Keep track of the user-facing number of channels in a ChannelBuffer

webrtc/modules/audio_processing/audio_buffer_unittest.cc

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..1ad532205336fba7fb2b2070fcbe5866f04c3777
--- /dev/null
+++ b/webrtc/modules/audio_processing/audio_buffer_unittest.cc
@@ -0,0 +1,40 @@
+/*
+ *  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 {
+
+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

[tlegrand-webrtc, 2016/06/28 13:48:05]

Fault tests?

[aluebs-webrtc, 2016/06/29 00:48:40]

I don't think there is a way to catch DCHECKs, right?

[hlundin-webrtc, 2016/06/29 08:21:16]

Yes, there is. You can use death tests. See
https://github.com/google/googletest/blob/master/googletest/docs/AdvancedGuid...,
and grep our code base for ASSERT_DEATH or EXPECT_DEATH.

[aluebs-webrtc, 2016/06/29 23:02:58]

Thank you for pointing that out. Now I added death tests.

+
+TEST(AudioBufferTest, SetNumChannelsSetsChannelBuffersNumChannels) {
+  const size_t kNumFrames = 480u;
+  const size_t kStereo = 2u;
+  const size_t kMono = 1u;
+  AudioBuffer ab(kNumFrames, kStereo, kNumFrames, kStereo, kNumFrames);
+  ExpectNumChannels(ab, kStereo);
+  ab.set_num_channels(kMono);
+  ExpectNumChannels(ab, kMono);
+  ab.InitForNewData();
+  ExpectNumChannels(ab, kStereo);
+}
+
+}  // namespace webrtc