Index: webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc |
diff --git a/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc b/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc |
deleted file mode 100644 |
index 675af70b4575fbc17508375f37b8ba5e0c79db4c..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc |
+++ /dev/null |
@@ -1,63 +0,0 @@ |
-/* |
- * Copyright (c) 2012 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. |
- */ |
- |
-// We don't test the value of pitch gain and lags as they are created by iSAC |
-// routines. However, interpolation of pitch-gain and lags is in a separate |
-// class and has its own unit-test. |
- |
-#include "webrtc/modules/audio_processing/vad/vad_audio_proc.h" |
- |
-#include <math.h> |
-#include <stdio.h> |
- |
-#include <string> |
- |
-#include "testing/gtest/include/gtest/gtest.h" |
-#include "webrtc/modules/audio_processing/vad/common.h" |
-#include "webrtc/modules/interface/module_common_types.h" |
-#include "webrtc/test/testsupport/fileutils.h" |
- |
-namespace webrtc { |
- |
-TEST(AudioProcessingTest, DISABLED_ComputingFirstSpectralPeak) { |
- VadAudioProc audioproc; |
- |
- std::string peak_file_name = |
- test::ResourcePath("audio_processing/agc/agc_spectral_peak", "dat"); |
- FILE* peak_file = fopen(peak_file_name.c_str(), "rb"); |
- ASSERT_TRUE(peak_file != NULL); |
- |
- std::string pcm_file_name = |
- test::ResourcePath("audio_processing/agc/agc_audio", "pcm"); |
- FILE* pcm_file = fopen(pcm_file_name.c_str(), "rb"); |
- ASSERT_TRUE(pcm_file != NULL); |
- |
- // Read 10 ms audio in each iteration. |
- const size_t kDataLength = kLength10Ms; |
- int16_t data[kDataLength] = {0}; |
- AudioFeatures features; |
- double sp[kMaxNumFrames]; |
- while (fread(data, sizeof(int16_t), kDataLength, pcm_file) == kDataLength) { |
- audioproc.ExtractFeatures(data, kDataLength, &features); |
- if (features.num_frames > 0) { |
- ASSERT_LT(features.num_frames, kMaxNumFrames); |
- // Read reference values. |
- const size_t num_frames = features.num_frames; |
- ASSERT_EQ(num_frames, fread(sp, sizeof(sp[0]), num_frames, peak_file)); |
- for (int n = 0; n < features.num_frames; n++) |
- EXPECT_NEAR(features.spectral_peak[n], sp[n], 3); |
- } |
- } |
- |
- fclose(peak_file); |
- fclose(pcm_file); |
-} |
- |
-} // namespace webrtc |