/*
- * This file is part of Libav.
+ * Copyright (c) Stefano Sabatini | stefasab at gmail.com
+ * Copyright (c) S.N. Hemanth Meenakshisundaram | smeenaks at ucsd.edu
*
- * Libav is free software; you can redistribute it and/or
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#include "libavutil/avassert.h"
#include "libavutil/audioconvert.h"
#include "audio.h"
samplesref->audio->nb_samples = nb_samples;
samplesref->audio->channel_layout = channel_layout;
- samplesref->audio->planar = av_sample_fmt_is_planar(sample_fmt);
- planes = samplesref->audio->planar ? av_get_channel_layout_nb_channels(channel_layout) : 1;
+ planes = av_sample_fmt_is_planar(sample_fmt) ?
+ av_get_channel_layout_nb_channels(channel_layout) : 1;
/* make sure the buffer gets read permission or it's useless for output */
samplesref->perms = perms | AV_PERM_READ;
return NULL;
}
-void ff_null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
-{
- ff_filter_samples(link->dst->outputs[0], samplesref);
-}
-
-/* FIXME: samplesref is same as link->cur_buf. Need to consider removing the redundant parameter. */
-void ff_default_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
+static int default_filter_samples(AVFilterLink *link,
+ AVFilterBufferRef *samplesref)
{
- AVFilterLink *outlink = NULL;
-
- if (inlink->dst->nb_outputs)
- outlink = inlink->dst->outputs[0];
-
- if (outlink) {
- outlink->out_buf = ff_default_get_audio_buffer(inlink, AV_PERM_WRITE,
- samplesref->audio->nb_samples);
- outlink->out_buf->pts = samplesref->pts;
- outlink->out_buf->audio->sample_rate = samplesref->audio->sample_rate;
- ff_filter_samples(outlink, avfilter_ref_buffer(outlink->out_buf, ~0));
- avfilter_unref_buffer(outlink->out_buf);
- outlink->out_buf = NULL;
- }
- avfilter_unref_buffer(samplesref);
- inlink->cur_buf = NULL;
+ 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 *dst = link->dstpad;
+ int64_t pts;
+ AVFilterBufferRef *buf_out;
+ int ret;
- FF_DPRINTF_START(NULL, filter_samples); ff_dlog_link(NULL, link, 1);
+ FF_TPRINTF_START(NULL, filter_samples); ff_tlog_link(NULL, link, 1);
if (!(filter_samples = dst->filter_samples))
- filter_samples = ff_default_filter_samples;
+ filter_samples = default_filter_samples;
/* 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) {
- int i, planar = av_sample_fmt_is_planar(samplesref->format);
- int planes = !planar ? 1:
- av_get_channel_layout_nb_channels(samplesref->audio->channel_layout);
-
av_log(link->dst, AV_LOG_DEBUG,
"Copying audio data in avfilter (have perms %x, need %x, reject %x)\n",
samplesref->perms, link->dstpad->min_perms, link->dstpad->rej_perms);
- link->cur_buf = ff_default_get_audio_buffer(link, dst->min_perms,
- samplesref->audio->nb_samples);
- link->cur_buf->pts = samplesref->pts;
- link->cur_buf->audio->sample_rate = samplesref->audio->sample_rate;
+ 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;
/* Copy actual data into new samples buffer */
- for (i = 0; i < planes; i++)
- memcpy(link->cur_buf->extended_data[i], samplesref->extended_data[i], samplesref->linesize[0]);
+ av_samples_copy(buf_out->extended_data, samplesref->extended_data,
+ 0, 0, samplesref->audio->nb_samples,
+ av_get_channel_layout_nb_channels(link->channel_layout),
+ link->format);
avfilter_unref_buffer(samplesref);
} else
- link->cur_buf = samplesref;
+ buf_out = samplesref;
- filter_samples(link, link->cur_buf);
+ link->cur_buf = buf_out;
+ pts = buf_out->pts;
+ 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;
+}
projects / ffmpeg / blobdiff