Change ThreadChecker to SequencedTaskChecker in internal::Call
In preparation of running DirectTransport on a TaskQueue in unit-tests, change the thread-checkers to sequence-checkers. This is necessary for Mac and iOS, where the TaskQueue is guaranteed to run sequentially, but not guaranteed to do so on only one thread.
TODO: Add the relevant BUGs.
BUG=None
Review-Url: https://codereview.webrtc.org/2999973002
Cr-Commit-Position: refs/heads/master@{#19404}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index ff5ebca..1b5bc45 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -44,9 +44,9 @@
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/optional.h"
#include "webrtc/rtc_base/ptr_util.h"
+#include "webrtc/rtc_base/sequenced_task_checker.h"
#include "webrtc/rtc_base/task_queue.h"
#include "webrtc/rtc_base/thread_annotations.h"
-#include "webrtc/rtc_base/thread_checker.h"
#include "webrtc/rtc_base/trace_event.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/cpu_info.h"
@@ -259,7 +259,7 @@
const std::unique_ptr<CallStats> call_stats_;
const std::unique_ptr<BitrateAllocator> bitrate_allocator_;
Call::Config config_;
- rtc::ThreadChecker configuration_thread_checker_;
+ rtc::SequencedTaskChecker configuration_sequence_checker_;
NetworkState audio_network_state_;
NetworkState video_network_state_;
@@ -312,9 +312,9 @@
using RtpStateMap = std::map<uint32_t, RtpState>;
RtpStateMap suspended_audio_send_ssrcs_
- GUARDED_BY(configuration_thread_checker_);
+ GUARDED_BY(configuration_sequence_checker_);
RtpStateMap suspended_video_send_ssrcs_
- GUARDED_BY(configuration_thread_checker_);
+ GUARDED_BY(configuration_sequence_checker_);
webrtc::RtcEventLog* event_log_;
@@ -452,7 +452,7 @@
}
Call::~Call() {
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RTC_CHECK(audio_send_ssrcs_.empty());
RTC_CHECK(video_send_ssrcs_.empty());
@@ -594,14 +594,14 @@
PacketReceiver* Call::Receiver() {
// TODO(solenberg): Some test cases in EndToEndTest use this from a different
// thread. Re-enable once that is fixed.
- // RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ // RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
return this;
}
webrtc::AudioSendStream* Call::CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
event_log_->LogAudioSendStreamConfig(CreateRtcLogStreamConfig(config));
rtc::Optional<RtpState> suspended_rtp_state;
@@ -637,7 +637,7 @@
void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyAudioSendStream");
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RTC_DCHECK(send_stream != nullptr);
send_stream->Stop();
@@ -667,7 +667,7 @@
webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
const webrtc::AudioReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
event_log_->LogAudioReceiveStreamConfig(CreateRtcLogStreamConfig(config));
AudioReceiveStream* receive_stream = new AudioReceiveStream(
&audio_receiver_controller_, transport_send_->packet_router(), config,
@@ -695,7 +695,7 @@
void Call::DestroyAudioReceiveStream(
webrtc::AudioReceiveStream* receive_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream");
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RTC_DCHECK(receive_stream != nullptr);
webrtc::internal::AudioReceiveStream* audio_receive_stream =
static_cast<webrtc::internal::AudioReceiveStream*>(receive_stream);
@@ -723,7 +723,7 @@
webrtc::VideoSendStream::Config config,
VideoEncoderConfig encoder_config) {
TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
video_send_delay_stats_->AddSsrcs(config);
for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size();
@@ -759,7 +759,7 @@
void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
RTC_DCHECK(send_stream != nullptr);
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
send_stream->Stop();
@@ -794,7 +794,7 @@
webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
webrtc::VideoReceiveStream::Config configuration) {
TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
VideoReceiveStream* receive_stream = new VideoReceiveStream(
&video_receiver_controller_, num_cpu_cores_,
@@ -826,7 +826,7 @@
void Call::DestroyVideoReceiveStream(
webrtc::VideoReceiveStream* receive_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RTC_DCHECK(receive_stream != nullptr);
VideoReceiveStream* receive_stream_impl =
static_cast<VideoReceiveStream*>(receive_stream);
@@ -853,7 +853,7 @@
FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
const FlexfecReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateFlexfecReceiveStream");
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RecoveredPacketReceiver* recovered_packet_receiver = this;
@@ -886,7 +886,7 @@
void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyFlexfecReceiveStream");
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RTC_DCHECK(receive_stream != nullptr);
{
@@ -908,7 +908,7 @@
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_RUN_ON(&configuration_thread_checker_);
+ // RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
Stats stats;
// Fetch available send/receive bitrates.
uint32_t send_bandwidth = 0;
@@ -933,7 +933,7 @@
void Call::SetBitrateConfig(
const webrtc::Call::Config::BitrateConfig& bitrate_config) {
TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
RTC_DCHECK_NE(bitrate_config.start_bitrate_bps, 0);
if (bitrate_config.max_bitrate_bps != -1) {
@@ -957,7 +957,7 @@
void Call::SetBitrateConfigMask(
const webrtc::Call::Config::BitrateConfigMask& mask) {
TRACE_EVENT0("webrtc", "Call::SetBitrateConfigMask");
- RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
bitrate_config_mask_ = mask;
UpdateCurrentBitrateConfig(mask.start_bitrate_bps);
@@ -1012,7 +1012,7 @@
}
void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
switch (media) {
case MediaType::AUDIO:
audio_network_state_ = state;
@@ -1074,7 +1074,7 @@
// TODO(honghaiz): Add tests for this method.
void Call::OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) {
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
// Check if the network route is connected.
if (!network_route.connected) {
LOG(LS_INFO) << "Transport " << transport_name << " is disconnected";
@@ -1111,7 +1111,7 @@
}
void Call::UpdateAggregateNetworkState() {
- RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
bool have_audio = false;
bool have_video = false;
@@ -1377,7 +1377,7 @@
// TODO(solenberg): Tests call this function on a network thread, libjingle
// calls on the worker thread. We should move towards always using a network
// thread. Then this check can be enabled.
- // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
+ // RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
if (RtpHeaderParser::IsRtcp(packet, length))
return DeliverRtcp(media_type, packet, length);
