Optional: Use nullopt and implicit construction in /logging
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: Ibe15d2814074a4cf67de18d6e04540076f1a9dc9
Reviewed-on: https://webrtc-review.googlesource.com/23611
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20862}diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
index 2221bf4..ce4297c 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
@@ -148,14 +148,14 @@
TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationBitrate) {
auto runtime_config = rtc::MakeUnique<AudioEncoderRuntimeConfig>();
const int bitrate_bps = RandomBitrate();
- runtime_config->bitrate_bps = rtc::Optional<int>(bitrate_bps);
+ runtime_config->bitrate_bps = bitrate_bps;
TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
}
TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationFrameLength) {
auto runtime_config = rtc::MakeUnique<AudioEncoderRuntimeConfig>();
const int frame_length_ms = prng_.Rand(1, 1000);
- runtime_config->frame_length_ms = rtc::Optional<int>(frame_length_ms);
+ runtime_config->frame_length_ms = frame_length_ms;
TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
}
@@ -163,7 +163,7 @@
// To simplify the test, we just check powers of two.
const float plr = std::pow(0.5f, prng_.Rand(1, 8));
auto runtime_config = rtc::MakeUnique<AudioEncoderRuntimeConfig>();
- runtime_config->uplink_packet_loss_fraction = rtc::Optional<float>(plr);
+ runtime_config->uplink_packet_loss_fraction = plr;
TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
}
@@ -172,7 +172,7 @@
// for safety's sake, we test both.
for (bool fec_enabled : {false, true}) {
auto runtime_config = rtc::MakeUnique<AudioEncoderRuntimeConfig>();
- runtime_config->enable_fec = rtc::Optional<bool>(fec_enabled);
+ runtime_config->enable_fec = fec_enabled;
TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
}
}
@@ -182,7 +182,7 @@
// for safety's sake, we test both.
for (bool dtx_enabled : {false, true}) {
auto runtime_config = rtc::MakeUnique<AudioEncoderRuntimeConfig>();
- runtime_config->enable_dtx = rtc::Optional<bool>(dtx_enabled);
+ runtime_config->enable_dtx = dtx_enabled;
TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
}
}
@@ -192,7 +192,7 @@
// for safety's sake, we test both.
for (size_t channels : {1, 2}) {
auto runtime_config = rtc::MakeUnique<AudioEncoderRuntimeConfig>();
- runtime_config->num_channels = rtc::Optional<size_t>(channels);
+ runtime_config->num_channels = channels;
TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
}
}
@@ -205,12 +205,12 @@
for (bool dtx_enabled : {false, true}) {
for (size_t channels : {1, 2}) {
auto runtime_config = rtc::MakeUnique<AudioEncoderRuntimeConfig>();
- runtime_config->bitrate_bps = rtc::Optional<int>(bitrate_bps);
- runtime_config->frame_length_ms = rtc::Optional<int>(frame_length_ms);
- runtime_config->uplink_packet_loss_fraction = rtc::Optional<float>(plr);
- runtime_config->enable_fec = rtc::Optional<bool>(fec_enabled);
- runtime_config->enable_dtx = rtc::Optional<bool>(dtx_enabled);
- runtime_config->num_channels = rtc::Optional<size_t>(channels);
+ runtime_config->bitrate_bps = bitrate_bps;
+ runtime_config->frame_length_ms = frame_length_ms;
+ runtime_config->uplink_packet_loss_fraction = plr;
+ runtime_config->enable_fec = fec_enabled;
+ runtime_config->enable_dtx = dtx_enabled;
+ runtime_config->num_channels = channels;
TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
}
diff --git a/logging/rtc_event_log/rtc_event_log_parser.cc b/logging/rtc_event_log/rtc_event_log_parser.cc
index 060c553..38697e9 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -585,18 +585,18 @@
const rtclog::AudioNetworkAdaptation& ana_event =
event.audio_network_adaptation();
if (ana_event.has_bitrate_bps())
- config->bitrate_bps = rtc::Optional<int>(ana_event.bitrate_bps());
+ config->bitrate_bps = ana_event.bitrate_bps();
if (ana_event.has_enable_fec())
- config->enable_fec = rtc::Optional<bool>(ana_event.enable_fec());
+ config->enable_fec = ana_event.enable_fec();
if (ana_event.has_enable_dtx())
- config->enable_dtx = rtc::Optional<bool>(ana_event.enable_dtx());
+ config->enable_dtx = ana_event.enable_dtx();
if (ana_event.has_frame_length_ms())
- config->frame_length_ms = rtc::Optional<int>(ana_event.frame_length_ms());
+ config->frame_length_ms = ana_event.frame_length_ms();
if (ana_event.has_num_channels())
- config->num_channels = rtc::Optional<size_t>(ana_event.num_channels());
+ config->num_channels = ana_event.num_channels();
if (ana_event.has_uplink_packet_loss_fraction())
config->uplink_packet_loss_fraction =
- rtc::Optional<float>(ana_event.uplink_packet_loss_fraction());
+ ana_event.uplink_packet_loss_fraction();
}
ParsedRtcEventLog::BweProbeClusterCreatedEvent
@@ -636,18 +636,15 @@
RTC_CHECK(pr_event.has_result());
if (pr_event.result() == rtclog::BweProbeResult::SUCCESS) {
RTC_CHECK(pr_event.has_bitrate_bps());
- res.bitrate_bps = rtc::Optional<uint64_t>(pr_event.bitrate_bps());
+ res.bitrate_bps = pr_event.bitrate_bps();
} else if (pr_event.result() ==
rtclog::BweProbeResult::INVALID_SEND_RECEIVE_INTERVAL) {
- res.failure_reason = rtc::Optional<ProbeFailureReason>(
- ProbeFailureReason::kInvalidSendReceiveInterval);
+ res.failure_reason = ProbeFailureReason::kInvalidSendReceiveInterval;
} else if (pr_event.result() ==
rtclog::BweProbeResult::INVALID_SEND_RECEIVE_RATIO) {
- res.failure_reason = rtc::Optional<ProbeFailureReason>(
- ProbeFailureReason::kInvalidSendReceiveRatio);
+ res.failure_reason = ProbeFailureReason::kInvalidSendReceiveRatio;
} else if (pr_event.result() == rtclog::BweProbeResult::TIMEOUT) {
- res.failure_reason =
- rtc::Optional<ProbeFailureReason>(ProbeFailureReason::kTimeout);
+ res.failure_reason = ProbeFailureReason::kTimeout;
} else {
RTC_NOTREACHED();
}
diff --git a/logging/rtc_event_log/rtc_event_log_unittest.cc b/logging/rtc_event_log/rtc_event_log_unittest.cc
index fbef901..1567323 100644
--- a/logging/rtc_event_log/rtc_event_log_unittest.cc
+++ b/logging/rtc_event_log/rtc_event_log_unittest.cc
@@ -291,13 +291,12 @@
void GenerateAudioNetworkAdaptation(const RtpHeaderExtensionMap& extensions,
AudioEncoderRuntimeConfig* config,
Random* prng) {
- config->bitrate_bps = rtc::Optional<int>(prng->Rand(0, 3000000));
- config->enable_fec = rtc::Optional<bool>(prng->Rand<bool>());
- config->enable_dtx = rtc::Optional<bool>(prng->Rand<bool>());
- config->frame_length_ms = rtc::Optional<int>(prng->Rand(10, 120));
- config->num_channels = rtc::Optional<size_t>(prng->Rand(1, 2));
- config->uplink_packet_loss_fraction =
- rtc::Optional<float>(prng->Rand<float>());
+ config->bitrate_bps = prng->Rand(0, 3000000);
+ config->enable_fec = prng->Rand<bool>();
+ config->enable_dtx = prng->Rand<bool>();
+ config->frame_length_ms = prng->Rand(10, 120);
+ config->num_channels = prng->Rand(1, 2);
+ config->uplink_packet_loss_fraction = prng->Rand<float>();
}
class RtcEventLogSession
