Add a wrapper around PushSincResampler and the old Resampler.

The old resampler is used whenever it supports the requested rates. Otherwise
the sinc resampler is enabled.

Integrated with output_mixer in order to test the change through
output_mixer_unittest. The sinc resampler will not yet be used, since we don't
feed VoE with any rates that trigger it.

BUG=webrtc:1395
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1355004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3915 4adac7df-926f-26a2-2b94-8c16560cd09d

1 files changed, 33 insertions, 0 deletions

diff --git a/common_audio/include/audio_util.h b/common_audio/include/audio_util.h
new file mode 100644
index 00000000..2196fc34
--- /dev/null
+++ b/common_audio/include/audio_util.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
+#define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Deinterleave audio from |interleaved| to the channel buffers pointed to
+// by |deinterleaved|. There must be sufficient space allocated in the
+// |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
+// per buffer).
+void Deinterleave(const int16_t* interleaved, int samples_per_channel,
+                  int num_channels, int16_t** deinterleaved);
+
+// Interleave audio from the channel buffers pointed to by |deinterleaved| to
+// |interleaved|. There must be sufficient space allocated in |interleaved|
+// (|samples_per_channel| * |num_channels|).
+void Interleave(const int16_t* const* deinterleaved, int samples_per_channel,
+                int num_channels, int16_t* interleaved);
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_