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}diff --git a/rtc_tools/event_log_visualizer/analyzer.cc b/rtc_tools/event_log_visualizer/analyzer.cc
index 3bed18b..a8dbc30 100644
--- a/rtc_tools/event_log_visualizer/analyzer.cc
+++ b/rtc_tools/event_log_visualizer/analyzer.cc
@@ -130,7 +130,7 @@
// This is much more reliable for outgoing streams than for incoming streams.
template <typename RtpPacketContainer>
-rtc::Optional<uint32_t> EstimateRtpClockFrequency(
+absl::optional<uint32_t> EstimateRtpClockFrequency(
const RtpPacketContainer& packets,
int64_t end_time_us) {
RTC_CHECK(packets.size() >= 2);
@@ -151,7 +151,7 @@
<< "Failed to estimate RTP clock frequency: Stream too short. ("
<< packets.size() << " packets, "
<< last_log_timestamp - first_log_timestamp << " us)";
- return rtc::nullopt;
+ return absl::nullopt;
}
double duration =
static_cast<double>(last_log_timestamp - first_log_timestamp) /
@@ -166,7 +166,7 @@
RTC_LOG(LS_WARNING) << "Failed to estimate RTP clock frequency: Estimate "
<< estimated_frequency
<< "not close to any stardard RTP frequency.";
- return rtc::nullopt;
+ return absl::nullopt;
}
constexpr float kLeftMargin = 0.01f;
@@ -174,7 +174,7 @@
constexpr float kBottomMargin = 0.02f;
constexpr float kTopMargin = 0.05f;
-rtc::Optional<double> NetworkDelayDiff_AbsSendTime(
+absl::optional<double> NetworkDelayDiff_AbsSendTime(
const LoggedRtpPacket& old_packet,
const LoggedRtpPacket& new_packet) {
if (old_packet.header.extension.hasAbsoluteSendTime &&
@@ -188,11 +188,11 @@
recv_time_diff - AbsSendTimeToMicroseconds(send_time_diff);
return delay_change_us / 1000;
} else {
- return rtc::nullopt;
+ return absl::nullopt;
}
}
-rtc::Optional<double> NetworkDelayDiff_CaptureTime(
+absl::optional<double> NetworkDelayDiff_CaptureTime(
const LoggedRtpPacket& old_packet,
const LoggedRtpPacket& new_packet) {
int64_t send_time_diff = WrappingDifference(
@@ -230,13 +230,13 @@
// store the result in a TimeSeries.
template <typename DataType, typename IterableType>
void ProcessPoints(rtc::FunctionView<float(const DataType&)> fx,
- rtc::FunctionView<rtc::Optional<float>(const DataType&)> fy,
+ rtc::FunctionView<absl::optional<float>(const DataType&)> fy,
const IterableType& data_view,
TimeSeries* result) {
for (size_t i = 0; i < data_view.size(); i++) {
const DataType& elem = data_view[i];
float x = fx(elem);
- rtc::Optional<float> y = fy(elem);
+ absl::optional<float> y = fy(elem);
if (y)
result->points.emplace_back(x, *y);
}
@@ -248,13 +248,13 @@
template <typename DataType, typename ResultType, typename IterableType>
void ProcessPairs(
rtc::FunctionView<float(const DataType&)> fx,
- rtc::FunctionView<rtc::Optional<ResultType>(const DataType&,
- const DataType&)> fy,
+ rtc::FunctionView<absl::optional<ResultType>(const DataType&,
+ const DataType&)> fy,
const IterableType& data,
TimeSeries* result) {
for (size_t i = 1; i < data.size(); i++) {
float x = fx(data[i]);
- rtc::Optional<ResultType> y = fy(data[i - 1], data[i]);
+ absl::optional<ResultType> y = fy(data[i - 1], data[i]);
if (y)
result->points.emplace_back(x, static_cast<float>(*y));
}
@@ -266,14 +266,14 @@
template <typename DataType, typename ResultType, typename IterableType>
void AccumulatePairs(
rtc::FunctionView<float(const DataType&)> fx,
- rtc::FunctionView<rtc::Optional<ResultType>(const DataType&,
- const DataType&)> fy,
+ rtc::FunctionView<absl::optional<ResultType>(const DataType&,
+ const DataType&)> fy,
const IterableType& data,
TimeSeries* result) {
ResultType sum = 0;
for (size_t i = 1; i < data.size(); i++) {
float x = fx(data[i]);
- rtc::Optional<ResultType> y = fy(data[i - 1], data[i]);
+ absl::optional<ResultType> y = fy(data[i - 1], data[i]);
if (y)
sum += *y;
result->points.emplace_back(x, static_cast<float>(sum));
@@ -287,7 +287,7 @@
template <typename DataType, typename ResultType, typename IterableType>
void MovingAverage(
rtc::FunctionView<float(int64_t)> fx,
- rtc::FunctionView<rtc::Optional<ResultType>(const DataType&)> fy,
+ rtc::FunctionView<absl::optional<ResultType>(const DataType&)> fy,
const IterableType& data_view,
int64_t begin_time,
int64_t end_time,
@@ -301,7 +301,7 @@
for (int64_t t = begin_time; t < end_time + step; t += step) {
while (window_index_end < data_view.size() &&
data_view[window_index_end].log_time_us() < t) {
- rtc::Optional<ResultType> value = fy(data_view[window_index_end]);
+ absl::optional<ResultType> value = fy(data_view[window_index_end]);
if (value)
sum_in_window += *value;
++window_index_end;
@@ -309,7 +309,7 @@
while (window_index_begin < data_view.size() &&
data_view[window_index_begin].log_time_us() <
t - window_duration_us) {
- rtc::Optional<ResultType> value = fy(data_view[window_index_begin]);
+ absl::optional<ResultType> value = fy(data_view[window_index_begin]);
if (value)
sum_in_window -= *value;
++window_index_begin;
@@ -465,7 +465,7 @@
while (start_iter != log_start_events.end()) {
int64_t start = start_iter->log_time_us();
++start_iter;
- rtc::Optional<int64_t> next_start;
+ absl::optional<int64_t> next_start;
if (start_iter != log_start_events.end())
next_start.emplace(start_iter->log_time_us());
if (end_iter != log_end_events.end() &&
@@ -537,7 +537,7 @@
TimeSeries time_series(GetStreamName(direction, stream.ssrc),
LineStyle::kBar);
auto GetPacketSize = [](const LoggedRtpPacket& packet) {
- return rtc::Optional<float>(packet.total_length);
+ return absl::optional<float>(packet.total_length);
};
auto ToCallTime = [this](const LoggedRtpPacket& packet) {
return this->ToCallTimeSec(packet.log_time_us());
@@ -599,7 +599,7 @@
uint32_t ssrc = playout_stream.first;
if (!MatchingSsrc(ssrc, desired_ssrc_))
continue;
- rtc::Optional<int64_t> last_playout_ms;
+ absl::optional<int64_t> last_playout_ms;
TimeSeries time_series(SsrcToString(ssrc), LineStyle::kBar);
for (const auto& playout_event : playout_stream.second) {
float x = ToCallTimeSec(playout_event.log_time_us());
@@ -1139,7 +1139,7 @@
cc.OnTransportFeedback(rtcp_iterator->transport_feedback);
std::vector<PacketFeedback> feedback = cc.GetTransportFeedbackVector();
SortPacketFeedbackVector(&feedback);
- rtc::Optional<uint32_t> bitrate_bps;
+ absl::optional<uint32_t> bitrate_bps;
if (!feedback.empty()) {
#if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
acknowledged_bitrate_estimator.IncomingPacketFeedbackVector(feedback);
@@ -1251,7 +1251,7 @@
clock.TimeInMicroseconds());
rscc.OnReceivedPacket(arrival_time_ms, payload, packet.rtp.header);
acked_bitrate.Update(payload, arrival_time_ms);
- rtc::Optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
+ absl::optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
if (bitrate_bps) {
uint32_t y = *bitrate_bps / 1000;
float x = ToCallTimeSec(clock.TimeInMicroseconds());
@@ -1383,7 +1383,7 @@
int64_t end_time_us = log_segments_.empty()
? std::numeric_limits<int64_t>::max()
: log_segments_.front().second;
- rtc::Optional<uint32_t> estimated_frequency =
+ absl::optional<uint32_t> estimated_frequency =
EstimateRtpClockFrequency(packets, end_time_us);
if (!estimated_frequency)
continue;
@@ -1463,11 +1463,11 @@
TimeSeries time_series("Audio encoder target bitrate", LineStyle::kLine,
PointStyle::kHighlight);
auto GetAnaBitrateBps = [](const LoggedAudioNetworkAdaptationEvent& ana_event)
- -> rtc::Optional<float> {
+ -> absl::optional<float> {
if (ana_event.config.bitrate_bps)
- return rtc::Optional<float>(
+ return absl::optional<float>(
static_cast<float>(*ana_event.config.bitrate_bps));
- return rtc::nullopt;
+ return absl::nullopt;
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
@@ -1488,9 +1488,9 @@
auto GetAnaFrameLengthMs =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.frame_length_ms)
- return rtc::Optional<float>(
+ return absl::optional<float>(
static_cast<float>(*ana_event.config.frame_length_ms));
- return rtc::Optional<float>();
+ return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
@@ -1511,9 +1511,9 @@
auto GetAnaPacketLoss =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.uplink_packet_loss_fraction)
- return rtc::Optional<float>(static_cast<float>(
+ return absl::optional<float>(static_cast<float>(
*ana_event.config.uplink_packet_loss_fraction));
- return rtc::Optional<float>();
+ return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
@@ -1535,9 +1535,9 @@
auto GetAnaFecEnabled =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.enable_fec)
- return rtc::Optional<float>(
+ return absl::optional<float>(
static_cast<float>(*ana_event.config.enable_fec));
- return rtc::Optional<float>();
+ return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
@@ -1558,9 +1558,9 @@
auto GetAnaDtxEnabled =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.enable_dtx)
- return rtc::Optional<float>(
+ return absl::optional<float>(
static_cast<float>(*ana_event.config.enable_dtx));
- return rtc::Optional<float>();
+ return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
@@ -1581,9 +1581,9 @@
auto GetAnaNumChannels =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.num_channels)
- return rtc::Optional<float>(
+ return absl::optional<float>(
static_cast<float>(*ana_event.config.num_channels));
- return rtc::Optional<float>();
+ return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
@@ -1605,7 +1605,7 @@
// that outlive the one constructed.
NetEqStreamInput(const std::vector<LoggedRtpPacketIncoming>* packet_stream,
const std::vector<LoggedAudioPlayoutEvent>* output_events,
- rtc::Optional<int64_t> end_time_ms)
+ absl::optional<int64_t> end_time_ms)
: packet_stream_(*packet_stream),
packet_stream_it_(packet_stream_.begin()),
output_events_it_(output_events->begin()),
@@ -1615,22 +1615,22 @@
RTC_DCHECK(output_events);
}
- rtc::Optional<int64_t> NextPacketTime() const override {
+ absl::optional<int64_t> NextPacketTime() const override {
if (packet_stream_it_ == packet_stream_.end()) {
- return rtc::nullopt;
+ return absl::nullopt;
}
if (end_time_ms_ && packet_stream_it_->rtp.log_time_ms() > *end_time_ms_) {
- return rtc::nullopt;
+ return absl::nullopt;
}
return packet_stream_it_->rtp.log_time_ms();
}
- rtc::Optional<int64_t> NextOutputEventTime() const override {
+ absl::optional<int64_t> NextOutputEventTime() const override {
if (output_events_it_ == output_events_end_) {
- return rtc::nullopt;
+ return absl::nullopt;
}
if (end_time_ms_ && output_events_it_->log_time_ms() > *end_time_ms_) {
- return rtc::nullopt;
+ return absl::nullopt;
}
return output_events_it_->log_time_ms();
}
@@ -1661,9 +1661,9 @@
bool ended() const override { return !NextEventTime(); }
- rtc::Optional<RTPHeader> NextHeader() const override {
+ absl::optional<RTPHeader> NextHeader() const override {
if (packet_stream_it_ == packet_stream_.end()) {
- return rtc::nullopt;
+ return absl::nullopt;
}
return packet_stream_it_->rtp.header;
}
@@ -1673,7 +1673,7 @@
std::vector<LoggedRtpPacketIncoming>::const_iterator packet_stream_it_;
std::vector<LoggedAudioPlayoutEvent>::const_iterator output_events_it_;
const std::vector<LoggedAudioPlayoutEvent>::const_iterator output_events_end_;
- const rtc::Optional<int64_t> end_time_ms_;
+ const absl::optional<int64_t> end_time_ms_;
};
namespace {
@@ -1683,7 +1683,7 @@
std::unique_ptr<test::NetEqStatsGetter> CreateNetEqTestAndRun(
const std::vector<LoggedRtpPacketIncoming>* packet_stream,
const std::vector<LoggedAudioPlayoutEvent>* output_events,
- rtc::Optional<int64_t> end_time_ms,
+ absl::optional<int64_t> end_time_ms,
const std::string& replacement_file_name,
int file_sample_rate_hz) {
std::unique_ptr<test::NetEqInput> input(
@@ -1759,10 +1759,10 @@
output_events_it = parsed_log_.audio_playout_events().cbegin();
}
- rtc::Optional<int64_t> end_time_ms =
+ absl::optional<int64_t> end_time_ms =
log_segments_.empty()
- ? rtc::nullopt
- : rtc::Optional<int64_t>(log_segments_.front().second / 1000);
+ ? absl::nullopt
+ : absl::optional<int64_t>(log_segments_.front().second / 1000);
neteq_stats[ssrc] = CreateNetEqTestAndRun(
audio_packets, &output_events_it->second, end_time_ms,
@@ -1786,8 +1786,8 @@
std::vector<float> send_times_s;
std::vector<float> arrival_delay_ms;
std::vector<float> corrected_arrival_delay_ms;
- std::vector<rtc::Optional<float>> playout_delay_ms;
- std::vector<rtc::Optional<float>> target_delay_ms;
+ std::vector<absl::optional<float>> playout_delay_ms;
+ std::vector<absl::optional<float>> target_delay_ms;
neteq_stats.at(ssrc)->delay_analyzer()->CreateGraphs(
&send_times_s, &arrival_delay_ms, &corrected_arrival_delay_ms,
&playout_delay_ms, &target_delay_ms);
@@ -2014,9 +2014,9 @@
: 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 @@
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 @@
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) {
diff --git a/rtc_tools/network_tester/BUILD.gn b/rtc_tools/network_tester/BUILD.gn
index 67c9357..544f122 100644
--- a/rtc_tools/network_tester/BUILD.gn
+++ b/rtc_tools/network_tester/BUILD.gn
@@ -41,13 +41,13 @@
deps = [
":network_tester_config_proto",
":network_tester_packet_proto",
- "../../api:optional",
"../../p2p",
"../../rtc_base:checks",
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_task_queue",
"../../rtc_base:sequenced_task_checker",
+ "//third_party/abseil-cpp/absl/types:optional",
]
if (!build_with_chromium && is_clang) {
diff --git a/rtc_tools/network_tester/config_reader.cc b/rtc_tools/network_tester/config_reader.cc
index 5ef1e4b..8ef88c7 100644
--- a/rtc_tools/network_tester/config_reader.cc
+++ b/rtc_tools/network_tester/config_reader.cc
@@ -31,10 +31,10 @@
ConfigReader::~ConfigReader() = default;
-rtc::Optional<ConfigReader::Config> ConfigReader::GetNextConfig() {
+absl::optional<ConfigReader::Config> ConfigReader::GetNextConfig() {
#ifdef WEBRTC_NETWORK_TESTER_PROTO
if (proto_config_index_ >= proto_all_configs_.configs_size())
- return rtc::nullopt;
+ return absl::nullopt;
auto proto_config = proto_all_configs_.configs(proto_config_index_++);
RTC_DCHECK(proto_config.has_packet_send_interval_ms());
RTC_DCHECK(proto_config.has_packet_size());
@@ -45,7 +45,7 @@
config.execution_time_ms = proto_config.execution_time_ms();
return config;
#else
- return rtc::nullopt;
+ return absl::nullopt;
#endif // WEBRTC_NETWORK_TESTER_PROTO
}
diff --git a/rtc_tools/network_tester/config_reader.h b/rtc_tools/network_tester/config_reader.h
index e03317f..deee245 100644
--- a/rtc_tools/network_tester/config_reader.h
+++ b/rtc_tools/network_tester/config_reader.h
@@ -14,7 +14,7 @@
#include <fstream>
#include <string>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/ignore_wundef.h"
@@ -40,7 +40,7 @@
explicit ConfigReader(const std::string& config_file_path);
~ConfigReader();
- rtc::Optional<Config> GetNextConfig();
+ absl::optional<Config> GetNextConfig();
private:
NetworkTesterAllConfigs proto_all_configs_;
diff --git a/rtc_tools/network_tester/test_controller.cc b/rtc_tools/network_tester/test_controller.cc
index a3a8833..e5bd92e 100644
--- a/rtc_tools/network_tester/test_controller.cc
+++ b/rtc_tools/network_tester/test_controller.cc
@@ -37,7 +37,7 @@
udp_transport_->SetRemoteAddress(rtc::SocketAddress(hostname, port));
NetworkTesterPacket packet;
packet.set_type(NetworkTesterPacket::HAND_SHAKING);
- SendData(packet, rtc::nullopt);
+ SendData(packet, absl::nullopt);
rtc::CritScope scoped_lock(&local_test_done_lock_);
local_test_done_ = false;
remote_test_done_ = false;
@@ -49,7 +49,7 @@
}
void TestController::SendData(const NetworkTesterPacket& packet,
- rtc::Optional<size_t> data_size) {
+ absl::optional<size_t> data_size) {
// Can be call from packet_sender or from test_controller thread.
size_t packet_size = packet.ByteSizeLong();
send_data_[0] = packet_size;
@@ -65,7 +65,7 @@
RTC_DCHECK_CALLED_SEQUENTIALLY(&packet_sender_checker_);
NetworkTesterPacket packet;
packet.set_type(NetworkTesterPacket::TEST_DONE);
- SendData(packet, rtc::nullopt);
+ SendData(packet, absl::nullopt);
rtc::CritScope scoped_lock(&local_test_done_lock_);
local_test_done_ = true;
}
@@ -92,7 +92,7 @@
NetworkTesterPacket packet;
packet.set_type(NetworkTesterPacket::TEST_START);
udp_transport_->SetRemoteAddress(remote_addr);
- SendData(packet, rtc::nullopt);
+ SendData(packet, absl::nullopt);
packet_sender_.reset(new PacketSender(this, config_file_path_));
packet_sender_->StartSending();
rtc::CritScope scoped_lock(&local_test_done_lock_);
diff --git a/rtc_tools/network_tester/test_controller.h b/rtc_tools/network_tester/test_controller.h
index 4c9ede1..cf65e17 100644
--- a/rtc_tools/network_tester/test_controller.h
+++ b/rtc_tools/network_tester/test_controller.h
@@ -49,7 +49,7 @@
void SendConnectTo(const std::string& hostname, int port);
void SendData(const NetworkTesterPacket& packet,
- rtc::Optional<size_t> data_size);
+ absl::optional<size_t> data_size);
void OnTestDone();
diff --git a/test/BUILD.gn b/test/BUILD.gn
index 8da8e19..60d725b 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -63,7 +63,6 @@
"..:webrtc_common",
"../:typedefs",
"../api:libjingle_peerconnection_api",
- "../api:optional",
"../api/video:video_frame",
"../api/video:video_frame_i420",
"../api/video_codecs:video_codecs_api",
@@ -75,6 +74,7 @@
"../rtc_base:rtc_base",
"../rtc_base:rtc_task_queue",
"../system_wrappers",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -389,9 +389,9 @@
deps = [
"..:webrtc_common",
"../:typedefs",
- "../api:optional",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
if (is_ios) {
deps += [ ":fileutils_objc" ]
@@ -453,10 +453,10 @@
deps = [
":fileutils",
":test_support",
- "../api:optional",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"//testing/gtest",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/test/call_test.cc b/test/call_test.cc
index c111b11..789f775 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -323,7 +323,7 @@
int width,
int height) {
frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
- width, height, rtc::nullopt, rtc::nullopt, framerate * speed, clock));
+ width, height, absl::nullopt, absl::nullopt, framerate * speed, clock));
video_send_stream_->SetSource(frame_generator_capturer_.get(),
DegradationPreference::MAINTAIN_FRAMERATE);
}
@@ -332,7 +332,7 @@
int width,
int height) {
frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
- width, height, rtc::nullopt, rtc::nullopt, framerate, clock_));
+ width, height, absl::nullopt, absl::nullopt, framerate, clock_));
video_send_stream_->SetSource(frame_generator_capturer_.get(),
DegradationPreference::MAINTAIN_FRAMERATE);
}
diff --git a/test/frame_generator.cc b/test/frame_generator.cc
index 040192d..cc8e6c0 100644
--- a/test/frame_generator.cc
+++ b/test/frame_generator.cc
@@ -402,7 +402,7 @@
size_t current_frame_num_;
VideoFrame* current_source_frame_;
- rtc::Optional<VideoFrame> current_frame_;
+ absl::optional<VideoFrame> current_frame_;
YuvFileGenerator file_generator_;
};
@@ -444,8 +444,8 @@
std::unique_ptr<FrameGenerator> FrameGenerator::CreateSquareGenerator(
int width,
int height,
- rtc::Optional<OutputType> type,
- rtc::Optional<int> num_squares) {
+ absl::optional<OutputType> type,
+ absl::optional<int> num_squares) {
return std::unique_ptr<FrameGenerator>(
new SquareGenerator(width, height, type.value_or(OutputType::I420),
num_squares.value_or(10)));
diff --git a/test/frame_generator.h b/test/frame_generator.h
index 049b5e7..65857af 100644
--- a/test/frame_generator.h
+++ b/test/frame_generator.h
@@ -67,8 +67,8 @@
static std::unique_ptr<FrameGenerator> CreateSquareGenerator(
int width,
int height,
- rtc::Optional<OutputType> type,
- rtc::Optional<int> num_squares);
+ absl::optional<OutputType> type,
+ absl::optional<int> num_squares);
// Creates a frame generator that repeatedly plays a set of yuv files.
// The frame_repeat_count determines how many times each frame is shown,
diff --git a/test/frame_generator_capturer.cc b/test/frame_generator_capturer.cc
index a9e80d7..61f96fa 100644
--- a/test/frame_generator_capturer.cc
+++ b/test/frame_generator_capturer.cc
@@ -87,8 +87,8 @@
FrameGeneratorCapturer* FrameGeneratorCapturer::Create(
int width,
int height,
- rtc::Optional<FrameGenerator::OutputType> type,
- rtc::Optional<int> num_squares,
+ absl::optional<FrameGenerator::OutputType> type,
+ absl::optional<int> num_squares,
int target_fps,
Clock* clock) {
std::unique_ptr<FrameGeneratorCapturer> capturer(new FrameGeneratorCapturer(
@@ -187,7 +187,7 @@
}
if (sink_) {
- rtc::Optional<VideoFrame> out_frame = AdaptFrame(*frame);
+ absl::optional<VideoFrame> out_frame = AdaptFrame(*frame);
if (out_frame)
sink_->OnFrame(*out_frame);
}
diff --git a/test/frame_generator_capturer.h b/test/frame_generator_capturer.h
index e135399..1aecf49 100644
--- a/test/frame_generator_capturer.h
+++ b/test/frame_generator_capturer.h
@@ -44,8 +44,8 @@
static FrameGeneratorCapturer* Create(
int width,
int height,
- rtc::Optional<FrameGenerator::OutputType> type,
- rtc::Optional<int> num_squares,
+ absl::optional<FrameGenerator::OutputType> type,
+ absl::optional<int> num_squares,
int target_fps,
Clock* clock);
@@ -98,7 +98,7 @@
std::unique_ptr<FrameGenerator> frame_generator_;
int target_fps_ RTC_GUARDED_BY(&lock_);
- rtc::Optional<int> wanted_fps_ RTC_GUARDED_BY(&lock_);
+ absl::optional<int> wanted_fps_ RTC_GUARDED_BY(&lock_);
VideoRotation fake_rotation_ = kVideoRotation_0;
int64_t first_frame_capture_time_;
diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index 6375dc3..684ce9a 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -260,11 +260,11 @@
deps = [
"../..:webrtc_common",
"../../:typedefs",
- "../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../modules/rtp_rtcp:rtp_rtcp_format",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -454,11 +454,11 @@
]
deps = [
":fuzz_data_helper",
- "../../api:optional",
"../../api/audio:audio_frame_api",
"../../modules/audio_processing",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/test/fuzzers/audio_decoder_fuzzer.cc b/test/fuzzers/audio_decoder_fuzzer.cc
index f6ac4cb..40a7315 100644
--- a/test/fuzzers/audio_decoder_fuzzer.cc
+++ b/test/fuzzers/audio_decoder_fuzzer.cc
@@ -12,8 +12,8 @@
#include <limits>
+#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder.h"
-#include "api/optional.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "rtc_base/checks.h"
diff --git a/test/fuzzers/neteq_rtp_fuzzer.cc b/test/fuzzers/neteq_rtp_fuzzer.cc
index 73cbda2..e28af90 100644
--- a/test/fuzzers/neteq_rtp_fuzzer.cc
+++ b/test/fuzzers/neteq_rtp_fuzzer.cc
@@ -65,11 +65,11 @@
FuzzHeader();
}
- rtc::Optional<int64_t> NextPacketTime() const override {
+ absl::optional<int64_t> NextPacketTime() const override {
return packet_->time_ms;
}
- rtc::Optional<int64_t> NextOutputEventTime() const override {
+ absl::optional<int64_t> NextOutputEventTime() const override {
return input_->NextOutputEventTime();
}
@@ -85,7 +85,7 @@
bool ended() const override { return ended_; }
- rtc::Optional<RTPHeader> NextHeader() const override {
+ absl::optional<RTPHeader> NextHeader() const override {
RTC_DCHECK(packet_);
return packet_->header;
}
diff --git a/test/fuzzers/neteq_signal_fuzzer.cc b/test/fuzzers/neteq_signal_fuzzer.cc
index 981ba28..611964d 100644
--- a/test/fuzzers/neteq_signal_fuzzer.cc
+++ b/test/fuzzers/neteq_signal_fuzzer.cc
@@ -88,11 +88,11 @@
output_event_period_ms_ = fuzz_data_.SelectOneOf(output_event_periods);
}
- rtc::Optional<int64_t> NextPacketTime() const override {
+ absl::optional<int64_t> NextPacketTime() const override {
return packet_->time_ms;
}
- rtc::Optional<int64_t> NextOutputEventTime() const override {
+ absl::optional<int64_t> NextOutputEventTime() const override {
return next_output_event_ms_;
}
@@ -124,7 +124,7 @@
bool ended() const override { return ended_; }
- rtc::Optional<RTPHeader> NextHeader() const override {
+ absl::optional<RTPHeader> NextHeader() const override {
RTC_DCHECK(packet_);
return packet_->header;
}
diff --git a/test/mock_audio_decoder_factory.h b/test/mock_audio_decoder_factory.h
index 3a43997..247f9fa 100644
--- a/test/mock_audio_decoder_factory.h
+++ b/test/mock_audio_decoder_factory.h
@@ -28,14 +28,14 @@
MOCK_METHOD1(IsSupportedDecoder, bool(const SdpAudioFormat&));
std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id) {
+ absl::optional<AudioCodecPairId> codec_pair_id) {
std::unique_ptr<AudioDecoder> return_value;
MakeAudioDecoderMock(format, codec_pair_id, &return_value);
return return_value;
}
MOCK_METHOD3(MakeAudioDecoderMock,
void(const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id,
+ absl::optional<AudioCodecPairId> codec_pair_id,
std::unique_ptr<AudioDecoder>* return_value));
// Creates a MockAudioDecoderFactory with no formats and that may not be
diff --git a/test/mock_audio_encoder.h b/test/mock_audio_encoder.h
index 7154e64..60425e0 100644
--- a/test/mock_audio_encoder.h
+++ b/test/mock_audio_encoder.h
@@ -43,7 +43,7 @@
MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes));
MOCK_METHOD2(OnReceivedUplinkBandwidth,
void(int target_audio_bitrate_bps,
- rtc::Optional<int64_t> probing_interval_ms));
+ absl::optional<int64_t> probing_interval_ms));
MOCK_METHOD1(OnReceivedUplinkPacketLossFraction,
void(float uplink_packet_loss_fraction));
diff --git a/test/mock_audio_encoder_factory.h b/test/mock_audio_encoder_factory.h
index 3eaa3b9..340602c 100644
--- a/test/mock_audio_encoder_factory.h
+++ b/test/mock_audio_encoder_factory.h
@@ -25,12 +25,12 @@
public:
MOCK_METHOD0(GetSupportedEncoders, std::vector<AudioCodecSpec>());
MOCK_METHOD1(QueryAudioEncoder,
- rtc::Optional<AudioCodecInfo>(const SdpAudioFormat& format));
+ absl::optional<AudioCodecInfo>(const SdpAudioFormat& format));
std::unique_ptr<AudioEncoder> MakeAudioEncoder(
int payload_type,
const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id) {
+ absl::optional<AudioCodecPairId> codec_pair_id) {
std::unique_ptr<AudioEncoder> return_value;
MakeAudioEncoderMock(payload_type, format, codec_pair_id, &return_value);
return return_value;
@@ -38,7 +38,7 @@
MOCK_METHOD4(MakeAudioEncoderMock,
void(int payload_type,
const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id,
+ absl::optional<AudioCodecPairId> codec_pair_id,
std::unique_ptr<AudioEncoder>* return_value));
// Creates a MockAudioEncoderFactory with no formats and that may not be
@@ -55,7 +55,7 @@
ON_CALL(*factory.get(), GetSupportedEncoders())
.WillByDefault(Return(std::vector<webrtc::AudioCodecSpec>()));
ON_CALL(*factory.get(), QueryAudioEncoder(_))
- .WillByDefault(Return(rtc::nullopt));
+ .WillByDefault(Return(absl::nullopt));
EXPECT_CALL(*factory.get(), GetSupportedEncoders()).Times(AnyNumber());
EXPECT_CALL(*factory.get(), QueryAudioEncoder(_)).Times(AnyNumber());
@@ -78,7 +78,7 @@
ON_CALL(*factory.get(), GetSupportedEncoders())
.WillByDefault(Return(std::vector<webrtc::AudioCodecSpec>()));
ON_CALL(*factory.get(), QueryAudioEncoder(_))
- .WillByDefault(Return(rtc::nullopt));
+ .WillByDefault(Return(absl::nullopt));
ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _, _))
.WillByDefault(SetArgPointee<3>(nullptr));
diff --git a/test/testsupport/fileutils.cc b/test/testsupport/fileutils.cc
index 7ba24c1..76a635a 100644
--- a/test/testsupport/fileutils.cc
+++ b/test/testsupport/fileutils.cc
@@ -223,9 +223,9 @@
return filename;
}
-rtc::Optional<std::vector<std::string>> ReadDirectory(std::string path) {
+absl::optional<std::vector<std::string>> ReadDirectory(std::string path) {
if (path.length() == 0)
- return rtc::Optional<std::vector<std::string>>();
+ return absl::optional<std::vector<std::string>>();
#if defined(WEBRTC_WIN)
// Append separator character if needed.
@@ -236,7 +236,7 @@
WIN32_FIND_DATA data;
HANDLE handle = ::FindFirstFile(rtc::ToUtf16(path + '*').c_str(), &data);
if (handle == INVALID_HANDLE_VALUE)
- return rtc::Optional<std::vector<std::string>>();
+ return absl::optional<std::vector<std::string>>();
// Populate output.
std::vector<std::string> found_entries;
@@ -257,7 +257,7 @@
// Init.
DIR* dir = ::opendir(path.c_str());
if (dir == nullptr)
- return rtc::Optional<std::vector<std::string>>();
+ return absl::optional<std::vector<std::string>>();
// Populate output.
std::vector<std::string> found_entries;
@@ -271,7 +271,7 @@
closedir(dir);
#endif
- return rtc::Optional<std::vector<std::string>>(std::move(found_entries));
+ return absl::optional<std::vector<std::string>>(std::move(found_entries));
}
bool CreateDir(const std::string& directory_name) {
diff --git a/test/testsupport/fileutils.h b/test/testsupport/fileutils.h
index b1eec8b..af39a94 100644
--- a/test/testsupport/fileutils.h
+++ b/test/testsupport/fileutils.h
@@ -16,7 +16,7 @@
#include <string>
#include <vector>
-#include "api/optional.h"
+#include "absl/types/optional.h"
namespace webrtc {
namespace test {
@@ -81,7 +81,7 @@
// of strings with one element for each found file or directory. Each element is
// a path created by prepending |dir| to the file/directory name. "." and ".."
// are never added in the returned vector.
-rtc::Optional<std::vector<std::string>> ReadDirectory(std::string path);
+absl::optional<std::vector<std::string>> ReadDirectory(std::string path);
// Creates a directory if it not already exists.
// Returns true if successful. Will print an error message to stderr and return
diff --git a/test/testsupport/fileutils_unittest.cc b/test/testsupport/fileutils_unittest.cc
index 7f85135..b39be6f 100644
--- a/test/testsupport/fileutils_unittest.cc
+++ b/test/testsupport/fileutils_unittest.cc
@@ -17,7 +17,7 @@
#include <list>
#include <string>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "test/gtest.h"
@@ -41,7 +41,7 @@
void CleanDir(const std::string& dir, size_t* num_deleted_entries) {
RTC_DCHECK(num_deleted_entries);
*num_deleted_entries = 0;
- rtc::Optional<std::vector<std::string>> dir_content = ReadDirectory(dir);
+ absl::optional<std::vector<std::string>> dir_content = ReadDirectory(dir);
EXPECT_TRUE(dir_content);
for (const auto& entry : *dir_content) {
if (DirExists(entry)) {
@@ -238,7 +238,7 @@
EXPECT_TRUE(DirExists(temp_subdir));
// Checks.
- rtc::Optional<std::vector<std::string>> dir_content =
+ absl::optional<std::vector<std::string>> dir_content =
ReadDirectory(temp_directory);
EXPECT_TRUE(dir_content);
EXPECT_EQ(2u, dir_content->size());
diff --git a/test/vcm_capturer.cc b/test/vcm_capturer.cc
index 82b5cbc..22631ac 100644
--- a/test/vcm_capturer.cc
+++ b/test/vcm_capturer.cc
@@ -108,7 +108,7 @@
void VcmCapturer::OnFrame(const VideoFrame& frame) {
rtc::CritScope lock(&crit_);
if (started_ && sink_) {
- rtc::Optional<VideoFrame> out_frame = AdaptFrame(frame);
+ absl::optional<VideoFrame> out_frame = AdaptFrame(frame);
if (out_frame)
sink_->OnFrame(*out_frame);
}
diff --git a/test/video_capturer.cc b/test/video_capturer.cc
index c81c9f8..4faf449 100644
--- a/test/video_capturer.cc
+++ b/test/video_capturer.cc
@@ -17,7 +17,7 @@
VideoCapturer::VideoCapturer() : video_adapter_(new cricket::VideoAdapter()) {}
VideoCapturer::~VideoCapturer() {}
-rtc::Optional<VideoFrame> VideoCapturer::AdaptFrame(const VideoFrame& frame) {
+absl::optional<VideoFrame> VideoCapturer::AdaptFrame(const VideoFrame& frame) {
int cropped_width = 0;
int cropped_height = 0;
int out_width = 0;
@@ -27,10 +27,10 @@
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::nullopt;
+ return absl::nullopt;
}
- rtc::Optional<VideoFrame> out_frame;
+ absl::optional<VideoFrame> out_frame;
if (out_height != frame.height() || out_width != frame.width()) {
// Video adapter has requested a down-scale. Allocate a new buffer and
// return scaled version.
diff --git a/test/video_capturer.h b/test/video_capturer.h
index 63e1cd8..d117b96 100644
--- a/test/video_capturer.h
+++ b/test/video_capturer.h
@@ -14,7 +14,7 @@
#include <memory>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "api/video/i420_buffer.h"
#include "api/video/video_frame.h"
#include "api/video/video_source_interface.h"
@@ -41,7 +41,7 @@
const rtc::VideoSinkWants& wants) override;
protected:
- rtc::Optional<VideoFrame> AdaptFrame(const VideoFrame& frame);
+ absl::optional<VideoFrame> AdaptFrame(const VideoFrame& frame);
rtc::VideoSinkWants GetSinkWants();
private:
