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@3915 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/common_audio/audio_util.cc b/webrtc/common_audio/audio_util.cc
new file mode 100644
index 0000000..a6114fd
--- /dev/null
+++ b/webrtc/common_audio/audio_util.cc
@@ -0,0 +1,41 @@
+/*
+ * 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.
+ */
+
+#include "webrtc/common_audio/include/audio_util.h"
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+void Deinterleave(const int16_t* interleaved, int samples_per_channel,
+ int num_channels, int16_t** deinterleaved) {
+ for (int i = 0; i < num_channels; i++) {
+ int16_t* channel = deinterleaved[i];
+ int interleaved_idx = i;
+ for (int j = 0; j < samples_per_channel; j++) {
+ channel[j] = interleaved[interleaved_idx];
+ interleaved_idx += num_channels;
+ }
+ }
+}
+
+void Interleave(const int16_t* const* deinterleaved, int samples_per_channel,
+ int num_channels, int16_t* interleaved) {
+ for (int i = 0; i < num_channels; ++i) {
+ const int16_t* channel = deinterleaved[i];
+ int interleaved_idx = i;
+ for (int j = 0; j < samples_per_channel; j++) {
+ interleaved[interleaved_idx] = channel[j];
+ interleaved_idx += num_channels;
+ }
+ }
+}
+
+} // namespace webrtc
