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Diffstat (limited to 'libavfilter/audio.c')
-rw-r--r--libavfilter/audio.c48
1 files changed, 47 insertions, 1 deletions
diff --git a/libavfilter/audio.c b/libavfilter/audio.c
index 6a86597342..0ebec3c2d0 100644
--- a/libavfilter/audio.c
+++ b/libavfilter/audio.c
@@ -156,7 +156,8 @@ static void default_filter_samples(AVFilterLink *link,
ff_filter_samples(link->dst->outputs[0], samplesref);
}
-void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
+void ff_filter_samples_framed(AVFilterLink *link,
+ AVFilterBufferRef *samplesref)
{
void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
AVFilterPad *dst = link->dstpad;
@@ -195,3 +196,48 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
filter_samples(link, buf_out);
ff_update_link_current_pts(link, pts);
}
+
+void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
+{
+ int insamples = samplesref->audio->nb_samples, inpos = 0, nb_samples;
+ AVFilterBufferRef *pbuf = link->partial_buf;
+ int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
+
+ if (!link->min_samples ||
+ (!pbuf &&
+ insamples >= link->min_samples && insamples <= link->max_samples)) {
+ ff_filter_samples_framed(link, samplesref);
+ return;
+ }
+ /* Handle framing (min_samples, max_samples) */
+ while (insamples) {
+ if (!pbuf) {
+ AVRational samples_tb = { 1, link->sample_rate };
+ int perms = link->dstpad->min_perms | AV_PERM_WRITE;
+ pbuf = ff_get_audio_buffer(link, perms, link->partial_buf_size);
+ if (!pbuf) {
+ av_log(link->dst, AV_LOG_WARNING,
+ "Samples dropped due to memory allocation failure.\n");
+ return;
+ }
+ avfilter_copy_buffer_ref_props(pbuf, samplesref);
+ pbuf->pts = samplesref->pts +
+ av_rescale_q(inpos, samples_tb, link->time_base);
+ pbuf->audio->nb_samples = 0;
+ }
+ nb_samples = FFMIN(insamples,
+ link->partial_buf_size - pbuf->audio->nb_samples);
+ av_samples_copy(pbuf->extended_data, samplesref->extended_data,
+ pbuf->audio->nb_samples, inpos,
+ nb_samples, nb_channels, link->format);
+ inpos += nb_samples;
+ insamples -= nb_samples;
+ pbuf->audio->nb_samples += nb_samples;
+ if (pbuf->audio->nb_samples >= link->min_samples) {
+ ff_filter_samples_framed(link, pbuf);
+ pbuf = NULL;
+ }
+ }
+ avfilter_unref_buffer(samplesref);
+ link->partial_buf = pbuf;
+}