On Wed, Apr 01, 2015 at 04:04:57PM +0000, Paul B Mahol wrote:
> Signed-off-by: Paul B Mahol <one...@gmail.com>
[...]
> +static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + ChorusContext *s = ctx->priv;
> + AVFrame *out_frame;
> + int c, i, n;
> +
> + if (av_frame_is_writable(frame)) {
> + out_frame = frame;
> + } else {
> + out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
> + if (!out_frame)
> + return AVERROR(ENOMEM);
> + av_frame_copy_props(out_frame, frame);
> + }
> +
> + for (c = 0; c < inlink->channels; c++) {
> + const float *src = (const float *)frame->extended_data[c];
> + float *dst = (float *)out_frame->extended_data[c];
> + float *chorusbuf = s->chorusbuf[c];
> + int *phase = s->phase[c];
> +
> + for (i = 0; i < frame->nb_samples; i++) {
> + float out, in = src[i];
> +
> + out = in * s->in_gain;
> +> + for (n = 0; n < s->num_chorus; n++) > + out += chorusbuf[(s->max_samples + s->counter[c] - > + s->lookup_table[n][phase[n]]) % > + s->max_samples] * s->decays[n]; the % should be avoided in the inner loop as its probably a major factor in the performace of the filter if there are many iterations a power of 2 could be used or alternatively the buffer could be reorganized in some other way like writing to a larger area than max_samples, ensuring that theres no need to handle wrap around but rather write more samples occasionally also other minor things could likely be factored out of the loop [...] -- Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB Breaking DRM is a little like attempting to break through a door even though the window is wide open and the only thing in the house is a bunch of things you dont want and which you would get tomorrow for free anyway
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