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-rw-r--r--libavfilter/audio.c23
1 files changed, 15 insertions, 8 deletions
diff --git a/libavfilter/audio.c b/libavfilter/audio.c
index a6fef9d..d473b99 100644
--- a/libavfilter/audio.c
+++ b/libavfilter/audio.c
@@ -1,21 +1,25 @@
/*
- * 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"
@@ -96,9 +100,9 @@ AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays(uint8_t **data,
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;
@@ -176,8 +180,9 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
AVFilterPad *dst = link->dstpad;
+ int64_t pts;
- 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;
@@ -185,6 +190,7 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
/* 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 size;
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);
@@ -204,6 +210,7 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
} else
link->cur_buf = samplesref;
+ pts = link->cur_buf->pts;
filter_samples(link, link->cur_buf);
+ ff_update_link_current_pts(link, pts);
}
-
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