Adding 120 ms frame length support in NetEq.

webrtc/modules/audio_coding/neteq/normal_unittest.cc

 /*
  *  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 <memory>
 #include <vector>
 
 #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/statistics_calculator.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 
 using ::testing::_;
+using ::testing::Invoke;
 
 namespace webrtc {
 
+namespace {
+
+int ExpandProcess120ms(AudioMultiVector* output) {
+  AudioMultiVector dummy_audio(1, 11520u);
+  dummy_audio.CopyTo(output);
+  return 0;
+}
+
+} // namespace
+
 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);
@@ skipping 71 matching lines @@
   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.
 }
 
+TEST(Normal, LastModeExpand120msPacket) {
+  WebRtcSpl_Init();
+  MockDecoderDatabase db;
+  const int kFs = 48000;
+  const size_t kPacketsizeBytes = 11520u;
+  const size_t kChannels = 1;
+  BackgroundNoise bgn(kChannels);
+  SyncBuffer sync_buffer(kChannels, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, kFs,
+                    kChannels);
+  Normal normal(kFs, &db, bgn, &expand);
+
+  int16_t input[kPacketsizeBytes] = {0};
+
+  std::unique_ptr<int16_t[]> mute_factor_array(new int16_t[kChannels]);
+  for (size_t i = 0; i < kChannels; ++i) {

[minyue-webrtc, 2016/05/02 10:44:53]

this is the second fix

+    mute_factor_array[i] = 16384;
+  }
+
+  AudioMultiVector output(kChannels);
+
+  EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
+  EXPECT_CALL(expand, Process(_)).WillOnce(Invoke(ExpandProcess120ms));
+  EXPECT_CALL(expand, Reset());
+  EXPECT_EQ(static_cast<int>(kPacketsizeBytes),
+            normal.Process(input,
+                           kPacketsizeBytes,
+                           kModeExpand,
+                           mute_factor_array.get(),
+                           &output));
+
+  EXPECT_EQ(kPacketsizeBytes, 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