#include "libavutil/avassert.h"
#include "libavutil/audioconvert.h"
+#include "libavutil/common.h"
#include "audio.h"
#include "avfilter.h"
int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
int planes = planar ? nb_channels : 1;
int linesize;
+ int full_perms = AV_PERM_READ | AV_PERM_WRITE | AV_PERM_PRESERVE |
+ AV_PERM_REUSE | AV_PERM_REUSE2 | AV_PERM_ALIGN;
+
+ av_assert1(!(perms & ~(full_perms | AV_PERM_NEG_LINESIZES)));
if (!(data = av_mallocz(sizeof(*data) * planes)))
goto fail;
if (av_samples_alloc(data, &linesize, nb_channels, nb_samples, link->format, 0) < 0)
goto fail;
- samplesref = avfilter_get_audio_buffer_ref_from_arrays(data, linesize, perms,
+ samplesref = avfilter_get_audio_buffer_ref_from_arrays(data, linesize, full_perms,
nb_samples, link->format,
link->channel_layout);
if (!samplesref)
return NULL;
}
-static void default_filter_samples(AVFilterLink *link,
- AVFilterBufferRef *samplesref)
+static int default_filter_samples(AVFilterLink *link,
+ AVFilterBufferRef *samplesref)
{
- ff_filter_samples(link->dst->outputs[0], samplesref);
+ return ff_filter_samples(link->dst->outputs[0], samplesref);
}
-void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
+int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
- void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
+ int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
+ AVFilterPad *src = link->srcpad;
AVFilterPad *dst = link->dstpad;
int64_t pts;
AVFilterBufferRef *buf_out;
+ int ret;
FF_TPRINTF_START(NULL, filter_samples); ff_tlog_link(NULL, link, 1);
if (!(filter_samples = dst->filter_samples))
filter_samples = default_filter_samples;
+ av_assert1((samplesref->perms & src->min_perms) == src->min_perms);
+ samplesref->perms &= ~ src->rej_perms;
+
/* prepare to copy the samples if the buffer has insufficient permissions */
if ((dst->min_perms & samplesref->perms) != dst->min_perms ||
dst->rej_perms & samplesref->perms) {
buf_out = ff_default_get_audio_buffer(link, dst->min_perms,
samplesref->audio->nb_samples);
+ if (!buf_out) {
+ avfilter_unref_buffer(samplesref);
+ return AVERROR(ENOMEM);
+ }
buf_out->pts = samplesref->pts;
buf_out->audio->sample_rate = samplesref->audio->sample_rate;
link->cur_buf = buf_out;
pts = buf_out->pts;
- filter_samples(link, buf_out);
+ ret = filter_samples(link, buf_out);
ff_update_link_current_pts(link, pts);
+ return ret;
+}
+
+int 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);
+ int ret = 0;
+
+ if (!link->min_samples ||
+ (!pbuf &&
+ insamples >= link->min_samples && insamples <= link->max_samples)) {
+ return ff_filter_samples_framed(link, samplesref);
+ }
+ /* 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 0;
+ }
+ 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) {
+ ret = ff_filter_samples_framed(link, pbuf);
+ pbuf = NULL;
+ }
+ }
+ avfilter_unref_buffer(samplesref);
+ link->partial_buf = pbuf;
+ return ret;
}