Replace rtc::Optional with absl::optional in test and rtc_tools
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'test rtc_tools'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ibb43c737f4c45fe300736382b0dd2d8ab32c6377
Reviewed-on: https://webrtc-review.googlesource.com/83944
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23642}
This commit is contained in:
Danil Chapovalov
committed by
Commit Bot
Commit Bot
parent
9bf31584d1
commit
431abd989b
@ -130,7 +130,7 @@ int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus) {
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// This is much more reliable for outgoing streams than for incoming streams.
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template <typename RtpPacketContainer>
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rtc::Optional<uint32_t> EstimateRtpClockFrequency(
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absl::optional<uint32_t> EstimateRtpClockFrequency(
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const RtpPacketContainer& packets,
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int64_t end_time_us) {
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RTC_CHECK(packets.size() >= 2);
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@ -151,7 +151,7 @@ rtc::Optional<uint32_t> EstimateRtpClockFrequency(
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<< "Failed to estimate RTP clock frequency: Stream too short. ("
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<< packets.size() << " packets, "
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<< last_log_timestamp - first_log_timestamp << " us)";
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return rtc::nullopt;
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return absl::nullopt;
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}
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double duration =
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static_cast<double>(last_log_timestamp - first_log_timestamp) /
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@ -166,7 +166,7 @@ rtc::Optional<uint32_t> EstimateRtpClockFrequency(
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RTC_LOG(LS_WARNING) << "Failed to estimate RTP clock frequency: Estimate "
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<< estimated_frequency
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<< "not close to any stardard RTP frequency.";
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return rtc::nullopt;
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return absl::nullopt;
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}
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constexpr float kLeftMargin = 0.01f;
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@ -174,7 +174,7 @@ constexpr float kRightMargin = 0.02f;
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constexpr float kBottomMargin = 0.02f;
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constexpr float kTopMargin = 0.05f;
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rtc::Optional<double> NetworkDelayDiff_AbsSendTime(
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absl::optional<double> NetworkDelayDiff_AbsSendTime(
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const LoggedRtpPacket& old_packet,
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const LoggedRtpPacket& new_packet) {
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if (old_packet.header.extension.hasAbsoluteSendTime &&
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@ -188,11 +188,11 @@ rtc::Optional<double> NetworkDelayDiff_AbsSendTime(
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recv_time_diff - AbsSendTimeToMicroseconds(send_time_diff);
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return delay_change_us / 1000;
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} else {
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return rtc::nullopt;
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return absl::nullopt;
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}
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}
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rtc::Optional<double> NetworkDelayDiff_CaptureTime(
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absl::optional<double> NetworkDelayDiff_CaptureTime(
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const LoggedRtpPacket& old_packet,
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const LoggedRtpPacket& new_packet) {
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int64_t send_time_diff = WrappingDifference(
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@ -230,13 +230,13 @@ rtc::Optional<double> NetworkDelayDiff_CaptureTime(
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// store the result in a TimeSeries.
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template <typename DataType, typename IterableType>
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void ProcessPoints(rtc::FunctionView<float(const DataType&)> fx,
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rtc::FunctionView<rtc::Optional<float>(const DataType&)> fy,
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rtc::FunctionView<absl::optional<float>(const DataType&)> fy,
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const IterableType& data_view,
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TimeSeries* result) {
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for (size_t i = 0; i < data_view.size(); i++) {
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const DataType& elem = data_view[i];
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float x = fx(elem);
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rtc::Optional<float> y = fy(elem);
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absl::optional<float> y = fy(elem);
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if (y)
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result->points.emplace_back(x, *y);
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}
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@ -248,13 +248,13 @@ void ProcessPoints(rtc::FunctionView<float(const DataType&)> fx,
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template <typename DataType, typename ResultType, typename IterableType>
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void ProcessPairs(
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rtc::FunctionView<float(const DataType&)> fx,
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rtc::FunctionView<rtc::Optional<ResultType>(const DataType&,
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const DataType&)> fy,
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rtc::FunctionView<absl::optional<ResultType>(const DataType&,
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const DataType&)> fy,
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const IterableType& data,
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TimeSeries* result) {
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for (size_t i = 1; i < data.size(); i++) {
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float x = fx(data[i]);
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rtc::Optional<ResultType> y = fy(data[i - 1], data[i]);
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absl::optional<ResultType> y = fy(data[i - 1], data[i]);
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if (y)
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result->points.emplace_back(x, static_cast<float>(*y));
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}
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@ -266,14 +266,14 @@ void ProcessPairs(
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template <typename DataType, typename ResultType, typename IterableType>
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void AccumulatePairs(
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rtc::FunctionView<float(const DataType&)> fx,
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rtc::FunctionView<rtc::Optional<ResultType>(const DataType&,
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const DataType&)> fy,
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rtc::FunctionView<absl::optional<ResultType>(const DataType&,
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const DataType&)> fy,
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const IterableType& data,
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TimeSeries* result) {
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ResultType sum = 0;
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for (size_t i = 1; i < data.size(); i++) {
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float x = fx(data[i]);
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rtc::Optional<ResultType> y = fy(data[i - 1], data[i]);
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absl::optional<ResultType> y = fy(data[i - 1], data[i]);
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if (y)
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sum += *y;
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result->points.emplace_back(x, static_cast<float>(sum));
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@ -287,7 +287,7 @@ void AccumulatePairs(
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template <typename DataType, typename ResultType, typename IterableType>
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void MovingAverage(
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rtc::FunctionView<float(int64_t)> fx,
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rtc::FunctionView<rtc::Optional<ResultType>(const DataType&)> fy,
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rtc::FunctionView<absl::optional<ResultType>(const DataType&)> fy,
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const IterableType& data_view,
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int64_t begin_time,
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int64_t end_time,
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@ -301,7 +301,7 @@ void MovingAverage(
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for (int64_t t = begin_time; t < end_time + step; t += step) {
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while (window_index_end < data_view.size() &&
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data_view[window_index_end].log_time_us() < t) {
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rtc::Optional<ResultType> value = fy(data_view[window_index_end]);
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absl::optional<ResultType> value = fy(data_view[window_index_end]);
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if (value)
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sum_in_window += *value;
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++window_index_end;
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@ -309,7 +309,7 @@ void MovingAverage(
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while (window_index_begin < data_view.size() &&
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data_view[window_index_begin].log_time_us() <
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t - window_duration_us) {
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rtc::Optional<ResultType> value = fy(data_view[window_index_begin]);
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absl::optional<ResultType> value = fy(data_view[window_index_begin]);
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if (value)
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sum_in_window -= *value;
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++window_index_begin;
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@ -465,7 +465,7 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLogNew& log,
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while (start_iter != log_start_events.end()) {
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int64_t start = start_iter->log_time_us();
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++start_iter;
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rtc::Optional<int64_t> next_start;
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absl::optional<int64_t> next_start;
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if (start_iter != log_start_events.end())
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next_start.emplace(start_iter->log_time_us());
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if (end_iter != log_end_events.end() &&
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@ -537,7 +537,7 @@ void EventLogAnalyzer::CreatePacketGraph(PacketDirection direction,
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TimeSeries time_series(GetStreamName(direction, stream.ssrc),
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LineStyle::kBar);
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auto GetPacketSize = [](const LoggedRtpPacket& packet) {
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return rtc::Optional<float>(packet.total_length);
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return absl::optional<float>(packet.total_length);
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};
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auto ToCallTime = [this](const LoggedRtpPacket& packet) {
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return this->ToCallTimeSec(packet.log_time_us());
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@ -599,7 +599,7 @@ void EventLogAnalyzer::CreatePlayoutGraph(Plot* plot) {
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uint32_t ssrc = playout_stream.first;
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if (!MatchingSsrc(ssrc, desired_ssrc_))
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continue;
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rtc::Optional<int64_t> last_playout_ms;
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absl::optional<int64_t> last_playout_ms;
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TimeSeries time_series(SsrcToString(ssrc), LineStyle::kBar);
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for (const auto& playout_event : playout_stream.second) {
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float x = ToCallTimeSec(playout_event.log_time_us());
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@ -1139,7 +1139,7 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
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cc.OnTransportFeedback(rtcp_iterator->transport_feedback);
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std::vector<PacketFeedback> feedback = cc.GetTransportFeedbackVector();
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SortPacketFeedbackVector(&feedback);
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rtc::Optional<uint32_t> bitrate_bps;
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absl::optional<uint32_t> bitrate_bps;
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if (!feedback.empty()) {
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#if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
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acknowledged_bitrate_estimator.IncomingPacketFeedbackVector(feedback);
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@ -1251,7 +1251,7 @@ void EventLogAnalyzer::CreateReceiveSideBweSimulationGraph(Plot* plot) {
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clock.TimeInMicroseconds());
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rscc.OnReceivedPacket(arrival_time_ms, payload, packet.rtp.header);
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acked_bitrate.Update(payload, arrival_time_ms);
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rtc::Optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
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absl::optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
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if (bitrate_bps) {
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uint32_t y = *bitrate_bps / 1000;
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float x = ToCallTimeSec(clock.TimeInMicroseconds());
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@ -1383,7 +1383,7 @@ void EventLogAnalyzer::CreatePacerDelayGraph(Plot* plot) {
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int64_t end_time_us = log_segments_.empty()
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? std::numeric_limits<int64_t>::max()
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: log_segments_.front().second;
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rtc::Optional<uint32_t> estimated_frequency =
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absl::optional<uint32_t> estimated_frequency =
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EstimateRtpClockFrequency(packets, end_time_us);
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if (!estimated_frequency)
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continue;
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@ -1463,11 +1463,11 @@ void EventLogAnalyzer::CreateAudioEncoderTargetBitrateGraph(Plot* plot) {
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TimeSeries time_series("Audio encoder target bitrate", LineStyle::kLine,
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PointStyle::kHighlight);
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auto GetAnaBitrateBps = [](const LoggedAudioNetworkAdaptationEvent& ana_event)
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-> rtc::Optional<float> {
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-> absl::optional<float> {
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if (ana_event.config.bitrate_bps)
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return rtc::Optional<float>(
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return absl::optional<float>(
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static_cast<float>(*ana_event.config.bitrate_bps));
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return rtc::nullopt;
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return absl::nullopt;
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};
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auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
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return this->ToCallTimeSec(packet.log_time_us());
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@ -1488,9 +1488,9 @@ void EventLogAnalyzer::CreateAudioEncoderFrameLengthGraph(Plot* plot) {
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auto GetAnaFrameLengthMs =
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[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
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if (ana_event.config.frame_length_ms)
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return rtc::Optional<float>(
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return absl::optional<float>(
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static_cast<float>(*ana_event.config.frame_length_ms));
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return rtc::Optional<float>();
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return absl::optional<float>();
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};
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auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
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return this->ToCallTimeSec(packet.log_time_us());
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@ -1511,9 +1511,9 @@ void EventLogAnalyzer::CreateAudioEncoderPacketLossGraph(Plot* plot) {
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auto GetAnaPacketLoss =
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[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
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if (ana_event.config.uplink_packet_loss_fraction)
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return rtc::Optional<float>(static_cast<float>(
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return absl::optional<float>(static_cast<float>(
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*ana_event.config.uplink_packet_loss_fraction));
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return rtc::Optional<float>();
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return absl::optional<float>();
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};
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auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
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return this->ToCallTimeSec(packet.log_time_us());
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@ -1535,9 +1535,9 @@ void EventLogAnalyzer::CreateAudioEncoderEnableFecGraph(Plot* plot) {
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auto GetAnaFecEnabled =
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[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
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if (ana_event.config.enable_fec)
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return rtc::Optional<float>(
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return absl::optional<float>(
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static_cast<float>(*ana_event.config.enable_fec));
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return rtc::Optional<float>();
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return absl::optional<float>();
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};
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auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
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return this->ToCallTimeSec(packet.log_time_us());
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@ -1558,9 +1558,9 @@ void EventLogAnalyzer::CreateAudioEncoderEnableDtxGraph(Plot* plot) {
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auto GetAnaDtxEnabled =
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[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
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if (ana_event.config.enable_dtx)
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return rtc::Optional<float>(
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return absl::optional<float>(
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static_cast<float>(*ana_event.config.enable_dtx));
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return rtc::Optional<float>();
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return absl::optional<float>();
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};
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auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
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return this->ToCallTimeSec(packet.log_time_us());
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@ -1581,9 +1581,9 @@ void EventLogAnalyzer::CreateAudioEncoderNumChannelsGraph(Plot* plot) {
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auto GetAnaNumChannels =
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[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
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if (ana_event.config.num_channels)
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return rtc::Optional<float>(
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return absl::optional<float>(
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static_cast<float>(*ana_event.config.num_channels));
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return rtc::Optional<float>();
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return absl::optional<float>();
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};
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auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
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return this->ToCallTimeSec(packet.log_time_us());
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@ -1605,7 +1605,7 @@ class NetEqStreamInput : public test::NetEqInput {
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// that outlive the one constructed.
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NetEqStreamInput(const std::vector<LoggedRtpPacketIncoming>* packet_stream,
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const std::vector<LoggedAudioPlayoutEvent>* output_events,
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rtc::Optional<int64_t> end_time_ms)
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absl::optional<int64_t> end_time_ms)
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: packet_stream_(*packet_stream),
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packet_stream_it_(packet_stream_.begin()),
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output_events_it_(output_events->begin()),
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@ -1615,22 +1615,22 @@ class NetEqStreamInput : public test::NetEqInput {
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RTC_DCHECK(output_events);
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}
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rtc::Optional<int64_t> NextPacketTime() const override {
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absl::optional<int64_t> NextPacketTime() const override {
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if (packet_stream_it_ == packet_stream_.end()) {
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return rtc::nullopt;
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return absl::nullopt;
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}
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if (end_time_ms_ && packet_stream_it_->rtp.log_time_ms() > *end_time_ms_) {
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return rtc::nullopt;
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return absl::nullopt;
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}
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return packet_stream_it_->rtp.log_time_ms();
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}
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rtc::Optional<int64_t> NextOutputEventTime() const override {
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absl::optional<int64_t> NextOutputEventTime() const override {
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if (output_events_it_ == output_events_end_) {
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return rtc::nullopt;
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return absl::nullopt;
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}
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if (end_time_ms_ && output_events_it_->log_time_ms() > *end_time_ms_) {
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return rtc::nullopt;
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return absl::nullopt;
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}
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return output_events_it_->log_time_ms();
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}
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@ -1661,9 +1661,9 @@ class NetEqStreamInput : public test::NetEqInput {
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bool ended() const override { return !NextEventTime(); }
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rtc::Optional<RTPHeader> NextHeader() const override {
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absl::optional<RTPHeader> NextHeader() const override {
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if (packet_stream_it_ == packet_stream_.end()) {
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return rtc::nullopt;
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return absl::nullopt;
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}
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return packet_stream_it_->rtp.header;
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}
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@ -1673,7 +1673,7 @@ class NetEqStreamInput : public test::NetEqInput {
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std::vector<LoggedRtpPacketIncoming>::const_iterator packet_stream_it_;
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std::vector<LoggedAudioPlayoutEvent>::const_iterator output_events_it_;
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const std::vector<LoggedAudioPlayoutEvent>::const_iterator output_events_end_;
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const rtc::Optional<int64_t> end_time_ms_;
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const absl::optional<int64_t> end_time_ms_;
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};
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namespace {
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@ -1683,7 +1683,7 @@ namespace {
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std::unique_ptr<test::NetEqStatsGetter> CreateNetEqTestAndRun(
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const std::vector<LoggedRtpPacketIncoming>* packet_stream,
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const std::vector<LoggedAudioPlayoutEvent>* output_events,
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rtc::Optional<int64_t> end_time_ms,
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absl::optional<int64_t> end_time_ms,
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const std::string& replacement_file_name,
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int file_sample_rate_hz) {
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std::unique_ptr<test::NetEqInput> input(
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@ -1759,10 +1759,10 @@ EventLogAnalyzer::NetEqStatsGetterMap EventLogAnalyzer::SimulateNetEq(
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output_events_it = parsed_log_.audio_playout_events().cbegin();
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}
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rtc::Optional<int64_t> end_time_ms =
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absl::optional<int64_t> end_time_ms =
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log_segments_.empty()
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? rtc::nullopt
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: rtc::Optional<int64_t>(log_segments_.front().second / 1000);
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? absl::nullopt
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: absl::optional<int64_t>(log_segments_.front().second / 1000);
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neteq_stats[ssrc] = CreateNetEqTestAndRun(
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audio_packets, &output_events_it->second, end_time_ms,
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@ -1786,8 +1786,8 @@ void EventLogAnalyzer::CreateAudioJitterBufferGraph(
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std::vector<float> send_times_s;
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std::vector<float> arrival_delay_ms;
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std::vector<float> corrected_arrival_delay_ms;
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std::vector<rtc::Optional<float>> playout_delay_ms;
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std::vector<rtc::Optional<float>> target_delay_ms;
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std::vector<absl::optional<float>> playout_delay_ms;
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std::vector<absl::optional<float>> target_delay_ms;
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neteq_stats.at(ssrc)->delay_analyzer()->CreateGraphs(
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&send_times_s, &arrival_delay_ms, &corrected_arrival_delay_ms,
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&playout_delay_ms, &target_delay_ms);
|
||||
@ -2014,9 +2014,9 @@ void EventLogAnalyzer::CreateStreamGapAlerts(PacketDirection direction) {
|
||||
: log_segments_.front().second;
|
||||
|
||||
SeqNumUnwrapper<uint16_t> seq_num_unwrapper;
|
||||
rtc::Optional<int64_t> last_seq_num;
|
||||
absl::optional<int64_t> last_seq_num;
|
||||
SeqNumUnwrapper<uint32_t> capture_time_unwrapper;
|
||||
rtc::Optional<int64_t> last_capture_time;
|
||||
absl::optional<int64_t> last_capture_time;
|
||||
// Check for gaps in sequence numbers and capture timestamps.
|
||||
for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
|
||||
for (const auto& packet : stream.packet_view) {
|
||||
@ -2060,7 +2060,7 @@ void EventLogAnalyzer::CreateTransmissionGapAlerts(PacketDirection direction) {
|
||||
for (const LoggedRtpPacket& rtp_packet : stream.packet_view)
|
||||
rtp_in_direction.emplace(rtp_packet.log_time_us(), &rtp_packet);
|
||||
}
|
||||
rtc::Optional<int64_t> last_rtp_time;
|
||||
absl::optional<int64_t> last_rtp_time;
|
||||
for (const auto& kv : rtp_in_direction) {
|
||||
int64_t timestamp = kv.first;
|
||||
if (timestamp > end_time_us) {
|
||||
@ -2075,7 +2075,7 @@ void EventLogAnalyzer::CreateTransmissionGapAlerts(PacketDirection direction) {
|
||||
last_rtp_time.emplace(timestamp);
|
||||
}
|
||||
|
||||
rtc::Optional<int64_t> last_rtcp_time;
|
||||
absl::optional<int64_t> last_rtcp_time;
|
||||
if (direction == kIncomingPacket) {
|
||||
for (const auto& rtcp : parsed_log_.incoming_rtcp_packets()) {
|
||||
if (rtcp.log_time_us() > end_time_us) {
|
||||
|
||||
Reference in New Issue
Block a user