1 files changed, 63 insertions, 0 deletions
diff --git a/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc b/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
new file mode 100644
index 0000000000..f509af476f
--- /dev/null
+++ b/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
@@ -0,0 +1,63 @@
+/*
+ *  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 (size_t n = 0; n < features.num_frames; n++)
+        EXPECT_NEAR(features.spectral_peak[n], sp[n], 3);
+    }
+  }
+
+  fclose(peak_file);
+  fclose(pcm_file);
+}
+
+}  // namespace webrtc