lavfi: fix erroneous use of AV_PERM_PRESERVE in ff_inplace_start_frame.

[ffmpeg] / libavfilter / audio.c

diff --git a/libavfilter/audio.c b/libavfilter/audio.c
index 14896081a3ef67d2e1e6d6cc6d94693151dca21c..aa87488f5fc56ffb9568920360b22d93a09e2a58 100644 (file)
--- a/libavfilter/audio.c
+++ b/libavfilter/audio.c
@@ -1,22 +1,27 @@
 /*
- * 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 "libavutil/common.h"
 
 #include "audio.h"
 #include "avfilter.h"
@@ -96,9 +101,9 @@ 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;
@@ -146,23 +151,29 @@ fail:
     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_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 = 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) {
@@ -172,6 +183,10 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
 
         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;
 
@@ -185,6 +200,55 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
     } else
         buf_out = samplesref;
 
-    filter_samples(link, buf_out);
+    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;
+}