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_unittest.cc b/webrtc/common_audio/audio_util_unittest.cc
new file mode 100644
index 0000000..9ffed73
--- /dev/null
+++ b/webrtc/common_audio/audio_util_unittest.cc
@@ -0,0 +1,55 @@
+/*
+ * 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 "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_audio/include/audio_util.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+void ExpectArraysEq(const int16_t* ref, const int16_t* test, int length) {
+ for (int i = 0; i < length; ++i) {
+ EXPECT_EQ(test[i], ref[i]);
+ }
+}
+
+TEST(AudioUtilTest, InterleavingStereo) {
+ const int16_t kInterleaved[] = {2, 3, 4, 9, 8, 27, 16, 81};
+ const int kSamplesPerChannel = 4;
+ const int kNumChannels = 2;
+ const int kLength = kSamplesPerChannel * kNumChannels;
+ int16_t left[kSamplesPerChannel], right[kSamplesPerChannel];
+ int16_t* deinterleaved[] = {left, right};
+ Deinterleave(kInterleaved, kSamplesPerChannel, kNumChannels, deinterleaved);
+ const int16_t kRefLeft[] = {2, 4, 8, 16};
+ const int16_t kRefRight[] = {3, 9, 27, 81};
+ ExpectArraysEq(left, kRefLeft, kSamplesPerChannel);
+ ExpectArraysEq(right, kRefRight, kSamplesPerChannel);
+
+ int16_t interleaved[kLength];
+ Interleave(deinterleaved, kSamplesPerChannel, kNumChannels, interleaved);
+ ExpectArraysEq(interleaved, kInterleaved, kLength);
+}
+
+TEST(AudioUtilTest, InterleavingMonoIsIdentical) {
+ const int16_t kInterleaved[] = {1, 2, 3, 4, 5};
+ const int kSamplesPerChannel = 5;
+ const int kNumChannels = 1;
+ int16_t mono[kSamplesPerChannel];
+ int16_t* deinterleaved[] = {mono};
+ Deinterleave(kInterleaved, kSamplesPerChannel, kNumChannels, deinterleaved);
+ ExpectArraysEq(mono, kInterleaved, kSamplesPerChannel);
+
+ int16_t interleaved[kSamplesPerChannel];
+ Interleave(deinterleaved, kSamplesPerChannel, kNumChannels, interleaved);
+ ExpectArraysEq(interleaved, mono, kSamplesPerChannel);
+}
+
+} // namespace webrtc