Merge remote-tracking branch 'qatar/master'

[ffmpeg] / libavfilter / audio.h

diff --git a/libavfilter/audio.h b/libavfilter/audio.h
index a3775037e9f4426577929c1fb15771fe3bce5a08..8fe4d8ee00d2a2887735617fa7188164fc8eaa64 100644 (file)
--- a/libavfilter/audio.h
+++ b/libavfilter/audio.h
@@ -1,18 +1,21 @@
 /*
- * 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
  */
 
@@ -20,6 +23,25 @@
 #define AVFILTER_AUDIO_H
 
 #include "avfilter.h"
+#include "internal.h"
+
+static const enum AVSampleFormat ff_packed_sample_fmts_array[] = {
+    AV_SAMPLE_FMT_U8,
+    AV_SAMPLE_FMT_S16,
+    AV_SAMPLE_FMT_S32,
+    AV_SAMPLE_FMT_FLT,
+    AV_SAMPLE_FMT_DBL,
+    AV_SAMPLE_FMT_NONE
+};
+
+static const enum AVSampleFormat ff_planar_sample_fmts_array[] = {
+    AV_SAMPLE_FMT_U8P,
+    AV_SAMPLE_FMT_S16P,
+    AV_SAMPLE_FMT_S32P,
+    AV_SAMPLE_FMT_FLTP,
+    AV_SAMPLE_FMT_DBLP,
+    AV_SAMPLE_FMT_NONE
+};
 
 /** default handler for get_audio_buffer() for audio inputs */
 AVFilterBufferRef *ff_default_get_audio_buffer(AVFilterLink *link, int perms,
@@ -42,4 +64,24 @@ AVFilterBufferRef *ff_null_get_audio_buffer(AVFilterLink *link, int perms,
 AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
                                              int nb_samples);
 
+/**
+ * Send a buffer of audio samples to the next filter.
+ *
+ * @param link       the output link over which the audio samples are being sent
+ * @param samplesref a reference to the buffer of audio samples being sent. The
+ *                   receiving filter will free this reference when it no longer
+ *                   needs it or pass it on to the next filter.
+ *
+ * @return >= 0 on success, a negative AVERROR on error. The receiving filter
+ * is responsible for unreferencing samplesref in case of error.
+ */
+int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref);
+
+/**
+ * Send a buffer of audio samples to the next link, without checking
+ * min_samples.
+ */
+int ff_filter_samples_framed(AVFilterLink *link,
+                             AVFilterBufferRef *samplesref);
+
 #endif /* AVFILTER_AUDIO_H */