Index: webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc |
diff --git a/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc |
deleted file mode 100644 |
index 09c569966b7f8789e7659df185e4aa561bdf1980..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc |
+++ /dev/null |
@@ -1,82 +0,0 @@ |
-/* |
- * Copyright (c) 2014 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/modules/audio_coding/neteq/audio_classifier.h" |
- |
-#include <math.h> |
-#include <stdio.h> |
-#include <stdlib.h> |
-#include <string.h> |
-#include <memory> |
-#include <string> |
- |
-#include "webrtc/test/gtest.h" |
-#include "webrtc/test/testsupport/fileutils.h" |
- |
-namespace webrtc { |
- |
-static const size_t kFrameSize = 960; |
- |
-TEST(AudioClassifierTest, AllZeroInput) { |
- int16_t in_mono[kFrameSize] = {0}; |
- |
- // Test all-zero vectors and let the classifier converge from its default |
- // to the expected value. |
- AudioClassifier zero_classifier; |
- for (int i = 0; i < 100; ++i) { |
- zero_classifier.Analysis(in_mono, kFrameSize, 1); |
- } |
- EXPECT_TRUE(zero_classifier.is_music()); |
-} |
- |
-void RunAnalysisTest(const std::string& audio_filename, |
- const std::string& data_filename, |
- size_t channels) { |
- AudioClassifier classifier; |
- std::unique_ptr<int16_t[]> in(new int16_t[channels * kFrameSize]); |
- bool is_music_ref; |
- |
- FILE* audio_file = fopen(audio_filename.c_str(), "rb"); |
- ASSERT_TRUE(audio_file != NULL) << "Failed to open file " << audio_filename |
- << std::endl; |
- FILE* data_file = fopen(data_filename.c_str(), "rb"); |
- ASSERT_TRUE(audio_file != NULL) << "Failed to open file " << audio_filename |
- << std::endl; |
- while (fread(in.get(), sizeof(int16_t), channels * kFrameSize, audio_file) == |
- channels * kFrameSize) { |
- bool is_music = |
- classifier.Analysis(in.get(), channels * kFrameSize, channels); |
- EXPECT_EQ(is_music, classifier.is_music()); |
- ASSERT_EQ(1u, fread(&is_music_ref, sizeof(is_music_ref), 1, data_file)); |
- EXPECT_EQ(is_music_ref, is_music); |
- } |
- fclose(audio_file); |
- fclose(data_file); |
-} |
- |
-TEST(AudioClassifierTest, DoAnalysisMono) { |
-#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) |
- RunAnalysisTest(test::ResourcePath("short_mixed_mono_48", "pcm"), |
- test::ResourcePath("short_mixed_mono_48_arm", "dat"), |
- 1); |
-#else |
- RunAnalysisTest(test::ResourcePath("short_mixed_mono_48", "pcm"), |
- test::ResourcePath("short_mixed_mono_48", "dat"), |
- 1); |
-#endif // WEBRTC_ARCH_ARM |
-} |
- |
-TEST(AudioClassifierTest, DoAnalysisStereo) { |
- RunAnalysisTest(test::ResourcePath("short_mixed_stereo_48", "pcm"), |
- test::ResourcePath("short_mixed_stereo_48", "dat"), |
- 2); |
-} |
- |
-} // namespace webrtc |