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Unified Diff: webrtc/modules/audio_processing/audio_buffer_unittest.cc

Issue 2053773002: Keep track of the user-facing number of channels in a ChannelBuffer (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebasing Created 4 years, 6 months ago
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Index: webrtc/modules/audio_processing/audio_buffer_unittest.cc
diff --git a/webrtc/modules/audio_processing/audio_buffer_unittest.cc b/webrtc/modules/audio_processing/audio_buffer_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..c6a5e1daad6f4b90f4fb6977d08fd84453db3a99
--- /dev/null
+++ b/webrtc/modules/audio_processing/audio_buffer_unittest.cc
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 2016 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/modules/audio_processing/audio_buffer.h"
+
+namespace webrtc {
+
+namespace {
+
+const size_t kNumFrames = 480u;
+const size_t kStereo = 2u;
+const size_t kMono = 1u;
+
+void ExpectNumChannels(const AudioBuffer& ab, size_t num_channels) {
+ EXPECT_EQ(ab.data()->num_channels(), num_channels);
+ EXPECT_EQ(ab.data_f()->num_channels(), num_channels);
+ EXPECT_EQ(ab.split_data()->num_channels(), num_channels);
+ EXPECT_EQ(ab.split_data_f()->num_channels(), num_channels);
+ EXPECT_EQ(ab.num_channels(), num_channels);
+}
+
+} // namespace
+
+TEST(AudioBufferTest, SetNumChannelsSetsChannelBuffersNumChannels) {
+ AudioBuffer ab(kNumFrames, kStereo, kNumFrames, kStereo, kNumFrames);
+ ExpectNumChannels(ab, kStereo);
+ ab.set_num_channels(kMono);
+ ExpectNumChannels(ab, kMono);
+ ab.InitForNewData();
+ ExpectNumChannels(ab, kStereo);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(AudioBufferTest, SetNumChannelsDeathTest) {
+ AudioBuffer ab(kNumFrames, kMono, kNumFrames, kMono, kNumFrames);
+ EXPECT_DEATH(ab.set_num_channels(kStereo), "num_channels");
+}
+#endif
+
+} // namespace webrtc
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