diff options
author | Paul B Mahol <onemda@gmail.com> | 2020-06-21 21:25:56 +0200 |
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committer | Paul B Mahol <onemda@gmail.com> | 2020-06-21 21:35:40 +0200 |
commit | 683a1599d449ca434466742ba73371d2763bc7b9 (patch) | |
tree | 961677914dcd7872f5514dd4d63c9d5f1b31f59c | |
parent | 842bc312ade8fab82465423b22c4fbe3bee63383 (diff) | |
download | ffmpeg-683a1599d449ca434466742ba73371d2763bc7b9.tar.gz |
avfilter/af_ladspa: add latency compensation
-rw-r--r-- | doc/filters.texi | 3 | ||||
-rw-r--r-- | libavfilter/af_ladspa.c | 60 |
2 files changed, 60 insertions, 3 deletions
diff --git a/doc/filters.texi b/doc/filters.texi index 5f0eb28f76..551604a143 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -4197,6 +4197,9 @@ If not specified, or the expressed duration is negative, the audio is supposed to be generated forever. Only used if plugin have zero inputs. +@item latency, l +Enable latency compensation, by default is disabled. +Only used if plugin have inputs. @end table @subsection Examples diff --git a/libavfilter/af_ladspa.c b/libavfilter/af_ladspa.c index 68537c5029..0d7ceb2777 100644 --- a/libavfilter/af_ladspa.c +++ b/libavfilter/af_ladspa.c @@ -64,6 +64,9 @@ typedef struct LADSPAContext { int nb_samples; int64_t pts; int64_t duration; + int in_trim; + int out_pad; + int latency; } LADSPAContext; #define OFFSET(x) offsetof(LADSPAContext, x) @@ -81,11 +84,28 @@ static const AVOption ladspa_options[] = { { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS }, { "duration", "set audio duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64=-1}, -1, INT64_MAX, FLAGS }, { "d", "set audio duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64=-1}, -1, INT64_MAX, FLAGS }, + { "latency", "enable latency compensation", OFFSET(latency), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS }, + { "l", "enable latency compensation", OFFSET(latency), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS }, { NULL } }; AVFILTER_DEFINE_CLASS(ladspa); +static int find_latency(AVFilterContext *ctx, LADSPAContext *s) +{ + int latency = 0; + + for (int ctl = 0; ctl < s->nb_outputcontrols; ctl++) { + if (av_strcasecmp("latency", s->desc->PortNames[s->ocmap[ctl]])) + continue; + + latency = lrintf(s->octlv[ctl]); + break; + } + + return latency; +} + static void print_ctl_info(AVFilterContext *ctx, int level, LADSPAContext *s, int ctl, unsigned long *map, LADSPA_Data *values, int print) @@ -143,12 +163,13 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) AVFilterContext *ctx = inlink->dst; LADSPAContext *s = ctx->priv; AVFrame *out; - int i, h, p; + int i, h, p, new_out_samples; av_assert0(in->channels == (s->nb_inputs * s->nb_handles)); if (!s->nb_outputs || (av_frame_is_writable(in) && s->nb_inputs == s->nb_outputs && + s->in_trim == 0 && s->out_pad == 0 && !(s->desc->Properties & LADSPA_PROPERTY_INPLACE_BROKEN))) { out = in; } else { @@ -176,6 +197,9 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) } s->desc->run(s->handles[h], in->nb_samples); + if (s->latency) + s->in_trim = s->out_pad = find_latency(ctx, s); + s->latency = 0; } for (i = 0; i < s->nb_outputcontrols; i++) @@ -184,6 +208,25 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) if (out != in) av_frame_free(&in); + new_out_samples = out->nb_samples; + if (s->in_trim > 0) { + int trim = FFMIN(new_out_samples, s->in_trim); + + new_out_samples -= trim; + s->in_trim -= trim; + } + + if (new_out_samples <= 0) { + av_frame_free(&out); + return 0; + } else if (new_out_samples < out->nb_samples) { + int offset = out->nb_samples - new_out_samples; + for (int ch = 0; ch < out->channels; ch++) + memmove(out->extended_data[ch], out->extended_data[ch] + sizeof(float) * offset, + sizeof(float) * new_out_samples); + out->nb_samples = new_out_samples; + } + return ff_filter_frame(ctx->outputs[0], out); } @@ -195,8 +238,19 @@ static int request_frame(AVFilterLink *outlink) int64_t t; int i; - if (ctx->nb_inputs) - return ff_request_frame(ctx->inputs[0]); + if (ctx->nb_inputs) { + int ret = ff_request_frame(ctx->inputs[0]); + + if (ret == AVERROR_EOF && s->out_pad > 0) { + AVFrame *frame = ff_get_audio_buffer(outlink, FFMIN(2048, s->out_pad)); + if (!frame) + return AVERROR(ENOMEM); + + s->out_pad -= frame->nb_samples; + return filter_frame(ctx->inputs[0], frame); + } + return ret; + } t = av_rescale(s->pts, AV_TIME_BASE, s->sample_rate); if (s->duration >= 0 && t >= s->duration) |