Optional: Use nullopt and implicit construction in /rtc_tools/event_log_visualizer
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.
R=terelius@webrtc.org
Bug: None
Change-Id: I18b44f6c1bd3ccb4288807eee883b87afa7900f4
Reviewed-on: https://webrtc-review.googlesource.com/23569
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20875}diff --git a/rtc_tools/event_log_visualizer/analyzer.cc b/rtc_tools/event_log_visualizer/analyzer.cc
index 0ef0803..bbbde3f 100644
--- a/rtc_tools/event_log_visualizer/analyzer.cc
+++ b/rtc_tools/event_log_visualizer/analyzer.cc
@@ -164,9 +164,9 @@
int64_t recv_time_diff = new_packet.timestamp - old_packet.timestamp;
double delay_change_us =
recv_time_diff - AbsSendTimeToMicroseconds(send_time_diff);
- return rtc::Optional<double>(delay_change_us / 1000);
+ return delay_change_us / 1000;
} else {
- return rtc::Optional<double>();
+ return rtc::nullopt;
}
}
@@ -199,7 +199,7 @@
kVideoSampleRate
<< "s";
}
- return rtc::Optional<double>(delay_change);
+ return delay_change;
}
// For each element in data, use |get_y()| to extract a y-coordinate and
@@ -477,7 +477,7 @@
log_segments_.push_back(
std::make_pair(*last_log_start, last_timestamp));
}
- last_log_start = rtc::Optional<uint64_t>(parsed_log_.GetTimestamp(i));
+ last_log_start = parsed_log_.GetTimestamp(i);
break;
}
case ParsedRtcEventLog::LOG_END: {
@@ -624,7 +624,7 @@
<< "Failed to estimate RTP clock frequency: Stream too short. ("
<< packets.size() << " packets, "
<< last_log_timestamp - first_log_timestamp << " us)";
- return rtc::Optional<uint32_t>();
+ return rtc::nullopt;
}
double duration =
static_cast<double>(last_log_timestamp - first_log_timestamp) / 1000000;
@@ -632,13 +632,13 @@
(last_rtp_timestamp - first_rtp_timestamp) / duration;
for (uint32_t f : {8000, 16000, 32000, 48000, 90000}) {
if (std::fabs(estimated_frequency - f) < 0.05 * f) {
- return rtc::Optional<uint32_t>(f);
+ return f;
}
}
RTC_LOG(LS_WARNING) << "Failed to estimate RTP clock frequency: Estimate "
<< estimated_frequency
<< "not close to any stardard RTP frequency.";
- return rtc::Optional<uint32_t>();
+ return rtc::nullopt;
}
void EventLogAnalyzer::CreatePacketGraph(PacketDirection desired_direction,
@@ -654,7 +654,7 @@
TimeSeries time_series(GetStreamName(stream_id), LineStyle::kBar);
ProcessPoints<LoggedRtpPacket>(
- [](const LoggedRtpPacket& packet) -> rtc::Optional<float> {
+ [](const LoggedRtpPacket& packet) {
return rtc::Optional<float>(packet.total_length);
},
packet_stream, begin_time_, &time_series);
@@ -805,7 +805,7 @@
int64_t diff =
WrappingDifference(new_packet.header.sequenceNumber,
old_packet.header.sequenceNumber, 1ul << 16);
- return rtc::Optional<float>(diff);
+ return diff;
},
packet_stream, begin_time_, &time_series);
plot->AppendTimeSeries(std::move(time_series));
@@ -1146,7 +1146,7 @@
TimeSeries time_series(GetStreamName(stream_id), LineStyle::kLine);
MovingAverage<LoggedRtpPacket, double>(
[](const LoggedRtpPacket& packet) {
- return rtc::Optional<double>(packet.total_length * 8.0 / 1000.0);
+ return packet.total_length * 8.0 / 1000.0;
},
packet_stream, begin_time_, end_time_, window_duration_, step_,
&time_series);
@@ -1613,9 +1613,8 @@
ProcessPoints<AudioNetworkAdaptationEvent>(
[](const AudioNetworkAdaptationEvent& ana_event) -> rtc::Optional<float> {
if (ana_event.config.bitrate_bps)
- return rtc::Optional<float>(
- static_cast<float>(*ana_event.config.bitrate_bps));
- return rtc::Optional<float>();
+ return static_cast<float>(*ana_event.config.bitrate_bps);
+ return rtc::nullopt;
},
audio_network_adaptation_events_, begin_time_, &time_series);
plot->AppendTimeSeries(std::move(time_series));
@@ -1729,25 +1728,24 @@
rtc::Optional<int64_t> NextPacketTime() const override {
if (packet_stream_it_ == packet_stream_.end()) {
- return rtc::Optional<int64_t>();
+ return rtc::nullopt;
}
if (end_time_us_ && packet_stream_it_->timestamp > *end_time_us_) {
- return rtc::Optional<int64_t>();
+ return rtc::nullopt;
}
// Convert from us to ms.
- return rtc::Optional<int64_t>(packet_stream_it_->timestamp / 1000);
+ return packet_stream_it_->timestamp / 1000;
}
rtc::Optional<int64_t> NextOutputEventTime() const override {
if (output_events_us_it_ == output_events_us_end_) {
- return rtc::Optional<int64_t>();
+ return rtc::nullopt;
}
if (end_time_us_ && *output_events_us_it_ > *end_time_us_) {
- return rtc::Optional<int64_t>();
+ return rtc::nullopt;
}
// Convert from us to ms.
- return rtc::Optional<int64_t>(
- rtc::checked_cast<int64_t>(*output_events_us_it_ / 1000));
+ return rtc::checked_cast<int64_t>(*output_events_us_it_ / 1000);
}
std::unique_ptr<PacketData> PopPacket() override {
@@ -1778,9 +1776,9 @@
rtc::Optional<RTPHeader> NextHeader() const override {
if (packet_stream_it_ == packet_stream_.end()) {
- return rtc::Optional<RTPHeader>();
+ return rtc::nullopt;
}
- return rtc::Optional<RTPHeader>(packet_stream_it_->header);
+ return packet_stream_it_->header;
}
private:
@@ -1876,7 +1874,7 @@
rtc::Optional<uint64_t> end_time_us =
log_segments_.empty()
- ? rtc::Optional<uint64_t>()
+ ? rtc::nullopt
: rtc::Optional<uint64_t>(log_segments_.front().second);
auto delay_cb = CreateNetEqTestAndRun(
