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Optional: Use nullopt and implicit construction in /rtc_tools/event_log_visualizer

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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}
This commit is contained in:
Oskar Sundbom
2017-11-16 10:53:09 +01:00
committed by Commit Bot
parent 69d276d7dc
commit 3928dbc1e7
1 changed files with 21 additions and 23 deletions
Download Patch File Download Diff File Expand all files Collapse all files

44
rtc_tools/event_log_visualizer/analyzer.cc
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@ -164,9 +164,9 @@ rtc::Optional<double> NetworkDelayDiff_AbsSendTime(
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 @@ rtc::Optional<double> NetworkDelayDiff_CaptureTime(
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 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
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 @@ rtc::Optional<uint32_t> EventLogAnalyzer::EstimateRtpClockFrequency(
<< "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 @@ rtc::Optional<uint32_t> EventLogAnalyzer::EstimateRtpClockFrequency(
(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 @@ void EventLogAnalyzer::CreatePacketGraph(PacketDirection desired_direction,
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 @@ void EventLogAnalyzer::CreateSequenceNumberGraph(Plot* plot) {
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 @@ void EventLogAnalyzer::CreateStreamBitrateGraph(
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 @@ void EventLogAnalyzer::CreateAudioEncoderTargetBitrateGraph(Plot* plot) {
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 @@ class NetEqStreamInput : public test::NetEqInput {
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 @@ class NetEqStreamInput : public test::NetEqInput {
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 @@ void EventLogAnalyzer::CreateAudioJitterBufferGraph(
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(