Add a unit testing framework.

Populate it with the beginnings of a resampler unit test to have it do someting.

Also fix a bug in resampler caught with the test ;)
Review URL: http://webrtc-codereview.appspot.com/135019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@595 4adac7df-926f-26a2-2b94-8c16560cd09d

src/common_audio/common_audio.gyp

@@ -8,9 +8,28 @@
 
 {
   'includes': [
-    '../common_settings.gypi', # Common settings
+    '../build/common.gypi',
     'signal_processing_library/main/source/spl.gypi',
     'resampler/main/source/resampler.gypi',
     'vad/main/source/vad.gypi',
   ],
+  'conditions': [
+    ['build_with_chromium==0', {
+      'targets' : [
+        {
+          'target_name': 'common_audio_unittests',
+          'type': 'executable',
+          'dependencies': [
+            '<(webrtc_root)/../test/test.gyp:test_support',
+            '<(webrtc_root)/../testing/gtest.gyp:gtest',
+            'resampler',
+          ],
+          'sources': [
+            '<(webrtc_root)/../test/run_all_unittests.cc',
+            'resampler/main/source/resampler_unittest.cc',
+          ],
+        },
+      ],
+    }],
+  ],
 }

src/common_audio/resampler/main/interface/resampler.h

@@ -22,7 +22,7 @@ namespace webrtc
 {
 
 // TODO(andrew): the implementation depends on the exact values of this enum.
-//   It should be rewritten in a less fragile way.
+// It should be rewritten in a less fragile way.
 enum ResamplerType
 {
     // 4 MSB = Number of channels
@@ -35,6 +35,7 @@ enum ResamplerType
     kResamplerInvalid = 0xff
 };
 
+// TODO(andrew): doesn't need to be part of the interface.
 enum ResamplerMode
 {
     kResamplerMode1To1,
@@ -63,6 +64,7 @@ class Resampler
 
 public:
     Resampler();
+    // TODO(andrew): use an init function instead.
     Resampler(int inFreq, int outFreq, ResamplerType type);
     ~Resampler();
 

src/common_audio/resampler/main/source/resampler.cc

@@ -62,8 +62,6 @@ Resampler::Resampler(int inFreq, int outFreq, ResamplerType type)
     slave_left_ = NULL;
     slave_right_ = NULL;
 
-    // TODO(andrew): looks like this class should use an init method
-    //   (and possibly a static create).
     Reset(inFreq, outFreq, type);
 }
 
@@ -213,7 +211,7 @@ int Resampler::Reset(int inFreq, int outFreq, ResamplerType type)
                 break;
             default:
                 my_type_ = kResamplerInvalid;
-                break;
+                return -1;
         }
     } else if (outFreq == 1)
     {
@@ -233,7 +231,7 @@ int Resampler::Reset(int inFreq, int outFreq, ResamplerType type)
                 break;
             default:
                 my_type_ = kResamplerInvalid;
-                break;
+                return -1;
         }
     } else if ((inFreq == 2) && (outFreq == 3))
     {

src/common_audio/resampler/main/source/resampler_unittest.cc

@@ -0,0 +1,119 @@
+/*
+ *  Copyright (c) 2011 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 "gtest/gtest.h"
+
+#include "common_audio/resampler/main/interface/resampler.h"
+
+// TODO(andrew): this is a work-in-progress. Many more tests are needed.
+
+namespace webrtc {
+namespace {
+const ResamplerType kTypes[] = {
+  kResamplerSynchronous,
+  kResamplerAsynchronous,
+  kResamplerSynchronousStereo,
+  kResamplerAsynchronousStereo
+  // kResamplerInvalid excluded
+};
+const size_t kTypesSize = sizeof(kTypes) / sizeof(*kTypes);
+
+// Rates we must support.
+const int kMaxRate = 96000;
+const int kRates[] = {
+  8000,
+  16000,
+  32000,
+  44000,
+  48000,
+  kMaxRate
+};
+const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
+const size_t kDataSize = kMaxRate / 100;
+
+// TODO(andrew): should we be supporting these combinations?
+bool ValidRates(int in_rate, int out_rate) {
+  // Not the most compact notation, for clarity.
+  if ((in_rate == 44000 && (out_rate == 48000 || out_rate == 96000)) ||
+      (out_rate == 44000 && (in_rate == 48000 || in_rate == 96000))  ||
+      (in_rate == 8000 && out_rate == 96000) ||
+      (in_rate == 96000 && out_rate == 8000)) {
+    return false;
+  }
+
+  return true;
+}
+
+class ResamplerTest : public testing::Test {
+ protected:
+  ResamplerTest();
+  virtual void SetUp();
+  virtual void TearDown();
+
+  Resampler rs_;
+  int16_t data_in_[kDataSize];
+  int16_t data_out_[kDataSize];
+};
+
+ResamplerTest::ResamplerTest() {
+}
+
+void ResamplerTest::SetUp() {
+}
+
+void ResamplerTest::TearDown() {
+}
+
+TEST_F(ResamplerTest, Reset) {
+  // The only failure mode for the constructor is if Reset() fails. For the
+  // time being then (until an Init function is added), we rely on Reset()
+  // to test the constructor.
+
+  // Check that all required combinations are supported.
+  for (size_t i = 0; i < kRatesSize; ++i) {
+    for (size_t j = 0; j < kRatesSize; ++j) {
+      for (size_t k = 0; k < kTypesSize; ++k) {
+        std::ostringstream ss;
+        ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j]
+            << ", type: " << kTypes[k];
+        SCOPED_TRACE(ss.str());
+        if (ValidRates(kRates[i], kRates[j]))
+          EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kTypes[k]));
+        else
+          EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kTypes[k]));
+      }
+    }
+  }
+}
+
+TEST_F(ResamplerTest, Synchronous) {
+  for (size_t i = 0; i < kRatesSize; ++i) {
+    for (size_t j = 0; j < kRatesSize; ++j) {
+      std::ostringstream ss;
+      ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
+      SCOPED_TRACE(ss.str());
+
+      if (ValidRates(kRates[i], kRates[j])) {
+        int in_length = kRates[i] / 100;
+        int out_length = 0;
+        EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kResamplerSynchronous));
+        EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
+                              out_length));
+        EXPECT_EQ(kRates[j] / 100, out_length);
+      } else {
+        EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kResamplerSynchronous));
+      }
+    }
+  }
+
+  // TODO(andrew): test stereo.
+}
+}  // namespace
+}  // namespace webrtc

test/run_all_unittests.cc

@@ -0,0 +1,16 @@
+/*
+ *  Copyright (c) 2011 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 "test/test_suite.h"
+
+int main(int argc, char** argv) {
+  webrtc::TestSuite test_suite(argc, argv);
+  return test_suite.Run();
+}

test/test.gyp

@@ -0,0 +1,27 @@
+# Copyright (c) 2011 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.
+
+# TODO(andrew): consider moving test_support to src/base/test.
+{
+  'includes': [
+    '../src/build/common.gypi',
+  ],
+  'targets': [
+    {
+      'target_name': 'test_support',
+      'type': 'static_library',
+      'dependencies': [
+        '../testing/gtest.gyp:gtest',
+      ],
+      'sources': [
+        'test_suite.cc',
+        'test_suite.h',
+      ],
+    },
+  ],
+}

test/test_suite.cc

@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 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 "test/test_suite.h"
+
+#include "gtest/gtest.h"
+
+namespace webrtc {
+TestSuite::TestSuite(int argc, char** argv) {
+  testing::InitGoogleTest(&argc, argv);
+}
+
+TestSuite::~TestSuite() {
+}
+
+int TestSuite::Run() {
+  Initialize();
+  int result = RUN_ALL_TESTS();
+  Shutdown();
+  return result;
+}
+
+void TestSuite::Initialize() {
+  // TODO(andrew): initialize singletons here (e.g. Trace).
+}
+
+void TestSuite::Shutdown() {
+}
+}  // namespace webrtc

test/test_suite.h

@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 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 TEST_TEST_SUITE_H_
+#define TEST_TEST_SUITE_H_
+
+// Derived from Chromium's src/base/test/test_suite.h.
+
+// Defines a basic test suite framework for running gtest based tests.  You can
+// instantiate this class in your main function and call its Run method to run
+// any gtest based tests that are linked into your executable.
+
+#include "src/system_wrappers/interface/constructor_magic.h"
+
+namespace webrtc {
+class TestSuite {
+ public:
+  TestSuite(int argc, char** argv);
+  virtual ~TestSuite();
+
+  int Run();
+
+ protected:
+  // Override these for custom initialization and shutdown handling.  Use these
+  // instead of putting complex code in your constructor/destructor.
+  virtual void Initialize();
+  virtual void Shutdown();
+
+  DISALLOW_COPY_AND_ASSIGN(TestSuite);
+};
+}  // namespace webrtc
+
+#endif  // TEST_TEST_SUITE_H_

webrtc.gyp

@@ -8,24 +8,26 @@
 
 {
   'includes': [
-    'src/common_settings.gypi', # Common settings
+    'src/build/common.gypi',
   ],
- 'targets': [
+  'targets': [
     {
-      'target_name': 'auto_tests',
+      'target_name': 'All',
       'type': 'none',
       'dependencies': [
-        'src/voice_engine/voice_engine.gyp:voe_auto_test',
+        'src/common_audio/common_audio.gyp:*',
+        # TODO(andrew): enable these when all tests build.
+        #'src/common_video/common_video.gyp:*',
+        #'src/modules/modules.gyp:*',
+        #'src/system_wrappers/source/system_wrappers.gyp:*',
+        # TODO(andrew): move the merge_lib targets to a private gyp so we can
+        # target "*" in these.
         'src/video_engine/video_engine.gyp:vie_auto_test',
-      ],
-    },
-    {
-      'target_name': 'cmd_test',
-      'type': 'none',
-      'dependencies': [
+        'src/voice_engine/voice_engine.gyp:voe_auto_test',
         'src/voice_engine/voice_engine.gyp:voe_cmd_test',
       ],
-    },    
+    },
+    # TODO(andrew): move peerconnection to its own gyp.
     {
       'target_name': 'peerconnection_server',
       'type': 'executable',
@@ -61,7 +63,7 @@
           ],
           'msvs_settings': {
             'VCLinkerTool': {
-             'SubSystem': '2',   # Windows
+             'SubSystem': '2',  # Windows
             },
           },
           'dependencies': [
@@ -116,5 +118,5 @@
         },
       ],  # targets
     }, ],  # OS="linux"
-  ],  # conditions  
+  ],  # conditions
 }