commit 3f6804d140275182a80d5942a7638f9836443737
author Oskar Sundbom <ossu@webrtc.org> Thu Nov 16 10:54:58 2017 +0100
committer Commit Bot <commit-bot@chromium.org> Thu Jan 11 13:31:01 2018 +0000
tree b8b62a04df090ea0c92d4e25c5d94760e690c629
parent 1db508352c219077af917d426c889da1b32b077c

Optional: Use nullopt and implicit construction in /test

Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split

Bug: None
Change-Id: Ifca0183c2675a8d4e74c21b86f1724d9b2c01f5d
Reviewed-on: https://webrtc-review.googlesource.com/23579
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21580}

test/call_test.cc

diff --git a/test/call_test.cc b/test/call_test.cc
index 5fbba88..1483216 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -218,9 +218,8 @@
   if (num_audio_streams > 0) {
     audio_send_config_ = AudioSendStream::Config(send_transport);
     audio_send_config_.rtp.ssrc = kAudioSendSsrc;
-    audio_send_config_.send_codec_spec =
-        rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
-            {kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}}});
+    audio_send_config_.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
+        kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}});
     audio_send_config_.encoder_factory = encoder_factory_;
   }
 

test/frame_generator.cc

diff --git a/test/frame_generator.cc b/test/frame_generator.cc
index e5d9950..d6cc4e5 100644
--- a/test/frame_generator.cc
+++ b/test/frame_generator.cc
@@ -336,13 +336,13 @@
     int offset_v = (i420_buffer->StrideV() * (pixels_scrolled_y / 2)) +
                    (pixels_scrolled_x / 2);
 
-    current_frame_ = rtc::Optional<webrtc::VideoFrame>(webrtc::VideoFrame(
+    current_frame_ = webrtc::VideoFrame(
         new rtc::RefCountedObject<webrtc::WrappedI420Buffer>(
             target_width_, target_height_, &i420_buffer->DataY()[offset_y],
             i420_buffer->StrideY(), &i420_buffer->DataU()[offset_u],
             i420_buffer->StrideU(), &i420_buffer->DataV()[offset_v],
             i420_buffer->StrideV(), KeepRefUntilDone(i420_buffer)),
-        kVideoRotation_0, 0));
+        kVideoRotation_0, 0);
   }
 
   Clock* const clock_;

test/mock_audio_encoder_factory.h

diff --git a/test/mock_audio_encoder_factory.h b/test/mock_audio_encoder_factory.h
index 9b63f7d..dea48da 100644
--- a/test/mock_audio_encoder_factory.h
+++ b/test/mock_audio_encoder_factory.h
@@ -51,7 +51,7 @@
     ON_CALL(*factory.get(), GetSupportedEncoders())
         .WillByDefault(Return(std::vector<webrtc::AudioCodecSpec>()));
     ON_CALL(*factory.get(), QueryAudioEncoder(_))
-        .WillByDefault(Return(rtc::Optional<AudioCodecInfo>()));
+        .WillByDefault(Return(rtc::nullopt));
 
     EXPECT_CALL(*factory.get(), GetSupportedEncoders()).Times(AnyNumber());
     EXPECT_CALL(*factory.get(), QueryAudioEncoder(_)).Times(AnyNumber());
@@ -74,7 +74,7 @@
     ON_CALL(*factory.get(), GetSupportedEncoders())
         .WillByDefault(Return(std::vector<webrtc::AudioCodecSpec>()));
     ON_CALL(*factory.get(), QueryAudioEncoder(_))
-        .WillByDefault(Return(rtc::Optional<AudioCodecInfo>()));
+        .WillByDefault(Return(rtc::nullopt));
     ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _))
         .WillByDefault(SetArgPointee<2>(nullptr));
 

test/video_capturer.cc

diff --git a/test/video_capturer.cc b/test/video_capturer.cc
index b95936f..b48bc1a 100644
--- a/test/video_capturer.cc
+++ b/test/video_capturer.cc
@@ -28,7 +28,7 @@
           frame.width(), frame.height(), frame.timestamp_us() * 1000,
           &cropped_width, &cropped_height, &out_width, &out_height)) {
     // Drop frame in order to respect frame rate constraint.
-    return rtc::Optional<VideoFrame>();
+    return rtc::nullopt;
   }
 
   rtc::Optional<VideoFrame> out_frame;