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Added thread checker to webrtc::Call.

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BUG=

Review URL: https://codereview.webrtc.org/1403353003

Cr-Commit-Position: refs/heads/master@{#10321}
This commit is contained in:
solenberg
2015-10-19 03:39:20 -07:00
committed by Commit bot
parent 22993e1a0c
commit 5a28939392
1 changed files with 28 additions and 1 deletions
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29
webrtc/call/call.cc
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@ -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 @@ class Call : public webrtc::Call, public PacketReceiver {
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 @@ Call::Call(const Call::Config& config)
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(const Call::Config& config)
}
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 @@ Call::~Call() {
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 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
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 @@ void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
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 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
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 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
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 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
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 @@ void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
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 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
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 @@ void Call::DestroyVideoReceiveStream(
}
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 @@ Call::Stats Call::GetStats() const {
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::SetBitrateConfig(
}
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::SignalNetworkState(NetworkState state) {
}
void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
channel_group_->OnSentPacket(sent_packet);
}
@ -568,6 +591,10 @@ PacketReceiver::DeliveryStatus Call::DeliverPacket(
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);