Commit 0689d5e1 authored by Nicolas George's avatar Nicolas George

lavfi: implement samples framing on links.

Links can be set up to group samples into buffers of
specified minimum and maximum size.
parent c9c4835f
......@@ -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;
}
......@@ -73,4 +73,11 @@ AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
*/
void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref);
/**
* Send a buffer of audio samples to the next link, without checking
* min_samples.
*/
void ff_filter_samples_framed(AVFilterLink *link,
AVFilterBufferRef *samplesref);
#endif /* AVFILTER_AUDIO_H */
......@@ -28,6 +28,7 @@
#include "avfilter.h"
#include "formats.h"
#include "internal.h"
#include "audio.h"
char *ff_get_ref_perms_string(char *buf, size_t buf_size, int perms)
{
......@@ -320,13 +321,20 @@ void ff_tlog_link(void *ctx, AVFilterLink *link, int end)
int ff_request_frame(AVFilterLink *link)
{
int ret = -1;
FF_TPRINTF_START(NULL, request_frame); ff_tlog_link(NULL, link, 1);
if (link->srcpad->request_frame)
return link->srcpad->request_frame(link);
ret = link->srcpad->request_frame(link);
else if (link->src->inputs[0])
return ff_request_frame(link->src->inputs[0]);
else return -1;
ret = ff_request_frame(link->src->inputs[0]);
if (ret == AVERROR_EOF && link->partial_buf) {
AVFilterBufferRef *pbuf = link->partial_buf;
link->partial_buf = NULL;
ff_filter_samples_framed(link, pbuf);
return 0;
}
return ret;
}
int ff_poll_frame(AVFilterLink *link)
......
......@@ -590,6 +590,32 @@ struct AVFilterLink {
* It is similar to the r_frae_rate field in AVStream.
*/
AVRational frame_rate;
/**
* Buffer partially filled with samples to achieve a fixed/minimum size.
*/
AVFilterBufferRef *partial_buf;
/**
* Size of the partial buffer to allocate.
* Must be between min_samples and max_samples.
*/
int partial_buf_size;
/**
* Minimum number of samples to filter at once. If filter_samples() is
* called with fewer samples, it will accumulate them in partial_buf.
* This field and the related ones must not be changed after filtering
* has started.
* If 0, all related fields are ignored.
*/
int min_samples;
/**
* Maximum number of samples to filter at once. If filter_samples() is
* called with more samples, it will split them.
*/
int max_samples;
};
/**
......
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