Revert "Use field clock instead of global clock where available"
This reverts commit f4453c3711eff9e8851364687cc55a31de5c059f.
Reason for revert: Broken downstream
Bug: webrtc:42223992
Original change's description:
> Use field clock instead of global clock where available
>
> Bug: webrtc:42223992
> Change-Id: Ia0eb6a5f9ebfe4031fc72026f31767b2ce7dcc2d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/404280
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#45333}
Bug: webrtc:42223992
Change-Id: I5c496d744b977edcda80a1877f9bae1f5864af2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/404581
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#45352}
diff --git a/api/units/timestamp.h b/api/units/timestamp.h
index 0317e9f..7831bba 100644
--- a/api/units/timestamp.h
+++ b/api/units/timestamp.h
@@ -60,10 +60,6 @@
constexpr T us() const {
return ToValue<T>();
}
- template <typename T = int64_t>
- constexpr T ns() const {
- return ToMultiple<1000, T>();
- }
constexpr int64_t seconds_or(int64_t fallback_value) const {
return ToFractionOr<1000000>(fallback_value);
diff --git a/api/units/timestamp_unittest.cc b/api/units/timestamp_unittest.cc
index c9254c2..1b88cb7 100644
--- a/api/units/timestamp_unittest.cc
+++ b/api/units/timestamp_unittest.cc
@@ -37,7 +37,6 @@
EXPECT_EQ(kTimestampSeconds.seconds(), kValue);
EXPECT_EQ(kTimestampMs.ms(), kValue);
EXPECT_EQ(kTimestampUs.us(), kValue);
- EXPECT_EQ(kTimestampUs.ns(), kValue * 1000);
}
TEST(TimestampTest, GetBackSameValues) {
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 90d3c70..eda5824 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -78,6 +78,7 @@
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/thread_annotations.h"
+#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/metrics.h"
#include "system_wrappers/include/ntp_time.h"
@@ -657,7 +658,7 @@
void ChannelReceive::OnRtpPacket(const RtpPacketReceived& packet) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
- Timestamp now = env_.clock().CurrentTime();
+ Timestamp now = Timestamp::Millis(TimeMillis());
last_received_rtp_timestamp_ = packet.Timestamp();
last_received_rtp_system_time_ = now;
@@ -754,7 +755,7 @@
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
// Store playout timestamp for the received RTCP packet
- UpdatePlayoutTimestamp(true, env_.clock().CurrentTime());
+ UpdatePlayoutTimestamp(true, Timestamp::Millis(TimeMillis()));
// Deliver RTCP packet to RTP/RTCP module for parsing
rtp_rtcp_->IncomingRtcpPacket(MakeArrayView(data, length));
diff --git a/audio/voip/voip_core.cc b/audio/voip/voip_core.cc
index d3ce340..7676811 100644
--- a/audio/voip/voip_core.cc
+++ b/audio/voip/voip_core.cc
@@ -40,6 +40,7 @@
#include "rtc_base/logging.h"
#include "rtc_base/random.h"
#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/time_utils.h"
namespace webrtc {
@@ -151,7 +152,7 @@
// Set local ssrc to random if not set by caller.
if (!local_ssrc) {
- Random random(env_.clock().TimeInMicroseconds());
+ Random random(TimeMicros());
local_ssrc = random.Rand<uint32_t>();
}
diff --git a/modules/audio_device/test_audio_device_impl.cc b/modules/audio_device/test_audio_device_impl.cc
index 80402d3..fe3d10d 100644
--- a/modules/audio_device/test_audio_device_impl.cc
+++ b/modules/audio_device/test_audio_device_impl.cc
@@ -25,6 +25,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/time_utils.h"
namespace webrtc {
namespace {
@@ -174,7 +175,7 @@
audio_buffer_->SetRecordedBuffer(
recording_buffer_.data(),
recording_buffer_.size() / capturer_->NumChannels(),
- std::make_optional(env_.clock().CurrentTime().ns()));
+ std::make_optional(TimeNanos()));
audio_buffer_->DeliverRecordedData();
}
if (!keep_capturing) {
diff --git a/p2p/base/port.cc b/p2p/base/port.cc
index 033e3f6..5a43cc7 100644
--- a/p2p/base/port.cc
+++ b/p2p/base/port.cc
@@ -53,6 +53,7 @@
#include "rtc_base/string_encode.h"
#include "rtc_base/string_utils.h"
#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "rtc_base/weak_ptr.h"
@@ -829,8 +830,7 @@
bool dead =
(state_ == State::INIT || state_ == State::PRUNED) &&
connections_.empty() &&
- env_.clock().TimeInMilliseconds() - last_time_all_connections_removed_ >=
- timeout_delay_;
+ TimeMillis() - last_time_all_connections_removed_ >= timeout_delay_;
if (dead) {
Destroy();
}
@@ -899,7 +899,7 @@
// fails and is removed before kPortTimeoutDelay, then this message will
// not cause the Port to be destroyed.
if (connections_.empty()) {
- last_time_all_connections_removed_ = env_.clock().TimeInMilliseconds();
+ last_time_all_connections_removed_ = TimeMillis();
PostDestroyIfDead(/*delayed=*/true);
}
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index ce0954a..be90539 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -1207,7 +1207,7 @@
requests_.push_back(std::move(request));
// "Now" using a monotonically increasing timer.
- int64_t cache_now_us = env_.clock().TimeInMicroseconds();
+ int64_t cache_now_us = TimeMicros();
if (cached_report_ &&
cache_now_us - cache_timestamp_us_ <= cache_lifetime_us_) {
// We have a fresh cached report to deliver. Deliver asynchronously, since
diff --git a/video/adaptation/overuse_frame_detector.cc b/video/adaptation/overuse_frame_detector.cc
index 827b0a5..959dbb0 100644
--- a/video/adaptation/overuse_frame_detector.cc
+++ b/video/adaptation/overuse_frame_detector.cc
@@ -600,7 +600,7 @@
!encode_usage_percent_)
return;
- int64_t now_ms = env_.clock().TimeInMilliseconds();
+ int64_t now_ms = TimeMillis();
const char* action = "NoAction";
if (IsOverusing(*encode_usage_percent_)) {
diff --git a/video/encoder_bitrate_adjuster.cc b/video/encoder_bitrate_adjuster.cc
index 1d17e10..d9d64e4 100644
--- a/video/encoder_bitrate_adjuster.cc
+++ b/video/encoder_bitrate_adjuster.cc
@@ -31,6 +31,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/experiments/rate_control_settings.h"
#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
#include "system_wrappers/include/clock.h"
#include "video/encoder_overshoot_detector.h"
#include "video/rate_utilization_tracker.h"
@@ -385,7 +386,7 @@
// Detectors may not exist, for instance if ScreenshareLayers is used.
auto& detector = overshoot_detectors_[stream_index][temporal_index];
if (detector) {
- detector->OnEncodedFrame(size.bytes(), clock_.TimeInMilliseconds());
+ detector->OnEncodedFrame(size.bytes(), TimeMillis());
}
if (media_rate_trackers_[stream_index]) {
media_rate_trackers_[stream_index]->OnDataProduced(size,
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index eccec79..4dff356 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -81,6 +81,7 @@
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/file_wrapper.h"
+#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/ntp_time.h"
@@ -586,7 +587,7 @@
char filename_buffer[256];
SimpleStringBuilder ssb(filename_buffer);
ssb << decoded_output_file << "/webrtc_receive_stream_" << remote_ssrc()
- << "-" << env_.clock().TimeInMicroseconds() << ".ivf";
+ << "-" << TimeMicros() << ".ivf";
video_decoder = CreateFrameDumpingDecoderWrapper(
std::move(video_decoder), FileWrapper::OpenWriteOnly(ssb.str()));
}
