Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_coding/neteq/expand_unittest.cc

@@ -0,0 +1,206 @@
+/*
+ *  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 Expand class.
+
+#include "webrtc/modules/audio_coding/neteq/expand.h"
+
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_coding/neteq/background_noise.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"
+#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "webrtc/rtc_base/safe_conversions.h"
+#include "webrtc/test/gtest.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+TEST(Expand, CreateAndDestroy) {
+  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);
+}
+
+TEST(Expand, CreateUsingFactory) {
+  int fs = 8000;
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(1, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  ExpandFactory expand_factory;
+  Expand* expand = expand_factory.Create(&bgn, &sync_buffer, &random_vector,
+                                         &statistics, fs, channels);
+  EXPECT_TRUE(expand != NULL);
+  delete expand;
+}
+
+namespace {
+class FakeStatisticsCalculator : public StatisticsCalculator {
+ public:
+  void LogDelayedPacketOutageEvent(int outage_duration_ms) override {
+    last_outage_duration_ms_ = outage_duration_ms;
+  }
+
+  int last_outage_duration_ms() const { return last_outage_duration_ms_; }
+
+ private:
+  int last_outage_duration_ms_ = 0;
+};
+
+// This is the same size that is given to the SyncBuffer object in NetEq.
+const size_t kNetEqSyncBufferLengthMs = 720;
+}  // namespace
+
+class ExpandTest : public ::testing::Test {
+ protected:
+  ExpandTest()
+      : input_file_(test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
+                    32000),
+        test_sample_rate_hz_(32000),
+        num_channels_(1),
+        background_noise_(num_channels_),
+        sync_buffer_(num_channels_,
+                     kNetEqSyncBufferLengthMs * test_sample_rate_hz_ / 1000),
+        expand_(&background_noise_,
+                &sync_buffer_,
+                &random_vector_,
+                &statistics_,
+                test_sample_rate_hz_,
+                num_channels_) {
+    WebRtcSpl_Init();
+    input_file_.set_output_rate_hz(test_sample_rate_hz_);
+  }
+
+  void SetUp() override {
+    // Fast-forward the input file until there is speech (about 1.1 second into
+    // the file).
+    const size_t speech_start_samples =
+        static_cast<size_t>(test_sample_rate_hz_ * 1.1f);
+    ASSERT_TRUE(input_file_.Seek(speech_start_samples));
+
+    // Pre-load the sync buffer with speech data.
+    std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]);
+    ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get()));
+    sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0);
+    ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
+  }
+
+  test::ResampleInputAudioFile input_file_;
+  int test_sample_rate_hz_;
+  size_t num_channels_;
+  BackgroundNoise background_noise_;
+  SyncBuffer sync_buffer_;
+  RandomVector random_vector_;
+  FakeStatisticsCalculator statistics_;
+  Expand expand_;
+};
+
+// This test calls the expand object to produce concealment data a few times,
+// and then ends by calling SetParametersForNormalAfterExpand. This simulates
+// the situation where the packet next up for decoding was just delayed, not
+// lost.
+TEST_F(ExpandTest, DelayedPacketOutage) {
+  AudioMultiVector output(num_channels_);
+  size_t sum_output_len_samples = 0;
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(0, expand_.Process(&output));
+    EXPECT_GT(output.Size(), 0u);
+    sum_output_len_samples += output.Size();
+    EXPECT_EQ(0, statistics_.last_outage_duration_ms());
+  }
+  expand_.SetParametersForNormalAfterExpand();
+  // Convert |sum_output_len_samples| to milliseconds.
+  EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples /
+                                   (test_sample_rate_hz_ / 1000)),
+            statistics_.last_outage_duration_ms());
+}
+
+// This test is similar to DelayedPacketOutage, but ends by calling
+// SetParametersForMergeAfterExpand. This simulates the situation where the
+// packet next up for decoding was actually lost (or at least a later packet
+// arrived before it).
+TEST_F(ExpandTest, LostPacketOutage) {
+  AudioMultiVector output(num_channels_);
+  size_t sum_output_len_samples = 0;
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(0, expand_.Process(&output));
+    EXPECT_GT(output.Size(), 0u);
+    sum_output_len_samples += output.Size();
+    EXPECT_EQ(0, statistics_.last_outage_duration_ms());
+  }
+  expand_.SetParametersForMergeAfterExpand();
+  EXPECT_EQ(0, statistics_.last_outage_duration_ms());
+}
+
+// This test is similar to the DelayedPacketOutage test above, but with the
+// difference that Expand::Reset() is called after 5 calls to Expand::Process().
+// This should reset the statistics, and will in the end lead to an outage of
+// 5 periods instead of 10.
+TEST_F(ExpandTest, CheckOutageStatsAfterReset) {
+  AudioMultiVector output(num_channels_);
+  size_t sum_output_len_samples = 0;
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(0, expand_.Process(&output));
+    EXPECT_GT(output.Size(), 0u);
+    sum_output_len_samples += output.Size();
+    if (i == 5) {
+      expand_.Reset();
+      sum_output_len_samples = 0;
+    }
+    EXPECT_EQ(0, statistics_.last_outage_duration_ms());
+  }
+  expand_.SetParametersForNormalAfterExpand();
+  // Convert |sum_output_len_samples| to milliseconds.
+  EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples /
+                                   (test_sample_rate_hz_ / 1000)),
+            statistics_.last_outage_duration_ms());
+}
+
+namespace {
+// Runs expand until Muted() returns true. Times out after 1000 calls.
+void ExpandUntilMuted(size_t num_channels, Expand* expand) {
+  EXPECT_FALSE(expand->Muted()) << "Instance is muted from the start";
+  AudioMultiVector output(num_channels);
+  int num_calls = 0;
+  while (!expand->Muted()) {
+    ASSERT_LT(num_calls++, 1000) << "Test timed out";
+    EXPECT_EQ(0, expand->Process(&output));
+  }
+}
+}  // namespace
+
+// Verifies that Muted() returns true after a long expand period. Also verifies
+// that Muted() is reset to false after calling Reset(),
+// SetParametersForMergeAfterExpand() and SetParametersForNormalAfterExpand().
+TEST_F(ExpandTest, Muted) {
+  ExpandUntilMuted(num_channels_, &expand_);
+  expand_.Reset();
+  EXPECT_FALSE(expand_.Muted());  // Should be back to unmuted.
+
+  ExpandUntilMuted(num_channels_, &expand_);
+  expand_.SetParametersForMergeAfterExpand();
+  EXPECT_FALSE(expand_.Muted());  // Should be back to unmuted.
+
+  expand_.Reset();  // Must reset in order to start a new expand period.
+  ExpandUntilMuted(num_channels_, &expand_);
+  expand_.SetParametersForNormalAfterExpand();
+  EXPECT_FALSE(expand_.Muted());  // Should be back to unmuted.
+}
+
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc