Work-in-progress upload to add worker task queue to PC factory.

webrtc/media/engine/webrtcvoiceengine_unittest.cc

 /*
  *  Copyright (c) 2008 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 116 matching lines @@
   EXPECT_CALL(apm, Initialize()).WillOnce(Return(0));
   EXPECT_CALL(apm, DetachAecDump());
   StrictMock<MockTransmitMixer> transmit_mixer;
   EXPECT_CALL(transmit_mixer, EnableStereoChannelSwapping(false));
   cricket::FakeWebRtcVoiceEngine voe(&apm, &transmit_mixer);
   EXPECT_FALSE(voe.IsInited());
   {
     cricket::WebRtcVoiceEngine engine(
         &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
         webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr,
-        new FakeVoEWrapper(&voe));
+        nullptr, new FakeVoEWrapper(&voe));
     EXPECT_TRUE(voe.IsInited());
   }
   EXPECT_FALSE(voe.IsInited());
 }
 
 class FakeAudioSink : public webrtc::AudioSinkInterface {
  public:
   void OnData(const Data& audio) override {}
 };
 
@@ skipping 29 matching lines @@
     EXPECT_CALL(apm_gc_, compression_gain_db()).WillRepeatedly(Return(5));
     EXPECT_CALL(apm_gc_, is_limiter_enabled()).WillRepeatedly(Return(true));
     EXPECT_CALL(apm_gc_, set_target_level_dbfs(1)).WillOnce(Return(0));
     EXPECT_CALL(apm_gc_, set_compression_gain_db(5)).WillRepeatedly(Return(0));
     EXPECT_CALL(apm_gc_, enable_limiter(true)).WillRepeatedly(Return(0));
     // TODO(kwiberg): We should use mock factories here, but a bunch of
     // the tests here probe the specific set of codecs provided by the builtin
     // factories. Those tests should probably be moved elsewhere.
     auto encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
     auto decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
-    engine_.reset(new cricket::WebRtcVoiceEngine(&adm_, encoder_factory,
-                                                 decoder_factory, nullptr,
-                                                 new FakeVoEWrapper(&voe_)));
+    engine_.reset(new cricket::WebRtcVoiceEngine(
+        &adm_, encoder_factory, decoder_factory, nullptr, nullptr,
+        new FakeVoEWrapper(&voe_)));
     send_parameters_.codecs.push_back(kPcmuCodec);
     recv_parameters_.codecs.push_back(kPcmuCodec);
     // Default Options.
     EXPECT_TRUE(IsHighPassFilterEnabled());
   }
 
   bool SetupChannel() {
     EXPECT_CALL(apm_, ApplyConfig(testing::_));
     EXPECT_CALL(apm_, SetExtraOptions(testing::_));
     channel_ = engine_->CreateChannel(&call_, cricket::MediaConfig(),
@@ skipping 3055 matching lines @@
 
   EXPECT_TRUE(GetRecvStream(kSsrcX).started());
 }
 
 // Tests that the library initializes and shuts down properly.
 TEST(WebRtcVoiceEngineTest, StartupShutdown) {
   // If the VoiceEngine wants to gather available codecs early, that's fine but
   // we never want it to create a decoder at this stage.
   cricket::WebRtcVoiceEngine engine(
       nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
-      webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr);
+      webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, nullptr);
   webrtc::RtcEventLogNullImpl event_log;
   std::unique_ptr<webrtc::Call> call(
       webrtc::Call::Create(webrtc::Call::Config(&event_log)));
   cricket::VoiceMediaChannel* channel = engine.CreateChannel(
       call.get(), cricket::MediaConfig(), cricket::AudioOptions());
   EXPECT_TRUE(channel != nullptr);
   delete channel;
 }
 
 // Tests that reference counting on the external ADM is correct.
 TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
   testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm;
   EXPECT_CALL(adm, AddRef()).Times(3).WillRepeatedly(Return(0));
   EXPECT_CALL(adm, Release()).Times(3).WillRepeatedly(Return(0));
   // Return 100ms just in case this function gets called.  If we don't,
   // we could enter a tight loop since the mock would return 0.
   EXPECT_CALL(adm, TimeUntilNextProcess()).WillRepeatedly(Return(100));
   {
     cricket::WebRtcVoiceEngine engine(
         &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
-        webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr);
+        webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr,
+        nullptr);
     webrtc::RtcEventLogNullImpl event_log;
     std::unique_ptr<webrtc::Call> call(
         webrtc::Call::Create(webrtc::Call::Config(&event_log)));
     cricket::VoiceMediaChannel* channel = engine.CreateChannel(
         call.get(), cricket::MediaConfig(), cricket::AudioOptions());
     EXPECT_TRUE(channel != nullptr);
     delete channel;
   }
 }
 
 // Verify the payload id of common audio codecs, including CN, ISAC, and G722.
 TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
   // TODO(ossu): Why are the payload types of codecs with non-static payload
   // type assignments checked here? It shouldn't really matter.
   cricket::WebRtcVoiceEngine engine(
       nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
-      webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr);
+      webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, nullptr);
   for (const cricket::AudioCodec& codec : engine.send_codecs()) {
     auto is_codec = [&codec](const char* name, int clockrate = 0) {
       return STR_CASE_CMP(codec.name.c_str(), name) == 0 &&
              (clockrate == 0 || codec.clockrate == clockrate);
     };
     if (is_codec("CN", 16000)) {
       EXPECT_EQ(105, codec.id);
     } else if (is_codec("CN", 32000)) {
       EXPECT_EQ(106, codec.id);
     } else if (is_codec("ISAC", 16000)) {
@@ skipping 20 matching lines @@
       ASSERT_TRUE(codec.params.find("useinbandfec") != codec.params.end());
       EXPECT_EQ("1", codec.params.find("useinbandfec")->second);
     }
   }
 }
 
 // Tests that VoE supports at least 32 channels
 TEST(WebRtcVoiceEngineTest, Has32Channels) {
   cricket::WebRtcVoiceEngine engine(
       nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
-      webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr);
+      webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, nullptr);
   webrtc::RtcEventLogNullImpl event_log;
   std::unique_ptr<webrtc::Call> call(
       webrtc::Call::Create(webrtc::Call::Config(&event_log)));
 
   cricket::VoiceMediaChannel* channels[32];
   int num_channels = 0;
   while (num_channels < arraysize(channels)) {
     cricket::VoiceMediaChannel* channel = engine.CreateChannel(
         call.get(), cricket::MediaConfig(), cricket::AudioOptions());
     if (!channel)
@@ skipping 13 matching lines @@
 TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
   // TODO(ossu): I'm not sure of the intent of this test. It's either:
   // - Check that our builtin codecs are usable by Channel.
   // - The codecs provided by the engine is usable by Channel.
   // It does not check that the codecs in the RecvParameters are actually
   // what we sent in - though it's probably reasonable to expect so, if
   // SetRecvParameters returns true.
   // I think it will become clear once audio decoder injection is completed.
   cricket::WebRtcVoiceEngine engine(
       nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
-      webrtc::CreateBuiltinAudioDecoderFactory(), nullptr);
+      webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, nullptr);
   webrtc::RtcEventLogNullImpl event_log;
   std::unique_ptr<webrtc::Call> call(
       webrtc::Call::Create(webrtc::Call::Config(&event_log)));
   cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),
                                            cricket::AudioOptions(), call.get());
   cricket::AudioRecvParameters parameters;
   parameters.codecs = engine.recv_codecs();
   EXPECT_TRUE(channel.SetRecvParameters(parameters));
 }
 
@@ skipping 16 matching lines @@
       webrtc::AudioCodecSpec{{"codec5", 8000, 2}, {8000, 1, 64000}});
 
   rtc::scoped_refptr<webrtc::MockAudioEncoderFactory> unused_encoder_factory =
       webrtc::MockAudioEncoderFactory::CreateUnusedFactory();
   rtc::scoped_refptr<webrtc::MockAudioDecoderFactory> mock_decoder_factory =
       new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>;
   EXPECT_CALL(*mock_decoder_factory.get(), GetSupportedDecoders())
       .WillOnce(Return(specs));
 
   cricket::WebRtcVoiceEngine engine(nullptr, unused_encoder_factory,
-                                    mock_decoder_factory, nullptr);
+                                    mock_decoder_factory, nullptr, nullptr);
   auto codecs = engine.recv_codecs();
   EXPECT_EQ(11, codecs.size());
 
   // Rather than just ASSERTing that there are enough codecs, ensure that we can
   // check the actual values safely, to provide better test results.
   auto get_codec =
       [&codecs](size_t index) -> const cricket::AudioCodec& {
         static const cricket::AudioCodec missing_codec(0, "<missing>", 0, 0, 0);
         if (codecs.size() > index)
           return codecs[index];
@@ skipping 28 matching lines @@
   // Without this cast, the comparison turned unsigned and, thus, failed for -1.
   const int num_specs = static_cast<int>(specs.size());
   EXPECT_GE(find_codec({"cn", 8000, 1}), num_specs);
   EXPECT_GE(find_codec({"cn", 16000, 1}), num_specs);
   EXPECT_EQ(find_codec({"cn", 32000, 1}), -1);
   EXPECT_GE(find_codec({"telephone-event", 8000, 1}), num_specs);
   EXPECT_GE(find_codec({"telephone-event", 16000, 1}), num_specs);
   EXPECT_GE(find_codec({"telephone-event", 32000, 1}), num_specs);
   EXPECT_GE(find_codec({"telephone-event", 48000, 1}), num_specs);
 }