Delete VAD methods from AcmReceiver and move functionality inside NetEq

webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
@@ skipping 40 matching lines @@
 }  // namespace
 
 class AcmReceiverTestOldApi : public AudioPacketizationCallback,
                               public ::testing::Test {
  protected:
   AcmReceiverTestOldApi()
       : timestamp_(0),
         packet_sent_(false),
         last_packet_send_timestamp_(timestamp_),
         last_frame_type_(kEmptyFrame) {
-    AudioCodingModule::Config config;
-    acm_.reset(new AudioCodingModuleImpl(config));
-    receiver_.reset(new AcmReceiver(config));
   }
 
   ~AcmReceiverTestOldApi() {}
 
   void SetUp() override {
+    acm_.reset(new AudioCodingModuleImpl(config_));
+    receiver_.reset(new AcmReceiver(config_));
     ASSERT_TRUE(receiver_.get() != NULL);
     ASSERT_TRUE(acm_.get() != NULL);
     codecs_ = RentACodec::Database();
 
     acm_->InitializeReceiver();
     acm_->RegisterTransportCallback(this);
 
     rtp_header_.header.sequenceNumber = 0;
     rtp_header_.header.timestamp = 0;
     rtp_header_.header.markerBit = false;
@@ skipping 67 matching lines @@
     if (ret_val < 0) {
       assert(false);
       return -1;
     }
     rtp_header_.header.sequenceNumber++;
     packet_sent_ = true;
     last_frame_type_ = frame_type;
     return 0;
   }
 
+  AudioCodingModule::Config config_;
   std::unique_ptr<AcmReceiver> receiver_;
   rtc::ArrayView<const CodecInst> codecs_;
   std::unique_ptr<AudioCodingModule> acm_;
   WebRtcRTPHeader rtp_header_;
   uint32_t timestamp_;
   bool packet_sent_;  // Set when SendData is called reset when inserting audio.
   uint32_t last_packet_send_timestamp_;
   FrameType last_frame_type_;
 };
 
@@ skipping 122 matching lines @@
     EXPECT_EQ(codec.inst.plfreq, receiver_->last_output_sample_rate_hz());
   }
 }
 
 #if defined(WEBRTC_ANDROID)
 #define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
 #else
 #define MAYBE_PostdecodingVad PostdecodingVad
 #endif
 TEST_F(AcmReceiverTestOldApi, MAYBE_PostdecodingVad) {
-  receiver_->EnableVad();
-  EXPECT_TRUE(receiver_->vad_enabled());
+  EXPECT_TRUE(config_.neteq_config.enable_post_decode_vad);
   const CodecIdInst codec(RentACodec::CodecId::kPCM16Bwb);
   ASSERT_EQ(
       0, receiver_->AddCodec(codec.id, codec.inst.pltype, codec.inst.channels,
                              codec.inst.plfreq, nullptr, ""));
   const int kNumPackets = 5;
   const int num_10ms_frames = codec.inst.pacsize / (codec.inst.plfreq / 100);
   AudioFrame frame;
   for (int n = 0; n < kNumPackets; ++n) {
     InsertOnePacketOfSilence(codec.id);
     for (int k = 0; k < num_10ms_frames; ++k)
       ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &frame));
   }
   EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_);
+}
 
-  receiver_->DisableVad();
-  EXPECT_FALSE(receiver_->vad_enabled());
+class AcmReceiverTestPostDecodeVadPassiveOldApi : public AcmReceiverTestOldApi {
+ protected:
+  AcmReceiverTestPostDecodeVadPassiveOldApi() {
+    config_.neteq_config.enable_post_decode_vad = false;
+  }
+};
 
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
+#else
+#define MAYBE_PostdecodingVad PostdecodingVad
+#endif
+TEST_F(AcmReceiverTestPostDecodeVadPassiveOldApi, MAYBE_PostdecodingVad) {
+  EXPECT_FALSE(config_.neteq_config.enable_post_decode_vad);
+  const CodecIdInst codec(RentACodec::CodecId::kPCM16Bwb);
+  ASSERT_EQ(
+      0, receiver_->AddCodec(codec.id, codec.inst.pltype, codec.inst.channels,
+                             codec.inst.plfreq, nullptr, ""));
+  const int kNumPackets = 5;
+  const int num_10ms_frames = codec.inst.pacsize / (codec.inst.plfreq / 100);
+  AudioFrame frame;
   for (int n = 0; n < kNumPackets; ++n) {
     InsertOnePacketOfSilence(codec.id);
     for (int k = 0; k < num_10ms_frames; ++k)
       ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &frame));
   }
   EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
 }
 
 #if defined(WEBRTC_ANDROID)
 #define MAYBE_LastAudioCodec DISABLED_LastAudioCodec
@@ skipping 62 matching lines @@
               receiver_->last_packet_sample_rate_hz());
     EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
     EXPECT_TRUE(CodecsEqual(c.inst, codec));
   }
 }
 #endif
 
 }  // namespace acm2
 
 }  // namespace webrtc