Rename Maybe to Optional

talk/app/webrtc/webrtcsession_unittest.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 3279 matching lines @@
   Init();
   SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(channel != NULL);
   ASSERT_EQ(1u, channel->send_streams().size());
   uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
   EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
 
   cricket::AudioOptions options;
-  options.echo_cancellation = rtc::Maybe<bool>(true);
+  options.echo_cancellation = rtc::Optional<bool>(true);
 
   rtc::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
   session_->SetAudioSend(send_ssrc, false, options, renderer.get());
   EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
-  EXPECT_EQ(rtc::Maybe<bool>(), channel->options().echo_cancellation);
+  EXPECT_EQ(rtc::Optional<bool>(), channel->options().echo_cancellation);
   EXPECT_TRUE(renderer->sink() != NULL);
 
   // This will trigger SetSink(NULL) to the |renderer|.
   session_->SetAudioSend(send_ssrc, true, options, NULL);
   EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
-  EXPECT_EQ(rtc::Maybe<bool>(true), channel->options().echo_cancellation);
+  EXPECT_EQ(rtc::Optional<bool>(true), channel->options().echo_cancellation);
   EXPECT_TRUE(renderer->sink() == NULL);
 }
 
 TEST_F(WebRtcSessionTest, AudioRendererForLocalStream) {
   Init();
   SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(channel != NULL);
   ASSERT_EQ(1u, channel->send_streams().size());
@@ skipping 662 matching lines @@
 
   SetLocalDescriptionWithoutError(offer);
 
   video_channel_ = media_engine_->GetVideoChannel(0);
   voice_channel_ = media_engine_->GetVoiceChannel(0);
 
   ASSERT_TRUE(video_channel_ != NULL);
   ASSERT_TRUE(voice_channel_ != NULL);
   const cricket::AudioOptions& audio_options = voice_channel_->options();
   const cricket::VideoOptions& video_options = video_channel_->options();
-  EXPECT_EQ(rtc::Maybe<bool>(true), audio_options.dscp);
-  EXPECT_EQ(rtc::Maybe<bool>(true), video_options.dscp);
+  EXPECT_EQ(rtc::Optional<bool>(true), audio_options.dscp);
+  EXPECT_EQ(rtc::Optional<bool>(true), video_options.dscp);
 }
 
 TEST_F(WebRtcSessionTest, TestSuspendBelowMinBitrateConstraint) {
   constraints_.reset(new FakeConstraints());
   constraints_->AddOptional(
       webrtc::MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
       true);
   Init();
   SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   SetLocalDescriptionWithoutError(offer);
 
   video_channel_ = media_engine_->GetVideoChannel(0);
 
   ASSERT_TRUE(video_channel_ != NULL);
   const cricket::VideoOptions& video_options = video_channel_->options();
-  EXPECT_EQ(rtc::Maybe<bool>(true), video_options.suspend_below_min_bitrate);
+  EXPECT_EQ(rtc::Optional<bool>(true), video_options.suspend_below_min_bitrate);
 }
 
 TEST_F(WebRtcSessionTest, TestNumUnsignalledRecvStreamsConstraint) {
   // Number of unsignalled receiving streams should be between 0 and
   // kMaxUnsignalledRecvStreams.
   SetAndVerifyNumUnsignalledRecvStreams(10, 10);
   SetAndVerifyNumUnsignalledRecvStreams(kMaxUnsignalledRecvStreams + 1,
                                         kMaxUnsignalledRecvStreams);
   SetAndVerifyNumUnsignalledRecvStreams(-1, 0);
 }
 
 TEST_F(WebRtcSessionTest, TestCombinedAudioVideoBweConstraint) {
   constraints_.reset(new FakeConstraints());
   constraints_->AddOptional(
       webrtc::MediaConstraintsInterface::kCombinedAudioVideoBwe,
       true);
   Init();
   SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   SetLocalDescriptionWithoutError(offer);
 
   voice_channel_ = media_engine_->GetVoiceChannel(0);
 
   ASSERT_TRUE(voice_channel_ != NULL);
   const cricket::AudioOptions& audio_options = voice_channel_->options();
-  EXPECT_EQ(rtc::Maybe<bool>(true), audio_options.combined_audio_video_bwe);
+  EXPECT_EQ(rtc::Optional<bool>(true), audio_options.combined_audio_video_bwe);
 }
 
 // Tests that we can renegotiate new media content with ICE candidates in the
 // new remote SDP.
 TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesInSdp) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   InitWithDtls(GetParam());
   SetFactoryDtlsSrtp();
 
   SendAudioOnlyStream2();
@@ skipping 153 matching lines @@
 }
 
 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled.  That test
 // currently fails because upon disconnection and reconnection OnIceComplete is
 // called more than once without returning to IceGatheringGathering.
 
 INSTANTIATE_TEST_CASE_P(WebRtcSessionTests,
                         WebRtcSessionTest,
                         testing::Values(ALREADY_GENERATED,
                                         DTLS_IDENTITY_STORE));