Replacing SequencedTaskChecker with SequenceChecker.
Bug: webrtc:9883
Change-Id: I5e3189da2a46e6f4ed1a3c5a5dfd2f7d75a16b5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130961
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27518}diff --git a/call/call.cc b/call/call.cc
index 38a4dd5..84f29d8 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -50,9 +50,9 @@
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
-#include "rtc_base/sequenced_task_checker.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/synchronization/rw_lock_wrapper.h"
+#include "rtc_base/synchronization/sequence_checker.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
@@ -285,7 +285,7 @@
const std::unique_ptr<CallStats> call_stats_;
const std::unique_ptr<BitrateAllocator> bitrate_allocator_;
Call::Config config_;
- rtc::SequencedTaskChecker configuration_sequence_checker_;
+ SequenceChecker configuration_sequence_checker_;
NetworkState audio_network_state_;
NetworkState video_network_state_;
@@ -493,7 +493,7 @@
}
Call::~Call() {
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
RTC_CHECK(audio_send_ssrcs_.empty());
RTC_CHECK(video_send_ssrcs_.empty());
@@ -698,14 +698,14 @@
}
PacketReceiver* Call::Receiver() {
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
return this;
}
webrtc::AudioSendStream* Call::CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
RTC_DCHECK(media_transport() == config.media_transport);
@@ -747,7 +747,7 @@
void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyAudioSendStream");
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
RTC_DCHECK(send_stream != nullptr);
send_stream->Stop();
@@ -776,7 +776,7 @@
webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
const webrtc::AudioReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
RegisterRateObserver();
event_log_->Log(absl::make_unique<RtcEventAudioReceiveStreamConfig>(
CreateRtcLogStreamConfig(config)));
@@ -806,7 +806,7 @@
void Call::DestroyAudioReceiveStream(
webrtc::AudioReceiveStream* receive_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream");
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
RTC_DCHECK(receive_stream != nullptr);
webrtc::internal::AudioReceiveStream* audio_receive_stream =
static_cast<webrtc::internal::AudioReceiveStream*>(receive_stream);
@@ -836,7 +836,7 @@
VideoEncoderConfig encoder_config,
std::unique_ptr<FecController> fec_controller) {
TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
RTC_DCHECK(media_transport() == config.media_transport);
@@ -891,7 +891,7 @@
void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
RTC_DCHECK(send_stream != nullptr);
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
send_stream->Stop();
@@ -929,7 +929,7 @@
webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
webrtc::VideoReceiveStream::Config configuration) {
TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
receive_side_cc_.SetSendPeriodicFeedback(
SendPeriodicFeedback(configuration.rtp.extensions));
@@ -967,7 +967,7 @@
void Call::DestroyVideoReceiveStream(
webrtc::VideoReceiveStream* receive_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
RTC_DCHECK(receive_stream != nullptr);
VideoReceiveStream* receive_stream_impl =
static_cast<VideoReceiveStream*>(receive_stream);
@@ -994,7 +994,7 @@
FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
const FlexfecReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateFlexfecReceiveStream");
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
RecoveredPacketReceiver* recovered_packet_receiver = this;
@@ -1026,7 +1026,7 @@
void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyFlexfecReceiveStream");
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
RTC_DCHECK(receive_stream != nullptr);
{
@@ -1052,7 +1052,7 @@
Call::Stats Call::GetStats() const {
// TODO(solenberg): Some test cases in EndToEndTest use this from a different
// thread. Re-enable once that is fixed.
- // RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ // RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
Stats stats;
// Fetch available send/receive bitrates.
std::vector<unsigned int> ssrcs;
@@ -1101,7 +1101,7 @@
}
void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
switch (media) {
case MediaType::AUDIO:
audio_network_state_ = state;
@@ -1141,7 +1141,7 @@
}
void Call::UpdateAggregateNetworkState() {
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
bool have_audio = false;
bool have_video = false;
@@ -1453,7 +1453,7 @@
MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
- RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+ RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
if (RtpHeaderParser::IsRtcp(packet.cdata(), packet.size()))
return DeliverRtcp(media_type, packet.cdata(), packet.size());
