Refactoring: move ownership of RtcEventLog from Call to PeerConnection
This CL is a pure refactoring which should not result in any functinal
changes. It moves ownership of the RtcEventLog from webrtc::Call to the
webrtc::PeerConnection object.
This is done so that we can add RtcEventLog support for ICE events -
which will require the TransportController to have a pointer to the
RtcEventLog. PeerConnection is the closest common owner of both Call and
TransportController (through WebRtcSession).
BUG=webrtc:6393
Review-Url: https://codereview.webrtc.org/2353033005
Cr-Commit-Position: refs/heads/master@{#14578}
diff --git a/webrtc/api/DEPS b/webrtc/api/DEPS
index fcb506d..ee97620 100644
--- a/webrtc/api/DEPS
+++ b/webrtc/api/DEPS
@@ -5,6 +5,7 @@
"+webrtc/media",
"+webrtc/p2p",
"+webrtc/pc",
+ "+webrtc/logging/rtc_event_log",
"+webrtc/modules/audio_device",
"+webrtc/modules/rtp_rtcp",
"+webrtc/modules/video_coding",
@@ -20,7 +21,7 @@
"peerconnection_jni\.cc": [
"+webrtc/voice_engine",
],
- "peerconnectionfactory.\cc": [
+ "peerconnectionfactory\.cc": [
"+webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h",
- ]
+ ],
}
diff --git a/webrtc/api/mediacontroller.cc b/webrtc/api/mediacontroller.cc
index e688e39..319dd1f 100644
--- a/webrtc/api/mediacontroller.cc
+++ b/webrtc/api/mediacontroller.cc
@@ -30,11 +30,14 @@
public:
MediaController(const cricket::MediaConfig& media_config,
rtc::Thread* worker_thread,
- cricket::ChannelManager* channel_manager)
+ cricket::ChannelManager* channel_manager,
+ webrtc::RtcEventLog* event_log)
: worker_thread_(worker_thread),
media_config_(media_config),
- channel_manager_(channel_manager) {
+ channel_manager_(channel_manager),
+ call_config_(event_log) {
RTC_DCHECK(worker_thread);
+ RTC_DCHECK(event_log);
worker_thread_->Invoke<void>(RTC_FROM_HERE,
rtc::Bind(&MediaController::Construct_w, this,
channel_manager_->media_engine()));
@@ -89,7 +92,8 @@
MediaControllerInterface* MediaControllerInterface::Create(
const cricket::MediaConfig& config,
rtc::Thread* worker_thread,
- cricket::ChannelManager* channel_manager) {
- return new MediaController(config, worker_thread, channel_manager);
+ cricket::ChannelManager* channel_manager,
+ webrtc::RtcEventLog* event_log) {
+ return new MediaController(config, worker_thread, channel_manager, event_log);
}
} // namespace webrtc
diff --git a/webrtc/api/mediacontroller.h b/webrtc/api/mediacontroller.h
index d7c76ab..5ada20d 100644
--- a/webrtc/api/mediacontroller.h
+++ b/webrtc/api/mediacontroller.h
@@ -21,6 +21,7 @@
namespace webrtc {
class Call;
class VoiceEngine;
+class RtcEventLog;
// The MediaController currently owns shared state between media channels, but
// in the future will create and own RtpSenders and RtpReceivers.
@@ -29,7 +30,8 @@
static MediaControllerInterface* Create(
const cricket::MediaConfig& config,
rtc::Thread* worker_thread,
- cricket::ChannelManager* channel_manager);
+ cricket::ChannelManager* channel_manager,
+ webrtc::RtcEventLog* event_log);
virtual ~MediaControllerInterface() {}
virtual void Close() = 0;
diff --git a/webrtc/api/peerconnection.cc b/webrtc/api/peerconnection.cc
index 7cc5d00..dd99e36 100644
--- a/webrtc/api/peerconnection.cc
+++ b/webrtc/api/peerconnection.cc
@@ -37,8 +37,10 @@
#include "webrtc/base/stringutils.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/call.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/sctp/sctpdataengine.h"
#include "webrtc/pc/channelmanager.h"
+#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/field_trial.h"
namespace {
@@ -571,6 +573,7 @@
ice_state_(kIceNew),
ice_connection_state_(kIceConnectionNew),
ice_gathering_state_(kIceGatheringNew),
+ event_log_(RtcEventLog::Create(webrtc::Clock::GetRealTimeClock())),
rtcp_cname_(GenerateRtcpCname()),
local_streams_(StreamCollection::Create()),
remote_streams_(StreamCollection::Create()) {}
@@ -619,8 +622,8 @@
return false;
}
- media_controller_.reset(
- factory_->CreateMediaController(configuration.media_config));
+ media_controller_.reset(factory_->CreateMediaController(
+ configuration.media_config, event_log_.get()));
session_.reset(new WebRtcSession(
media_controller_.get(), factory_->network_thread(),
@@ -2343,10 +2346,10 @@
bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
int64_t max_size_bytes) {
- return media_controller_->call_w()->StartEventLog(file, max_size_bytes);
+ return event_log_->StartLogging(file, max_size_bytes);
}
void PeerConnection::StopRtcEventLog_w() {
- media_controller_->call_w()->StopEventLog();
+ event_log_->StopLogging();
}
} // namespace webrtc
diff --git a/webrtc/api/peerconnection.h b/webrtc/api/peerconnection.h
index 3087160..f5b0af8 100644
--- a/webrtc/api/peerconnection.h
+++ b/webrtc/api/peerconnection.h
@@ -29,6 +29,7 @@
class MediaStreamObserver;
class VideoRtpReceiver;
+class RtcEventLog;
// Populates |session_options| from |rtc_options|, and returns true if options
// are valid.
@@ -392,6 +393,8 @@
IceGatheringState ice_gathering_state_;
std::unique_ptr<cricket::PortAllocator> port_allocator_;
+ // The EventLog needs to outlive the media controller.
+ std::unique_ptr<RtcEventLog> event_log_;
std::unique_ptr<MediaControllerInterface> media_controller_;
// One PeerConnection has only one RTCP CNAME.
@@ -426,7 +429,6 @@
std::vector<
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
receivers_;
-
std::unique_ptr<WebRtcSession> session_;
std::unique_ptr<StatsCollector> stats_;
rtc::scoped_refptr<RTCStatsCollector> stats_collector_;
diff --git a/webrtc/api/peerconnectionfactory.cc b/webrtc/api/peerconnectionfactory.cc
index f49c291..e76701e 100644
--- a/webrtc/api/peerconnectionfactory.cc
+++ b/webrtc/api/peerconnectionfactory.cc
@@ -302,10 +302,11 @@
}
webrtc::MediaControllerInterface* PeerConnectionFactory::CreateMediaController(
- const cricket::MediaConfig& config) const {
+ const cricket::MediaConfig& config,
+ webrtc::RtcEventLog* event_log) const {
RTC_DCHECK(signaling_thread_->IsCurrent());
return MediaControllerInterface::Create(config, worker_thread_,
- channel_manager_.get());
+ channel_manager_.get(), event_log);
}
cricket::TransportController* PeerConnectionFactory::CreateTransportController(
diff --git a/webrtc/api/peerconnectionfactory.h b/webrtc/api/peerconnectionfactory.h
index 7684b95..7a30ab4 100644
--- a/webrtc/api/peerconnectionfactory.h
+++ b/webrtc/api/peerconnectionfactory.h
@@ -29,6 +29,8 @@
namespace webrtc {
+class RtcEventLog;
+
class PeerConnectionFactory : public PeerConnectionFactoryInterface {
public:
void SetOptions(const Options& options) override;
@@ -89,7 +91,8 @@
void StopRtcEventLog() override {}
virtual webrtc::MediaControllerInterface* CreateMediaController(
- const cricket::MediaConfig& config) const;
+ const cricket::MediaConfig& config,
+ RtcEventLog* event_log) const;
virtual cricket::TransportController* CreateTransportController(
cricket::PortAllocator* port_allocator,
bool redetermine_role_on_ice_restart);
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc
index d91336c..a4a11e1 100644
--- a/webrtc/api/peerconnectioninterface_unittest.cc
+++ b/webrtc/api/peerconnectioninterface_unittest.cc
@@ -558,12 +558,13 @@
class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory {
public:
webrtc::MediaControllerInterface* CreateMediaController(
- const cricket::MediaConfig& config) const override {
+ const cricket::MediaConfig& config,
+ webrtc::RtcEventLog* event_log) const override {
create_media_controller_called_ = true;
create_media_controller_config_ = config;
webrtc::MediaControllerInterface* mc =
- PeerConnectionFactory::CreateMediaController(config);
+ PeerConnectionFactory::CreateMediaController(config, event_log);
EXPECT_TRUE(mc != nullptr);
return mc;
}
diff --git a/webrtc/api/rtcstatscollector_unittest.cc b/webrtc/api/rtcstatscollector_unittest.cc
index 7103864..0da772f 100644
--- a/webrtc/api/rtcstatscollector_unittest.cc
+++ b/webrtc/api/rtcstatscollector_unittest.cc
@@ -29,6 +29,7 @@
#include "webrtc/base/thread_checker.h"
#include "webrtc/base/timedelta.h"
#include "webrtc/base/timeutils.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakemediaengine.h"
#include "webrtc/p2p/base/port.h"
@@ -112,14 +113,15 @@
RTCStatsCollectorTestHelper()
: worker_thread_(rtc::Thread::Current()),
network_thread_(rtc::Thread::Current()),
- channel_manager_(new cricket::ChannelManager(
- new cricket::FakeMediaEngine(),
- worker_thread_,
- network_thread_)),
+ channel_manager_(
+ new cricket::ChannelManager(new cricket::FakeMediaEngine(),
+ worker_thread_,
+ network_thread_)),
media_controller_(
MediaControllerInterface::Create(cricket::MediaConfig(),
worker_thread_,
- channel_manager_.get())),
+ channel_manager_.get(),
+ &event_log_)),
session_(media_controller_.get()),
pc_() {
// Default return values for mocks.
@@ -148,6 +150,7 @@
private:
rtc::ScopedFakeClock fake_clock_;
+ webrtc::RtcEventLogNullImpl event_log_;
rtc::Thread* const worker_thread_;
rtc::Thread* const network_thread_;
std::unique_ptr<cricket::ChannelManager> channel_manager_;
diff --git a/webrtc/api/rtpsenderreceiver_unittest.cc b/webrtc/api/rtpsenderreceiver_unittest.cc
index 3fccfb9..4e98f0c 100644
--- a/webrtc/api/rtpsenderreceiver_unittest.cc
+++ b/webrtc/api/rtpsenderreceiver_unittest.cc
@@ -24,6 +24,7 @@
#include "webrtc/api/videotrack.h"
#include "webrtc/api/videotracksource.h"
#include "webrtc/base/gunit.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakemediaengine.h"
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/media/engine/fakewebrtccall.h"
@@ -56,7 +57,7 @@
channel_manager_(media_engine_,
rtc::Thread::Current(),
rtc::Thread::Current()),
- fake_call_(webrtc::Call::Config()),
+ fake_call_(Call::Config(&event_log_)),
fake_media_controller_(&channel_manager_, &fake_call_),
stream_(MediaStream::Create(kStreamLabel1)) {
// Create channels to be used by the RtpSenders and RtpReceivers.
@@ -218,6 +219,7 @@
}
protected:
+ webrtc::RtcEventLogNullImpl event_log_;
cricket::FakeMediaEngine* media_engine_;
cricket::FakeTransportController fake_transport_controller_;
cricket::ChannelManager channel_manager_;
diff --git a/webrtc/api/statscollector_unittest.cc b/webrtc/api/statscollector_unittest.cc
index acef448..8d5978f 100644
--- a/webrtc/api/statscollector_unittest.cc
+++ b/webrtc/api/statscollector_unittest.cc
@@ -29,6 +29,7 @@
#include "webrtc/base/fakesslidentity.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/network.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakemediaengine.h"
#include "webrtc/media/base/test/mock_mediachannel.h"
#include "webrtc/p2p/base/faketransportcontroller.h"
@@ -493,7 +494,8 @@
media_controller_(
webrtc::MediaControllerInterface::Create(cricket::MediaConfig(),
worker_thread_,
- channel_manager_.get())),
+ channel_manager_.get(),
+ &event_log_)),
session_(media_controller_.get()) {
// By default, we ignore session GetStats calls.
EXPECT_CALL(session_, GetTransportStats(_)).WillRepeatedly(Return(false));
@@ -751,6 +753,7 @@
srtp_crypto_suite);
}
+ webrtc::RtcEventLogNullImpl event_log_;
rtc::Thread* const worker_thread_;
rtc::Thread* const network_thread_;
cricket::FakeMediaEngine* media_engine_;
diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc
index 6c0170e..ba9ea2b 100644
--- a/webrtc/api/webrtcsession_unittest.cc
+++ b/webrtc/api/webrtcsession_unittest.cc
@@ -35,6 +35,7 @@
#include "webrtc/base/stringutils.h"
#include "webrtc/base/thread.h"
#include "webrtc/base/virtualsocketserver.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakemediaengine.h"
#include "webrtc/media/base/fakevideorenderer.h"
#include "webrtc/media/base/mediachannel.h"
@@ -331,15 +332,15 @@
WebRtcSessionTest()
: media_engine_(new cricket::FakeMediaEngine()),
data_engine_(new cricket::FakeDataEngine()),
- channel_manager_(
- new cricket::ChannelManager(media_engine_,
- data_engine_,
- rtc::Thread::Current())),
- fake_call_(webrtc::Call::Config()),
+ channel_manager_(new cricket::ChannelManager(media_engine_,
+ data_engine_,
+ rtc::Thread::Current())),
+ fake_call_(webrtc::Call::Config(&event_log_)),
media_controller_(
webrtc::MediaControllerInterface::Create(cricket::MediaConfig(),
rtc::Thread::Current(),
- channel_manager_.get())),
+ channel_manager_.get(),
+ &event_log_)),
tdesc_factory_(new cricket::TransportDescriptionFactory()),
desc_factory_(
new cricket::MediaSessionDescriptionFactory(channel_manager_.get(),
@@ -1480,6 +1481,7 @@
allocator_->set_flags(cricket::PORTALLOCATOR_DISABLE_TCP);
}
+ webrtc::RtcEventLogNullImpl event_log_;
cricket::FakeMediaEngine* media_engine_;
cricket::FakeDataEngine* data_engine_;
std::unique_ptr<cricket::ChannelManager> channel_manager_;
diff --git a/webrtc/call.h b/webrtc/call.h
index 193124e..9855b60 100644
--- a/webrtc/call.h
+++ b/webrtc/call.h
@@ -26,6 +26,7 @@
namespace webrtc {
class AudioProcessing;
+class RtcEventLog;
const char* Version();
@@ -72,6 +73,10 @@
class Call {
public:
struct Config {
+ explicit Config(RtcEventLog* event_log) : event_log(event_log) {
+ RTC_DCHECK(event_log);
+ }
+
static const int kDefaultStartBitrateBps;
// Bitrate config used until valid bitrate estimates are calculated. Also
@@ -89,6 +94,10 @@
// Audio Processing Module to be used in this call.
// TODO(solenberg): Change this to a shared_ptr once we can use C++11.
AudioProcessing* audio_processing = nullptr;
+
+ // RtcEventLog to use for this call. Required.
+ // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
+ RtcEventLog* event_log = nullptr;
};
struct Stats {
@@ -151,10 +160,6 @@
virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
- virtual bool StartEventLog(rtc::PlatformFile log_file,
- int64_t max_size_bytes) = 0;
- virtual void StopEventLog() = 0;
-
virtual ~Call() {}
};
diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc
index 60e1d8c..80949ba 100644
--- a/webrtc/call/bitrate_estimator_tests.cc
+++ b/webrtc/call/bitrate_estimator_tests.cc
@@ -109,7 +109,7 @@
virtual void SetUp() {
AudioState::Config audio_state_config;
audio_state_config.voice_engine = &mock_voice_engine_;
- Call::Config config;
+ Call::Config config(&event_log_);
config.audio_state = AudioState::Create(audio_state_config);
receiver_call_.reset(Call::Create(config));
sender_call_.reset(Call::Create(config));
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index dd08d77..5d6bbab 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -109,13 +109,6 @@
void OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
uint32_t max_padding_bitrate_bps) override;
- bool StartEventLog(rtc::PlatformFile log_file,
- int64_t max_size_bytes) override {
- return event_log_->StartLogging(log_file, max_size_bytes);
- }
-
- void StopEventLog() override { event_log_->StopLogging(); }
-
private:
DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
size_t length);
@@ -171,8 +164,7 @@
std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_);
VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
-
- std::unique_ptr<webrtc::RtcEventLog> event_log_;
+ webrtc::RtcEventLog* event_log_;
// The following members are only accessed (exclusively) from one thread and
// from the destructor, and therefore doesn't need any explicit
@@ -237,7 +229,7 @@
video_network_state_(kNetworkUp),
receive_crit_(RWLockWrapper::CreateRWLock()),
send_crit_(RWLockWrapper::CreateRWLock()),
- event_log_(RtcEventLog::Create(webrtc::Clock::GetRealTimeClock())),
+ event_log_(config.event_log),
first_packet_sent_ms_(-1),
received_bytes_per_second_counter_(clock_, nullptr, true),
received_audio_bytes_per_second_counter_(clock_, nullptr, true),
@@ -249,11 +241,12 @@
pacer_bitrate_kbps_counter_(clock_, nullptr, true),
remb_(clock_),
congestion_controller_(
- new CongestionController(clock_, this, &remb_, event_log_.get())),
+ new CongestionController(clock_, this, &remb_, event_log_)),
video_send_delay_stats_(new SendDelayStats(clock_)),
start_ms_(clock_->TimeInMilliseconds()),
worker_queue_("call_worker_queue") {
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(config.event_log != nullptr);
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);
@@ -261,7 +254,6 @@
RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
config.bitrate_config.start_bitrate_bps);
}
-
Trace::CreateTrace();
call_stats_->RegisterStatsObserver(congestion_controller_.get());
@@ -380,7 +372,7 @@
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
AudioSendStream* send_stream = new AudioSendStream(
config, config_.audio_state, &worker_queue_, congestion_controller_.get(),
- bitrate_allocator_.get(), event_log_.get());
+ bitrate_allocator_.get(), event_log_);
{
WriteLockScoped write_lock(*send_crit_);
RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
@@ -415,9 +407,8 @@
const webrtc::AudioReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
- AudioReceiveStream* receive_stream =
- new AudioReceiveStream(congestion_controller_.get(), config,
- config_.audio_state, event_log_.get());
+ AudioReceiveStream* receive_stream = new AudioReceiveStream(
+ congestion_controller_.get(), config, config_.audio_state, event_log_);
{
WriteLockScoped write_lock(*receive_crit_);
RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
@@ -470,9 +461,8 @@
VideoSendStream* send_stream = new VideoSendStream(
num_cpu_cores_, module_process_thread_.get(), &worker_queue_,
call_stats_.get(), congestion_controller_.get(), bitrate_allocator_.get(),
- video_send_delay_stats_.get(), &remb_, event_log_.get(),
- std::move(config), std::move(encoder_config),
- suspended_video_send_ssrcs_);
+ video_send_delay_stats_.get(), &remb_, event_log_, std::move(config),
+ std::move(encoder_config), suspended_video_send_ssrcs_);
{
WriteLockScoped write_lock(*send_crit_);
@@ -887,7 +877,7 @@
}
}
- if (event_log_ && rtcp_delivered)
+ if (rtcp_delivered)
event_log_->LogRtcpPacket(kIncomingPacket, media_type, packet, length);
return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index 4324d81..43d7aa5 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -19,6 +19,7 @@
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
#include "webrtc/config.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
@@ -165,9 +166,9 @@
AudioState::Config send_audio_state_config;
send_audio_state_config.voice_engine = voice_engine;
- Call::Config sender_config;
+ Call::Config sender_config(&event_log_);
sender_config.audio_state = AudioState::Create(send_audio_state_config);
- Call::Config receiver_config;
+ Call::Config receiver_config(&event_log_);
receiver_config.audio_state = sender_config.audio_state;
CreateCalls(sender_config, receiver_config);
@@ -685,6 +686,7 @@
Call::Config GetSenderCallConfig() override {
Call::Config config = EndToEndTest::GetSenderCallConfig();
+ config.event_log = &event_log_;
config.bitrate_config.start_bitrate_bps = kInitialBitrateKbps * 1000;
return config;
}
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
index 2fe4fde..d452927 100644
--- a/webrtc/call/call_unittest.cc
+++ b/webrtc/call/call_unittest.cc
@@ -13,6 +13,7 @@
#include "webrtc/api/call/audio_state.h"
#include "webrtc/call.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/mock_voice_engine.h"
@@ -25,7 +26,7 @@
: voice_engine_(decoder_factory) {
webrtc::AudioState::Config audio_state_config;
audio_state_config.voice_engine = &voice_engine_;
- webrtc::Call::Config config;
+ webrtc::Call::Config config(&event_log_);
config.audio_state = webrtc::AudioState::Create(audio_state_config);
call_.reset(webrtc::Call::Create(config));
}
@@ -34,6 +35,7 @@
private:
testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_;
+ webrtc::RtcEventLogNullImpl event_log_;
std::unique_ptr<webrtc::Call> call_;
};
} // namespace
diff --git a/webrtc/call/packet_injection_tests.cc b/webrtc/call/packet_injection_tests.cc
index 42f4fae..df19cf3 100644
--- a/webrtc/call/packet_injection_tests.cc
+++ b/webrtc/call/packet_injection_tests.cc
@@ -10,6 +10,7 @@
#include <memory>
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/null_transport.h"
@@ -37,8 +38,8 @@
uint8_t payload_type,
const uint8_t* packet,
size_t length) {
- CreateSenderCall(Call::Config());
- CreateReceiverCall(Call::Config());
+ CreateSenderCall(Call::Config(&event_log_));
+ CreateReceiverCall(Call::Config(&event_log_));
test::NullTransport null_transport;
CreateSendConfig(1, 0, &null_transport);
diff --git a/webrtc/call/rampup_tests.cc b/webrtc/call/rampup_tests.cc
index 0199cc8..df64f41 100644
--- a/webrtc/call/rampup_tests.cc
+++ b/webrtc/call/rampup_tests.cc
@@ -68,7 +68,7 @@
}
Call::Config RampUpTester::GetSenderCallConfig() {
- Call::Config call_config;
+ Call::Config call_config(&event_log_);
if (start_bitrate_bps_ != 0) {
call_config.bitrate_config.start_bitrate_bps = start_bitrate_bps_;
}
@@ -382,7 +382,7 @@
}
Call::Config RampUpDownUpTester::GetReceiverCallConfig() {
- Call::Config config;
+ Call::Config config(&event_log_);
config.bitrate_config.min_bitrate_bps = 10000;
return config;
}
diff --git a/webrtc/call/rampup_tests.h b/webrtc/call/rampup_tests.h
index f3779e2..da1ca74 100644
--- a/webrtc/call/rampup_tests.h
+++ b/webrtc/call/rampup_tests.h
@@ -17,6 +17,7 @@
#include "webrtc/base/event.h"
#include "webrtc/call.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/test/call_test.h"
namespace webrtc {
@@ -57,6 +58,7 @@
const std::string& units) const;
void TriggerTestDone();
+ webrtc::RtcEventLogNullImpl event_log_;
rtc::Event event_;
Clock* const clock_;
FakeNetworkPipe::Config forward_transport_config_;
diff --git a/webrtc/media/DEPS b/webrtc/media/DEPS
index e444563..580ac5c 100644
--- a/webrtc/media/DEPS
+++ b/webrtc/media/DEPS
@@ -3,6 +3,7 @@
"+webrtc/base",
"+webrtc/call",
"+webrtc/common_video",
+ "+webrtc/logging/rtc_event_log",
"+webrtc/modules/audio_coding",
"+webrtc/modules/audio_device",
"+webrtc/modules/audio_processing",
diff --git a/webrtc/media/base/videoengine_unittest.h b/webrtc/media/base/videoengine_unittest.h
index ac8ed47..6b281bb 100644
--- a/webrtc/media/base/videoengine_unittest.h
+++ b/webrtc/media/base/videoengine_unittest.h
@@ -19,6 +19,7 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/call.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakenetworkinterface.h"
#include "webrtc/media/base/fakevideocapturer.h"
#include "webrtc/media/base/fakevideorenderer.h"
@@ -79,7 +80,7 @@
public sigslot::has_slots<> {
protected:
VideoMediaChannelTest<E, C>()
- : call_(webrtc::Call::Create(webrtc::Call::Config())) {}
+ : call_(webrtc::Call::Create(webrtc::Call::Config(&event_log_))) {}
virtual cricket::VideoCodec DefaultCodec() = 0;
@@ -1076,6 +1077,7 @@
EXPECT_EQ(1, renderer2_.num_rendered_frames());
}
+ webrtc::RtcEventLogNullImpl event_log_;
const std::unique_ptr<webrtc::Call> call_;
E engine_;
std::unique_ptr<cricket::FakeVideoCapturer> video_capturer_;
diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
index 88f9927..d021eca 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -510,11 +510,4 @@
}
}
-bool FakeCall::StartEventLog(rtc::PlatformFile log_file,
- int64_t max_size_bytes) {
- return false;
-}
-
-void FakeCall::StopEventLog() {}
-
} // namespace cricket
diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h
index 5719070..db3db18 100644
--- a/webrtc/media/engine/fakewebrtccall.h
+++ b/webrtc/media/engine/fakewebrtccall.h
@@ -245,10 +245,6 @@
webrtc::NetworkState state) override;
void OnSentPacket(const rtc::SentPacket& sent_packet) override;
- bool StartEventLog(rtc::PlatformFile log_file,
- int64_t max_size_bytes) override;
- void StopEventLog() override;
-
webrtc::Call::Config config_;
webrtc::NetworkState audio_network_state_;
webrtc::NetworkState video_network_state_;
diff --git a/webrtc/media/engine/webrtcvideoengine2_unittest.cc b/webrtc/media/engine/webrtcvideoengine2_unittest.cc
index fd41640..a447956 100644
--- a/webrtc/media/engine/webrtcvideoengine2_unittest.cc
+++ b/webrtc/media/engine/webrtcvideoengine2_unittest.cc
@@ -16,6 +16,7 @@
#include "webrtc/base/arraysize.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/stringutils.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/testutils.h"
#include "webrtc/media/base/videoengine_unittest.h"
#include "webrtc/media/engine/fakewebrtccall.h"
@@ -96,7 +97,7 @@
WebRtcVideoEngine2Test() : WebRtcVideoEngine2Test("") {}
explicit WebRtcVideoEngine2Test(const char* field_trials)
: override_field_trials_(field_trials),
- call_(webrtc::Call::Create(webrtc::Call::Config())),
+ call_(webrtc::Call::Create(webrtc::Call::Config(&event_log_))),
engine_() {
std::vector<VideoCodec> engine_codecs = engine_.codecs();
RTC_DCHECK(!engine_codecs.empty());
@@ -133,6 +134,7 @@
void TestExtendedEncoderOveruse(bool use_external_encoder);
webrtc::test::ScopedFieldTrials override_field_trials_;
+ webrtc::RtcEventLogNullImpl event_log_;
// Used in WebRtcVideoEngine2VoiceTest, but defined here so it's properly
// initialized when the constructor is called.
std::unique_ptr<webrtc::Call> call_;
@@ -412,7 +414,7 @@
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(kVp8Codec);
std::unique_ptr<VideoMediaChannel> channel;
- FakeCall* fake_call = new FakeCall(webrtc::Call::Config());
+ FakeCall* fake_call = new FakeCall(webrtc::Call::Config(&event_log_));
call_.reset(fake_call);
if (use_external_encoder) {
channel.reset(
@@ -462,8 +464,7 @@
encoder_factory.AddSupportedVideoCodecType(webrtc::kVideoCodecVP8, "VP8");
std::vector<cricket::VideoCodec> codecs;
codecs.push_back(kVp8Codec);
-
- FakeCall* fake_call = new FakeCall(webrtc::Call::Config());
+ FakeCall* fake_call = new FakeCall(webrtc::Call::Config(&event_log_));
call_.reset(fake_call);
std::unique_ptr<VideoMediaChannel> channel(
SetUpForExternalEncoderFactory(&encoder_factory, codecs));
@@ -880,7 +881,7 @@
explicit WebRtcVideoChannel2Test(const char* field_trials)
: WebRtcVideoEngine2Test(field_trials), last_ssrc_(0) {}
void SetUp() override {
- fake_call_.reset(new FakeCall(webrtc::Call::Config()));
+ fake_call_.reset(new FakeCall(webrtc::Call::Config(&event_log_)));
engine_.Init();
channel_.reset(
engine_.CreateChannel(fake_call_.get(), MediaConfig(), VideoOptions()));
@@ -3724,7 +3725,8 @@
class WebRtcVideoChannel2SimulcastTest : public testing::Test {
public:
- WebRtcVideoChannel2SimulcastTest() : fake_call_(webrtc::Call::Config()) {}
+ WebRtcVideoChannel2SimulcastTest()
+ : fake_call_(webrtc::Call::Config(&event_log_)) {}
void SetUp() override {
engine_.Init();
@@ -3839,6 +3841,7 @@
return streams[streams.size() - 1];
}
+ webrtc::RtcEventLogNullImpl event_log_;
FakeCall fake_call_;
WebRtcVideoEngine2 engine_;
std::unique_ptr<VideoMediaChannel> channel_;
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 8be3b9d..80c3fda 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -17,6 +17,7 @@
#include "webrtc/call.h"
#include "webrtc/p2p/base/faketransportcontroller.h"
#include "webrtc/test/field_trial.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakemediaengine.h"
#include "webrtc/media/base/fakenetworkinterface.h"
#include "webrtc/media/base/fakertp.h"
@@ -62,6 +63,7 @@
engine) { // volume
}
};
+
} // namespace
// Tests that our stub library "works".
@@ -97,7 +99,8 @@
WebRtcVoiceEngineTestFake() : WebRtcVoiceEngineTestFake("") {}
explicit WebRtcVoiceEngineTestFake(const char* field_trials)
- : call_(webrtc::Call::Config()), override_field_trials_(field_trials) {
+ : call_(webrtc::Call::Config(&event_log_)),
+ override_field_trials_(field_trials) {
auto factory = webrtc::MockAudioDecoderFactory::CreateUnusedFactory();
EXPECT_CALL(adm_, AddRef()).WillOnce(Return(0));
EXPECT_CALL(adm_, Release()).WillOnce(Return(0));
@@ -492,6 +495,7 @@
protected:
StrictMock<webrtc::test::MockAudioDeviceModule> adm_;
+ webrtc::RtcEventLogNullImpl event_log_;
cricket::FakeCall call_;
cricket::FakeWebRtcVoiceEngine voe_;
std::unique_ptr<cricket::WebRtcVoiceEngine> engine_;
@@ -3361,8 +3365,9 @@
// we never want it to create a decoder at this stage.
cricket::WebRtcVoiceEngine engine(
nullptr, webrtc::MockAudioDecoderFactory::CreateUnusedFactory());
+ webrtc::RtcEventLogNullImpl event_log;
std::unique_ptr<webrtc::Call> call(
- webrtc::Call::Create(webrtc::Call::Config()));
+ webrtc::Call::Create(webrtc::Call::Config(&event_log)));
cricket::VoiceMediaChannel* channel = engine.CreateChannel(
call.get(), cricket::MediaConfig(), cricket::AudioOptions());
EXPECT_TRUE(channel != nullptr);
@@ -3377,8 +3382,9 @@
{
cricket::WebRtcVoiceEngine engine(
&adm, webrtc::MockAudioDecoderFactory::CreateUnusedFactory());
+ webrtc::RtcEventLogNullImpl event_log;
std::unique_ptr<webrtc::Call> call(
- webrtc::Call::Create(webrtc::Call::Config()));
+ webrtc::Call::Create(webrtc::Call::Config(&event_log)));
cricket::VoiceMediaChannel* channel = engine.CreateChannel(
call.get(), cricket::MediaConfig(), cricket::AudioOptions());
EXPECT_TRUE(channel != nullptr);
@@ -3475,8 +3481,9 @@
TEST(WebRtcVoiceEngineTest, Has32Channels) {
cricket::WebRtcVoiceEngine engine(
nullptr, webrtc::MockAudioDecoderFactory::CreateUnusedFactory());
+ webrtc::RtcEventLogNullImpl event_log;
std::unique_ptr<webrtc::Call> call(
- webrtc::Call::Create(webrtc::Call::Config()));
+ webrtc::Call::Create(webrtc::Call::Config(&event_log)));
cricket::VoiceMediaChannel* channels[32];
int num_channels = 0;
@@ -3507,8 +3514,9 @@
// I think it will become clear once audio decoder injection is completed.
cricket::WebRtcVoiceEngine engine(
nullptr, webrtc::CreateBuiltinAudioDecoderFactory());
+ webrtc::RtcEventLogNullImpl event_log;
std::unique_ptr<webrtc::Call> call(
- webrtc::Call::Create(webrtc::Call::Config()));
+ webrtc::Call::Create(webrtc::Call::Config(&event_log)));
cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),
cricket::AudioOptions(), call.get());
cricket::AudioRecvParameters parameters;
diff --git a/webrtc/pc/DEPS b/webrtc/pc/DEPS
index ca4f789..e1c9e39 100644
--- a/webrtc/pc/DEPS
+++ b/webrtc/pc/DEPS
@@ -1,6 +1,7 @@
include_rules = [
"+webrtc/api",
"+webrtc/base",
+ "+webrtc/logging/rtc_event_log",
"+webrtc/media",
"+webrtc/p2p",
"+third_party/libsrtp"
diff --git a/webrtc/pc/channelmanager_unittest.cc b/webrtc/pc/channelmanager_unittest.cc
index e4e243c..0beb940 100644
--- a/webrtc/pc/channelmanager_unittest.cc
+++ b/webrtc/pc/channelmanager_unittest.cc
@@ -12,6 +12,7 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/thread.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakemediaengine.h"
#include "webrtc/media/base/fakevideocapturer.h"
#include "webrtc/media/base/testutils.h"
@@ -35,10 +36,8 @@
ChannelManagerTest()
: fme_(new cricket::FakeMediaEngine()),
fdme_(new cricket::FakeDataEngine()),
- cm_(new cricket::ChannelManager(fme_,
- fdme_,
- rtc::Thread::Current())),
- fake_call_(webrtc::Call::Config()),
+ cm_(new cricket::ChannelManager(fme_, fdme_, rtc::Thread::Current())),
+ fake_call_(webrtc::Call::Config(&event_log_)),
fake_mc_(cm_, &fake_call_),
transport_controller_(
new cricket::FakeTransportController(ICEROLE_CONTROLLING)) {}
@@ -56,6 +55,7 @@
fme_ = NULL;
}
+ webrtc::RtcEventLogNullImpl event_log_;
rtc::Thread network_;
rtc::Thread worker_;
cricket::FakeMediaEngine* fme_;
diff --git a/webrtc/test/DEPS b/webrtc/test/DEPS
index 969a10c..ccf57d0 100644
--- a/webrtc/test/DEPS
+++ b/webrtc/test/DEPS
@@ -2,6 +2,7 @@
"+webrtc/base",
"+webrtc/call",
"+webrtc/common_video",
+ "+webrtc/logging/rtc_event_log",
"+webrtc/media/base",
"+webrtc/modules/audio_coding",
"+webrtc/modules/audio_device",
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index 8c6ac45..57aca89 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -382,11 +382,11 @@
}
Call::Config BaseTest::GetSenderCallConfig() {
- return Call::Config();
+ return Call::Config(&event_log_);
}
Call::Config BaseTest::GetReceiverCallConfig() {
- return Call::Config();
+ return Call::Config(&event_log_);
}
void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
index 29bdbf2..ff84782 100644
--- a/webrtc/test/call_test.h
+++ b/webrtc/test/call_test.h
@@ -14,6 +14,7 @@
#include <vector>
#include "webrtc/call.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_audio_device.h"
#include "webrtc/test/fake_decoder.h"
@@ -90,6 +91,7 @@
Clock* const clock_;
+ webrtc::RtcEventLogNullImpl event_log_;
std::unique_ptr<Call> sender_call_;
std::unique_ptr<PacketTransport> send_transport_;
VideoSendStream::Config video_send_config_;
@@ -179,6 +181,8 @@
virtual void OnFrameGeneratorCapturerCreated(
FrameGeneratorCapturer* frame_generator_capturer);
+
+ webrtc::RtcEventLogNullImpl event_log_;
};
class SendTest : public BaseTest {
diff --git a/webrtc/video/DEPS b/webrtc/video/DEPS
index 7e53144..fafad7b 100644
--- a/webrtc/video/DEPS
+++ b/webrtc/video/DEPS
@@ -2,6 +2,7 @@
"+webrtc/base",
"+webrtc/call",
"+webrtc/common_video",
+ "+webrtc/logging/rtc_event_log",
"+webrtc/media/base",
"+webrtc/modules/bitrate_controller",
"+webrtc/modules/congestion_controller",
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 631f4d9..ce05dcf 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -23,6 +23,7 @@
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
#include "webrtc/common_video/include/frame_callback.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakevideorenderer.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
@@ -131,7 +132,7 @@
};
TEST_F(EndToEndTest, ReceiverCanBeStartedTwice) {
- CreateCalls(Call::Config(), Call::Config());
+ CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_));
test::NullTransport transport;
CreateSendConfig(1, 0, &transport);
@@ -146,7 +147,7 @@
}
TEST_F(EndToEndTest, ReceiverCanBeStoppedTwice) {
- CreateCalls(Call::Config(), Call::Config());
+ CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_));
test::NullTransport transport;
CreateSendConfig(1, 0, &transport);
@@ -194,7 +195,7 @@
rtc::Event event_;
};
- CreateCalls(Call::Config(), Call::Config());
+ CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_));
test::DirectTransport sender_transport(sender_call_.get());
test::DirectTransport receiver_transport(receiver_call_.get());
@@ -244,7 +245,7 @@
rtc::Event event_;
} renderer;
- CreateCalls(Call::Config(), Call::Config());
+ CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_));
test::DirectTransport sender_transport(sender_call_.get());
test::DirectTransport receiver_transport(receiver_call_.get());
@@ -771,7 +772,7 @@
// TODO(holmer): Investigate why we don't send FEC packets when the bitrate
// is 10 kbps.
Call::Config GetSenderCallConfig() override {
- Call::Config config;
+ Call::Config config(&event_log_);
const int kMinBitrateBps = 30000;
config.bitrate_config.min_bitrate_bps = kMinBitrateBps;
return config;
@@ -1102,7 +1103,7 @@
rtc::Event delivered_packet_;
};
- CreateCalls(Call::Config(), Call::Config());
+ CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_));
test::DirectTransport send_transport(sender_call_.get());
test::DirectTransport receive_transport(receiver_call_.get());
@@ -1243,8 +1244,10 @@
virtual ~MultiStreamTest() {}
void RunTest() {
- std::unique_ptr<Call> sender_call(Call::Create(Call::Config()));
- std::unique_ptr<Call> receiver_call(Call::Create(Call::Config()));
+ webrtc::RtcEventLogNullImpl event_log;
+ Call::Config config(&event_log);
+ std::unique_ptr<Call> sender_call(Call::Create(config));
+ std::unique_ptr<Call> receiver_call(Call::Create(config));
std::unique_ptr<test::DirectTransport> sender_transport(
CreateSendTransport(sender_call.get()));
std::unique_ptr<test::DirectTransport> receiver_transport(
@@ -1739,7 +1742,7 @@
EncodedFrameTestObserver post_encode_observer;
EncodedFrameTestObserver pre_decode_observer;
- CreateCalls(Call::Config(), Call::Config());
+ CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_));
test::DirectTransport sender_transport(sender_call_.get());
test::DirectTransport receiver_transport(receiver_call_.get());
@@ -1890,7 +1893,7 @@
}
Call::Config GetSenderCallConfig() override {
- Call::Config config;
+ Call::Config config(&event_log_);
// Set a high start bitrate to reduce the test completion time.
config.bitrate_config.start_bitrate_bps = remb_bitrate_bps_;
return config;
@@ -3269,7 +3272,8 @@
std::map<uint32_t, bool> ssrc_observed_ GUARDED_BY(crit_);
} observer(use_rtx);
- CreateCalls(Call::Config(), Call::Config());
+ Call::Config config(&event_log_);
+ CreateCalls(config, config);
test::PacketTransport send_transport(sender_call_.get(), &observer,
test::PacketTransport::kSender,
@@ -3564,8 +3568,7 @@
TEST_F(EndToEndTest, CallReportsRttForSender) {
static const int kSendDelayMs = 30;
static const int kReceiveDelayMs = 70;
-
- CreateCalls(Call::Config(), Call::Config());
+ CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_));
FakeNetworkPipe::Config config;
config.queue_delay_ms = kSendDelayMs;
@@ -3607,7 +3610,7 @@
MediaType network_to_bring_down,
VideoEncoder* encoder,
Transport* transport) {
- CreateSenderCall(Call::Config());
+ CreateSenderCall(Call::Config(&event_log_));
sender_call_->SignalChannelNetworkState(network_to_bring_down, kNetworkDown);
CreateSendConfig(1, 0, transport);
@@ -3626,7 +3629,8 @@
void EndToEndTest::VerifyNewVideoReceiveStreamsRespectNetworkState(
MediaType network_to_bring_down,
Transport* transport) {
- CreateCalls(Call::Config(), Call::Config());
+ Call::Config config(&event_log_);
+ CreateCalls(config, config);
receiver_call_->SignalChannelNetworkState(network_to_bring_down,
kNetworkDown);
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 9adcc16..dabe500 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -25,6 +25,7 @@
#include "webrtc/base/timeutils.h"
#include "webrtc/call.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/include/cpu_info.h"
@@ -1151,7 +1152,8 @@
<< "!";
}
- Call::Config call_config;
+ webrtc::RtcEventLogNullImpl event_log;
+ Call::Config call_config(&event_log_);
call_config.bitrate_config = params.common.call_bitrate_config;
CreateCalls(call_config, call_config);
@@ -1261,7 +1263,8 @@
// TODO(ivica): Remove bitrate_config and use the default Call::Config(), to
// match the full stack tests.
- Call::Config call_config;
+ webrtc::RtcEventLogNullImpl event_log;
+ Call::Config call_config(&event_log_);
call_config.bitrate_config = params_.common.call_bitrate_config;
::VoiceEngineState voe;
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index dd8513f..d138893 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -60,8 +60,7 @@
};
TEST_F(VideoSendStreamTest, CanStartStartedStream) {
- Call::Config call_config;
- CreateSenderCall(call_config);
+ CreateSenderCall(Call::Config(&event_log_));
test::NullTransport transport;
CreateSendConfig(1, 0, &transport);
@@ -72,8 +71,7 @@
}
TEST_F(VideoSendStreamTest, CanStopStoppedStream) {
- Call::Config call_config;
- CreateSenderCall(call_config);
+ CreateSenderCall(Call::Config(&event_log_));
test::NullTransport transport;
CreateSendConfig(1, 0, &transport);
@@ -735,7 +733,7 @@
}
Call::Config GetSenderCallConfig() override {
- Call::Config config;
+ Call::Config config(&event_log_);
const int kMinBitrateBps = 30000;
config.bitrate_config.min_bitrate_bps = kMinBitrateBps;
return config;
@@ -1438,7 +1436,7 @@
int last_initialized_frame_height_ GUARDED_BY(&crit_);
};
- CreateSenderCall(Call::Config());
+ CreateSenderCall(Call::Config(&event_log_));
test::NullTransport transport;
CreateSendConfig(1, 0, &transport);
EncoderObserver encoder;
@@ -1494,7 +1492,7 @@
int start_bitrate_kbps_ GUARDED_BY(crit_);
};
- CreateSenderCall(Call::Config());
+ CreateSenderCall(Call::Config(&event_log_));
test::NullTransport transport;
CreateSendConfig(1, 0, &transport);
@@ -1575,7 +1573,7 @@
int bitrate_kbps_ GUARDED_BY(crit_);
};
- CreateSenderCall(Call::Config());
+ CreateSenderCall(Call::Config(&event_log_));
test::NullTransport transport;
CreateSendConfig(1, 0, &transport);
@@ -1630,7 +1628,7 @@
};
// Initialize send stream.
- CreateSenderCall(Call::Config());
+ CreateSenderCall(Call::Config(&event_log_));
test::NullTransport transport;
CreateSendConfig(1, 0, &transport);
@@ -2269,7 +2267,7 @@
}
Call::Config GetSenderCallConfig() override {
- Call::Config config;
+ Call::Config config(&event_log_);
config.bitrate_config.min_bitrate_bps = kMinBitrateKbps * 1000;
config.bitrate_config.start_bitrate_bps = kStartBitrateKbps * 1000;
config.bitrate_config.max_bitrate_bps = kMaxBitrateKbps * 1000;