Upstream version 10.39.225.0

[platform/framework/web/crosswalk.git] / src / third_party / webrtc / modules / audio_coding / neteq / normal_unittest.cc

diff --git a/src/third_party/webrtc/modules/audio_coding/neteq/normal_unittest.cc b/src/third_party/webrtc/modules/audio_coding/neteq/normal_unittest.cc
index c855865..e96359a 100644 (file)
--- a/src/third_party/webrtc/modules/audio_coding/neteq/normal_unittest.cc
+++ b/src/third_party/webrtc/modules/audio_coding/neteq/normal_unittest.cc
@@ -14,12 +14,18 @@
 
 #include <vector>
 
-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "webrtc/modules/audio_coding/neteq/mock/mock_expand.h"
 #include "webrtc/modules/audio_coding/neteq/random_vector.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+using ::testing::_;
 
 namespace webrtc {
 
@@ -35,6 +41,80 @@ TEST(Normal, CreateAndDestroy) {
   EXPECT_CALL(db, Die());  // Called when |db| goes out of scope.
 }
 
+TEST(Normal, AvoidDivideByZero) {
+  WebRtcSpl_Init();
+  MockDecoderDatabase db;
+  int fs = 8000;
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(1, 1000);
+  RandomVector random_vector;
+  MockExpand expand(&bgn, &sync_buffer, &random_vector, fs, channels);
+  Normal normal(fs, &db, bgn, &expand);
+
+  int16_t input[1000] = {0};
+  scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
+  for (size_t i = 0; i < channels; ++i) {
+    mute_factor_array[i] = 16384;
+  }
+  AudioMultiVector output(channels);
+
+  // Zero input length.
+  EXPECT_EQ(
+      0,
+      normal.Process(input, 0, kModeExpand, mute_factor_array.get(), &output));
+  EXPECT_EQ(0u, output.Size());
+
+  // Try to make energy_length >> scaling = 0;
+  EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
+  EXPECT_CALL(expand, Process(_));
+  EXPECT_CALL(expand, Reset());
+  // If input_size_samples < 64, then energy_length in Normal::Process() will
+  // be equal to input_size_samples. Since the input is all zeros, decoded_max
+  // will be zero, and scaling will be >= 6. Thus, energy_length >> scaling = 0,
+  // and using this as a denominator would lead to problems.
+  int input_size_samples = 63;
+  EXPECT_EQ(input_size_samples,
+            normal.Process(input,
+                           input_size_samples,
+                           kModeExpand,
+                           mute_factor_array.get(),
+                           &output));
+
+  EXPECT_CALL(db, Die());      // Called when |db| goes out of scope.
+  EXPECT_CALL(expand, Die());  // Called when |expand| goes out of scope.
+}
+
+TEST(Normal, InputLengthAndChannelsDoNotMatch) {
+  WebRtcSpl_Init();
+  MockDecoderDatabase db;
+  int fs = 8000;
+  size_t channels = 2;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(channels, 1000);
+  RandomVector random_vector;
+  MockExpand expand(&bgn, &sync_buffer, &random_vector, fs, channels);
+  Normal normal(fs, &db, bgn, &expand);
+
+  int16_t input[1000] = {0};
+  scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
+  for (size_t i = 0; i < channels; ++i) {
+    mute_factor_array[i] = 16384;
+  }
+  AudioMultiVector output(channels);
+
+  // Let the number of samples be one sample less than 80 samples per channel.
+  size_t input_len = 80 * channels - 1;
+  EXPECT_EQ(
+      0,
+      normal.Process(
+          input, input_len, kModeExpand, mute_factor_array.get(), &output));
+  EXPECT_EQ(0u, output.Size());
+
+  EXPECT_CALL(db, Die());      // Called when |db| goes out of scope.
+  EXPECT_CALL(expand, Die());  // Called when |expand| goes out of scope.
+}
+
 // TODO(hlundin): Write more tests.
 
 }  // namespace webrtc