Added thread checker to webrtc::Call.
BUG=
Review URL: https://codereview.webrtc.org/1403353003
Cr-Commit-Position: refs/heads/master@{#10321}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index ce5bceb..c2b1ec7 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -18,6 +18,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
+#include "webrtc/base/thread_checker.h"
#include "webrtc/call.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/common.h"
@@ -96,6 +97,7 @@
const rtc::scoped_ptr<ChannelGroup> channel_group_;
volatile int next_channel_id_;
Call::Config config_;
+ rtc::ThreadChecker configuration_thread_checker_;
// Needs to be held while write-locking |receive_crit_| or |send_crit_|. This
// ensures that we have a consistent network state signalled to all senders
@@ -144,6 +146,7 @@
receive_crit_(RWLockWrapper::CreateRWLock()),
send_crit_(RWLockWrapper::CreateRWLock()),
event_log_(nullptr) {
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
config.bitrate_config.min_bitrate_bps);
@@ -168,6 +171,7 @@
}
Call::~Call() {
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_CHECK(audio_send_ssrcs_.empty());
RTC_CHECK(video_send_ssrcs_.empty());
RTC_CHECK(video_send_streams_.empty());
@@ -179,11 +183,17 @@
Trace::ReturnTrace();
}
-PacketReceiver* Call::Receiver() { return this; }
+PacketReceiver* Call::Receiver() {
+ // TODO(solenberg): Some test cases in EndToEndTest use this from a different
+ // thread. Re-enable once that is fixed.
+ // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+ return this;
+}
webrtc::AudioSendStream* Call::CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
AudioSendStream* send_stream = new AudioSendStream(config);
{
rtc::CritScope lock(&network_enabled_crit_);
@@ -200,6 +210,7 @@
void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyAudioSendStream");
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK(send_stream != nullptr);
send_stream->Stop();
@@ -218,6 +229,7 @@
webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
const webrtc::AudioReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
AudioReceiveStream* receive_stream = new AudioReceiveStream(
channel_group_->GetRemoteBitrateEstimator(false), config);
{
@@ -233,6 +245,7 @@
void Call::DestroyAudioReceiveStream(
webrtc::AudioReceiveStream* receive_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream");
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK(receive_stream != nullptr);
webrtc::internal::AudioReceiveStream* audio_receive_stream =
static_cast<webrtc::internal::AudioReceiveStream*>(receive_stream);
@@ -256,6 +269,7 @@
const webrtc::VideoSendStream::Config& config,
const VideoEncoderConfig& encoder_config) {
TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
// TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
// the call has already started.
@@ -285,6 +299,7 @@
void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
RTC_DCHECK(send_stream != nullptr);
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
send_stream->Stop();
@@ -318,6 +333,7 @@
webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
const webrtc::VideoReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
VideoReceiveStream* receive_stream = new VideoReceiveStream(
num_cpu_cores_, channel_group_.get(),
rtc::AtomicOps::Increment(&next_channel_id_), config,
@@ -351,6 +367,7 @@
void Call::DestroyVideoReceiveStream(
webrtc::VideoReceiveStream* receive_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK(receive_stream != nullptr);
VideoReceiveStream* receive_stream_impl = nullptr;
{
@@ -376,6 +393,9 @@
}
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(configuration_thread_checker_.CalledOnValidThread());
Stats stats;
// Fetch available send/receive bitrates.
uint32_t send_bandwidth = 0;
@@ -402,6 +422,7 @@
void Call::SetBitrateConfig(
const webrtc::Call::Config::BitrateConfig& bitrate_config) {
TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
if (bitrate_config.max_bitrate_bps != -1)
RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
@@ -422,6 +443,7 @@
}
void Call::SignalNetworkState(NetworkState state) {
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
// Take crit for entire function, it needs to be held while updating streams
// to guarantee a consistent state across streams.
rtc::CritScope lock(&network_enabled_crit_);
@@ -445,6 +467,7 @@
}
void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
channel_group_->OnSentPacket(sent_packet);
}
@@ -568,6 +591,10 @@
const uint8_t* packet,
size_t length,
const PacketTime& packet_time) {
+ // 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());
if (RtpHeaderParser::IsRtcp(packet, length))
return DeliverRtcp(media_type, packet, length);
