->mean_sum[0]->extended_data[ch]; \
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type *mean_sumy = (type *)s->mean_sum[1]->extended_data[ch]; \
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type *num_sum = (type *)s->num_sum->extended_data[ch]; \
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type *den_sumx = (type *)s->den_sum[0]->extended_data[ch]; \
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type *den_sumy = (type *)s->den_sum[1]->extended_data[ch]; \
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type *dst = (type *)out->extended_data[ch]; \
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\
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used = s->used; \
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if (!used) { \
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num_sum[0] = square_sum_##suffix(x, y, size); \
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den_sumx[0] = square_sum_##suffix(x, x, size); \
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den_sumy[0] = square_sum_##suffix(y, y, size); \
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mean_sumx[0] = mean_sum_##suffix(x, size); \
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mean_sumy[0] = mean_sum_##suffix(y, size); \
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used = 1; \
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} \
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\
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for (int n = 0; n < out->nb_samples; n++) { \
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const int idx = n + size; \
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type num, den, xm, ym; \
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\
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xm = mean_sumx[0] / size; \
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ym = mean_sumy[0] / size; \
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num = num_sum[0] - size * xm * ym; \
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den = sqrtfun(FMAX(den_sumx[0] - size * xm * xm, zero)) * \
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sqrtfun(FMAX(den_sumy[0] - size * ym * ym, zero)); \
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\
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dst[n] = den <= small ? zero : CLIP(num / den); \
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\
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mean_sumx[0]-= x[n]; \
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mean_sumx[0]+= x[idx]; \
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mean_sumy[0]-= y[n]; \
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mean_sumy[0]+= y[idx]; \
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num_sum[0] -= x[n] * y[n]; \
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num_sum[0] += x[idx] * y[idx]; \
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den_sumx[0] -= x[n] * x[n]; \
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den_sumx[0] += x[idx] * x[idx]; \
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den_sumx[0] = FMAX(den_sumx[0], zero); \
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den_sumy[0] -= y[n] * y[n]; \
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den_sumy[0] += y[idx] * y[idx]; \
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den_sumy[0] = FMAX(den_sumy[0], zero); \
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} \
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} \
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\
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return used; \
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}
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XCORRELATE_BEST(f, float, 0.f, 1e-6f, sqrtf, fmaxf, clipf)
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XCORRELATE_BEST(d, double, 0.0, 1e-9, sqrt, fmax, clipd)
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static int activate(AVFilterContext *ctx)
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{
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AudioXCorrelateContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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AVFrame *frame = NULL;
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int ret, status;
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int available;
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int64_t pts;
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FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
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FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
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for (int i = 0; i < 2; i++) {
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for (int i = 0; i < 2 && !s->eof; i++) {
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ret = ff_inlink_consume_frame(ctx->inputs[i], &frame);
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if (ret > 0) {
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if (s->pts == AV_NOPTS_VALUE)
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@ -221,20 +286,20 @@ static int activate(AVFilterContext *ctx)
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}
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available = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
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if (available > s->size || (s->eof && available > 0)) {
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const int out_samples = s->eof ? available : available - s->size;
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if (available > s->size) {
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const int out_samples = available - s->size;
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AVFrame *out;
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if (!s->cache[0] || s->cache[0]->nb_samples < available) {
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av_frame_free(&s->cache[0]);
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s->cache[0] = ff_get_audio_buffer(ctx->outputs[0], available);
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s->cache[0] = ff_get_audio_buffer(outlink, available);
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if (!s->cache[0])
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return AVERROR(ENOMEM);
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}
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if (!s->cache[1] || s->cache[1]->nb_samples < available) {
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av_frame_free(&s->cache[1]);
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s->cache[1] = ff_get_audio_buffer(ctx->outputs[0], available);
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s->cache[1] = ff_get_audio_buffer(outlink, available);
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if (!s->cache[1])
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return AVERROR(ENOMEM);
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}
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@ -247,7 +312,7 @@ static int activate(AVFilterContext *ctx)
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if (ret < 0)
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return ret;
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out = ff_get_audio_buffer(ctx->outputs[0], out_samples);
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out = ff_get_audio_buffer(outlink, out_samples);
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if (!out)
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return AVERROR(ENOMEM);
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@ -259,18 +324,31 @@ static int activate(AVFilterContext *ctx)
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av_audio_fifo_drain(s->fifo[0], out_samples);
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av_audio_fifo_drain(s->fifo[1], out_samples);
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return ff_filter_frame(ctx->outputs[0], out);
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return ff_filter_frame(outlink, out);
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}
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for (int i = 0; i < 2 && !s->eof; i++) {
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if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts))
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if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
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AVFrame *silence = ff_get_audio_buffer(outlink, s->size);
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s->eof = 1;
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if (!silence)
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return AVERROR(ENOMEM);
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av_audio_fifo_write(s->fifo[0], (void **)silence->extended_data,
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silence->nb_samples);
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av_audio_fifo_write(s->fifo[1], (void **)silence->extended_data,
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silence->nb_samples);
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av_frame_free(&silence);
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}
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}
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if (s->eof &&
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(av_audio_fifo_size(s->fifo[0]) <= 0 ||
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av_audio_fifo_size(s->fifo[1]) <= 0)) {
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ff_outlink_set_status(ctx->outputs[0], AVERROR_EOF, s->pts);
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(av_audio_fifo_size(s->fifo[0]) <= s->size ||
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av_audio_fifo_size(s->fifo[1]) <= s->size)) {
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ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
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return 0;
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}
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@ -280,7 +358,7 @@ static int activate(AVFilterContext *ctx)
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return 0;
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}
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if (ff_outlink_frame_wanted(ctx->outputs[0]) && !s->eof) {
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if (ff_outlink_frame_wanted(outlink) && !s->eof) {
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for (int i = 0; i < 2; i++) {
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if (av_audio_fifo_size(s->fifo[i]) > s->size)
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continue;
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@ -316,12 +394,14 @@ static int config_output(AVFilterLink *outlink)
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switch (s->algo) {
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case 0: s->xcorrelate = xcorrelate_slow_f; break;
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case 1: s->xcorrelate = xcorrelate_fast_f; break;
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case 2: s->xcorrelate = xcorrelate_best_f; break;
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}
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if (outlink->format == AV_SAMPLE_FMT_DBLP) {
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switch (s->algo) {
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case 0: s->xcorrelate = xcorrelate_slow_d; break;
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case 1: s->xcorrelate = xcorrelate_fast_d; break;
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case 2: s->xcorrelate = xcorrelate_best_d; break;
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}
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}
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@ -366,10 +446,11 @@ static const AVFilterPad outputs[] = {
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#define OFFSET(x) offsetof(AudioXCorrelateContext, x)
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static const AVOption axcorrelate_options[] = {
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{ "size", "set segment size", OFFSET(size), AV_OPT_TYPE_INT, {.i64=256}, 2, 131072, AF },
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{ "algo", "set algorithm", OFFSET(algo), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "algo" },
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{ "size", "set the segment size", OFFSET(size), AV_OPT_TYPE_INT, {.i64=256}, 2, 131072, AF },
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{ "algo", "set the algorithm", OFFSET(algo), AV_OPT_TYPE_INT, {.i64=2}, 0, 2, AF, "algo" },
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{ "slow", "slow algorithm", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "algo" },
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{ "fast", "fast algorithm", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "algo" },
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{ "best", "best algorithm", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "algo" },
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{ NULL }
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};
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