Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 126 insertions, 0 deletions

diff --git a/webrtc/modules/audio_coding/neteq/normal_unittest.cc b/webrtc/modules/audio_coding/neteq/normal_unittest.cc
new file mode 100644
index 0000000000..1ac32f46a7
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/normal_unittest.cc
@@ -0,0 +1,126 @@
+/*
+ *  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.
+ */
+
+// Unit tests for Normal class.
+
+#include "webrtc/modules/audio_coding/neteq/normal.h"
+
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.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/statistics_calculator.h"
+#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
+
+using ::testing::_;
+
+namespace webrtc {
+
+TEST(Normal, CreateAndDestroy) {
+  MockDecoderDatabase db;
+  int fs = 8000;
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(1, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
+  Normal normal(fs, &db, bgn, &expand);
+  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;
+  StatisticsCalculator statistics;
+  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
+                    channels);
+  Normal normal(fs, &db, bgn, &expand);
+
+  int16_t input[1000] = {0};
+  rtc::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;
+  StatisticsCalculator statistics;
+  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
+                    channels);
+  Normal normal(fs, &db, bgn, &expand);
+
+  int16_t input[1000] = {0};
+  rtc::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