Move RtcEventLog object from inside VoiceEngine to Call.
In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
BUG=webrtc:4741,webrtc:5603,chromium:609749
R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1748403002 .
Cr-Commit-Position: refs/heads/master@{#13321}
diff --git a/webrtc/api/java/src/org/webrtc/PeerConnection.java b/webrtc/api/java/src/org/webrtc/PeerConnection.java
index ad8362d..3808eb4 100644
--- a/webrtc/api/java/src/org/webrtc/PeerConnection.java
+++ b/webrtc/api/java/src/org/webrtc/PeerConnection.java
@@ -253,6 +253,23 @@
return nativeGetStats(observer, (track == null) ? 0 : track.nativeTrack);
}
+ // Starts recording an RTC event log. Ownership of the file is transfered to
+ // the native code. If an RTC event log is already being recorded, it will be
+ // stopped and a new one will start using the provided file. Logging will
+ // continue until the stopRtcEventLog function is called. The max_size_bytes
+ // argument is ignored, it is added for future use.
+ public boolean startRtcEventLog(
+ int file_descriptor, long max_size_bytes) {
+ return nativeStartRtcEventLog(
+ nativePeerConnection, file_descriptor, max_size_bytes);
+ }
+
+ // Stops recording an RTC event log. If no RTC event log is currently being
+ // recorded, this call will have no effect.
+ public void stopRtcEventLog() {
+ nativeStopRtcEventLog(nativePeerConnection);
+ }
+
// TODO(fischman): add support for DTMF-related methods once that API
// stabilizes.
public native SignalingState signalingState();
@@ -303,4 +320,10 @@
private native List<RtpSender> nativeGetSenders();
private native List<RtpReceiver> nativeGetReceivers();
+
+ private static native boolean nativeStartRtcEventLog(
+ long nativePeerConnection, int file_descriptor, long max_size_bytes);
+
+ private static native void nativeStopRtcEventLog(long nativePeerConnection);
+
}
diff --git a/webrtc/api/java/src/org/webrtc/PeerConnectionFactory.java b/webrtc/api/java/src/org/webrtc/PeerConnectionFactory.java
index 0c1ef3c..2edd56e 100644
--- a/webrtc/api/java/src/org/webrtc/PeerConnectionFactory.java
+++ b/webrtc/api/java/src/org/webrtc/PeerConnectionFactory.java
@@ -148,28 +148,6 @@
nativeStopAecDump(nativeFactory);
}
- // Starts recording an RTC event log. Ownership of the file is transfered to
- // the native code. If an RTC event log is already being recorded, it will be
- // stopped and a new one will start using the provided file.
- public boolean startRtcEventLog(int file_descriptor) {
- return startRtcEventLog(file_descriptor, -1);
- }
-
- // Same as above, but allows setting an upper limit to the size of the
- // generated logfile.
- public boolean startRtcEventLog(int file_descriptor,
- int filesize_limit_bytes) {
- return nativeStartRtcEventLog(nativeFactory,
- file_descriptor,
- filesize_limit_bytes);
- }
-
- // Stops recording an RTC event log. If no RTC event log is currently being
- // recorded, this call will have no effect.
- public void stopRtcEventLog() {
- nativeStopRtcEventLog(nativeFactory);
- }
-
@Deprecated
public void setOptions(Options options) {
nativeSetOptions(nativeFactory, options);
@@ -275,12 +253,6 @@
private static native void nativeStopAecDump(long nativeFactory);
- private static native boolean nativeStartRtcEventLog(long nativeFactory,
- int file_descriptor,
- int filesize_limit_bytes);
-
- private static native void nativeStopRtcEventLog(long nativeFactory);
-
@Deprecated
public native void nativeSetOptions(long nativeFactory, Options options);
diff --git a/webrtc/api/peerconnection.cc b/webrtc/api/peerconnection.cc
index e0b4fca..a337d00 100644
--- a/webrtc/api/peerconnection.cc
+++ b/webrtc/api/peerconnection.cc
@@ -31,10 +31,12 @@
#include "webrtc/api/videocapturertracksource.h"
#include "webrtc/api/videotrack.h"
#include "webrtc/base/arraysize.h"
+#include "webrtc/base/bind.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/trace_event.h"
+#include "webrtc/call.h"
#include "webrtc/media/sctp/sctpdataengine.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/system_wrappers/include/field_trial.h"
@@ -1263,6 +1265,18 @@
}
}
+bool PeerConnection::StartRtcEventLog(rtc::PlatformFile file,
+ int64_t max_size_bytes) {
+ return factory_->worker_thread()->Invoke<bool>(
+ RTC_FROM_HERE, rtc::Bind(&PeerConnection::StartRtcEventLog_w, this, file,
+ max_size_bytes));
+}
+
+void PeerConnection::StopRtcEventLog() {
+ factory_->worker_thread()->Invoke<void>(
+ RTC_FROM_HERE, rtc::Bind(&PeerConnection::StopRtcEventLog_w, this));
+}
+
const SessionDescriptionInterface* PeerConnection::local_description() const {
return session_->local_description();
}
@@ -2232,4 +2246,12 @@
return true;
}
+bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
+ int64_t max_size_bytes) {
+ return media_controller_->call_w()->StartEventLog(file, max_size_bytes);
+}
+
+void PeerConnection::StopRtcEventLog_w() {
+ media_controller_->call_w()->StopEventLog();
+}
} // namespace webrtc
diff --git a/webrtc/api/peerconnection.h b/webrtc/api/peerconnection.h
index fc754b9..c4a9a60 100644
--- a/webrtc/api/peerconnection.h
+++ b/webrtc/api/peerconnection.h
@@ -136,6 +136,10 @@
void RegisterUMAObserver(UMAObserver* observer) override;
+ bool StartRtcEventLog(rtc::PlatformFile file,
+ int64_t max_size_bytes) override;
+ void StopRtcEventLog() override;
+
void Close() override;
// Virtual for unit tests.
@@ -360,6 +364,13 @@
// is applied.
bool ReconfigurePortAllocator_n(const RTCConfiguration& configuration);
+ // Starts recording an Rtc EventLog using the supplied platform file.
+ // This function should only be called from the worker thread.
+ bool StartRtcEventLog_w(rtc::PlatformFile file, int64_t max_size_bytes);
+ // Starts recording an Rtc EventLog using the supplied platform file.
+ // This function should only be called from the worker thread.
+ void StopRtcEventLog_w();
+
// Storing the factory as a scoped reference pointer ensures that the memory
// in the PeerConnectionFactoryImpl remains available as long as the
// PeerConnection is running. It is passed to PeerConnection as a raw pointer.
diff --git a/webrtc/api/peerconnectionfactory.cc b/webrtc/api/peerconnectionfactory.cc
index 745de3f..b79e7a2 100644
--- a/webrtc/api/peerconnectionfactory.cc
+++ b/webrtc/api/peerconnectionfactory.cc
@@ -220,17 +220,6 @@
channel_manager_->StopAecDump();
}
-bool PeerConnectionFactory::StartRtcEventLog(rtc::PlatformFile file,
- int64_t max_size_bytes) {
- RTC_DCHECK(signaling_thread_->IsCurrent());
- return channel_manager_->StartRtcEventLog(file, max_size_bytes);
-}
-
-void PeerConnectionFactory::StopRtcEventLog() {
- RTC_DCHECK(signaling_thread_->IsCurrent());
- channel_manager_->StopRtcEventLog();
-}
-
rtc::scoped_refptr<PeerConnectionInterface>
PeerConnectionFactory::CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration_in,
diff --git a/webrtc/api/peerconnectionfactory.h b/webrtc/api/peerconnectionfactory.h
index d4bc0da..39a6402 100644
--- a/webrtc/api/peerconnectionfactory.h
+++ b/webrtc/api/peerconnectionfactory.h
@@ -80,12 +80,15 @@
bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
void StopAecDump() override;
- bool StartRtcEventLog(rtc::PlatformFile file) override {
- return StartRtcEventLog(file, -1);
- }
+ // TODO(ivoc) Remove after Chrome is updated.
+ bool StartRtcEventLog(rtc::PlatformFile file) override { return false; }
+ // TODO(ivoc) Remove after Chrome is updated.
bool StartRtcEventLog(rtc::PlatformFile file,
- int64_t max_size_bytes) override;
- void StopRtcEventLog() override;
+ int64_t max_size_bytes) override {
+ return false;
+ }
+ // TODO(ivoc) Remove after Chrome is updated.
+ void StopRtcEventLog() override {}
virtual webrtc::MediaControllerInterface* CreateMediaController(
const cricket::MediaConfig& config) const;
diff --git a/webrtc/api/peerconnectionfactoryproxy.h b/webrtc/api/peerconnectionfactoryproxy.h
index aad97e8..227a685 100644
--- a/webrtc/api/peerconnectionfactoryproxy.h
+++ b/webrtc/api/peerconnectionfactoryproxy.h
@@ -70,6 +70,8 @@
CreateAudioTrack, const std::string&, AudioSourceInterface*)
PROXY_METHOD2(bool, StartAecDump, rtc::PlatformFile, int64_t)
PROXY_METHOD0(void, StopAecDump)
+ // TODO(ivoc): Remove the StartRtcEventLog and StopRtcEventLog functions as
+ // soon as they are removed from PeerConnectionFactoryInterface.
PROXY_METHOD1(bool, StartRtcEventLog, rtc::PlatformFile)
PROXY_METHOD2(bool, StartRtcEventLog, rtc::PlatformFile, int64_t)
PROXY_METHOD0(void, StopRtcEventLog)
diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h
index 0ac37e1..e574355 100644
--- a/webrtc/api/peerconnectioninterface.h
+++ b/webrtc/api/peerconnectioninterface.h
@@ -489,6 +489,17 @@
virtual IceConnectionState ice_connection_state() = 0;
virtual IceGatheringState ice_gathering_state() = 0;
+ // Starts RtcEventLog using existing file. Takes ownership of |file| and
+ // passes it on to Call, which will take the ownership. If the
+ // operation fails the file will be closed. The logging will stop
+ // automatically after 10 minutes have passed, or when the StopRtcEventLog
+ // function is called.
+ virtual bool StartRtcEventLog(rtc::PlatformFile file,
+ int64_t max_size_bytes) = 0;
+
+ // Stops logging the RtcEventLog.
+ virtual void StopRtcEventLog() = 0;
+
// Terminates all media and closes the transport.
virtual void Close() = 0;
@@ -655,25 +666,19 @@
// Stops logging the AEC dump.
virtual void StopAecDump() = 0;
- // Starts RtcEventLog using existing file. Takes ownership of |file| and
- // passes it on to VoiceEngine, which will take the ownership. If the
- // operation fails the file will be closed. The logging will stop
- // automatically after 10 minutes have passed, or when the StopRtcEventLog
- // function is called. A maximum filesize in bytes can be set, the logging
- // will be stopped before exceeding this limit. If max_size_bytes is set to a
- // value <= 0, no limit will be used.
- // This function as well as the StopRtcEventLog don't really belong on this
- // interface, this is a temporary solution until we move the logging object
- // from inside voice engine to webrtc::Call, which will happen when the VoE
- // restructuring effort is further along.
- // TODO(ivoc): Move this into being:
- // PeerConnection => MediaController => webrtc::Call.
+ // This function is deprecated and will be removed when Chrome is updated to
+ // use the equivalent function on PeerConnectionInterface.
+ // TODO(ivoc) Remove after Chrome is updated.
virtual bool StartRtcEventLog(rtc::PlatformFile file,
int64_t max_size_bytes) = 0;
- // Deprecated, use the version above.
+ // This function is deprecated and will be removed when Chrome is updated to
+ // use the equivalent function on PeerConnectionInterface.
+ // TODO(ivoc) Remove after Chrome is updated.
virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
- // Stops logging the RtcEventLog.
+ // This function is deprecated and will be removed when Chrome is updated to
+ // use the equivalent function on PeerConnectionInterface.
+ // TODO(ivoc) Remove after Chrome is updated.
virtual void StopRtcEventLog() = 0;
protected:
diff --git a/webrtc/api/peerconnectionproxy.h b/webrtc/api/peerconnectionproxy.h
index d35d5ba..37f2e89 100644
--- a/webrtc/api/peerconnectionproxy.h
+++ b/webrtc/api/peerconnectionproxy.h
@@ -74,6 +74,8 @@
PROXY_METHOD0(IceState, ice_state)
PROXY_METHOD0(IceConnectionState, ice_connection_state)
PROXY_METHOD0(IceGatheringState, ice_gathering_state)
+ PROXY_METHOD2(bool, StartRtcEventLog, rtc::PlatformFile, int64_t)
+ PROXY_METHOD0(void, StopRtcEventLog)
PROXY_METHOD0(void, Close)
END_SIGNALING_PROXY()
diff --git a/webrtc/audio/DEPS b/webrtc/audio/DEPS
index b9605bb..82c3165 100644
--- a/webrtc/audio/DEPS
+++ b/webrtc/audio/DEPS
@@ -9,3 +9,12 @@
"+webrtc/modules/rtp_rtcp",
"+webrtc/system_wrappers",
]
+
+specific_include_rules = {
+ "audio_receive_stream_unittest\.cc": [
+ "+webrtc/call/mock",
+ ],
+ "audio_send_stream_unittest\.cc": [
+ "+webrtc/call/mock",
+ ],
+}
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index a684003..0a2bc2b 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -81,7 +81,8 @@
AudioReceiveStream::AudioReceiveStream(
CongestionController* congestion_controller,
const webrtc::AudioReceiveStream::Config& config,
- const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
+ const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+ webrtc::RtcEventLog* event_log)
: config_(config),
audio_state_(audio_state),
rtp_header_parser_(RtpHeaderParser::Create()) {
@@ -93,6 +94,7 @@
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
+ channel_proxy_->SetRtcEventLog(event_log);
channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
// TODO(solenberg): Config NACK history window (which is a packet count),
// using the actual packet size for the configured codec.
@@ -144,6 +146,7 @@
LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
channel_proxy_->DeRegisterExternalTransport();
channel_proxy_->ResetCongestionControlObjects();
+ channel_proxy_->SetRtcEventLog(nullptr);
if (remote_bitrate_estimator_) {
remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
}
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index 284f98e..24924c9a 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -22,6 +22,7 @@
namespace webrtc {
class CongestionController;
class RemoteBitrateEstimator;
+class RtcEventLog;
namespace voe {
class ChannelProxy;
@@ -33,7 +34,8 @@
public:
AudioReceiveStream(CongestionController* congestion_controller,
const webrtc::AudioReceiveStream::Config& config,
- const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
+ const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+ webrtc::RtcEventLog* event_log);
~AudioReceiveStream() override;
// webrtc::AudioReceiveStream implementation.
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index aed1d1a..dd66cc6 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -15,6 +15,7 @@
#include "webrtc/audio/audio_receive_stream.h"
#include "webrtc/audio/conversion.h"
+#include "webrtc/call/mock/mock_rtc_event_log.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
@@ -70,7 +71,8 @@
decoder_factory_(new rtc::RefCountedObject<MockAudioDecoderFactory>),
congestion_controller_(&simulated_clock_,
&bitrate_observer_,
- &remote_bitrate_observer_) {
+ &remote_bitrate_observer_,
+ &event_log_) {
using testing::Invoke;
EXPECT_CALL(voice_engine_,
@@ -109,6 +111,12 @@
.Times(1);
EXPECT_CALL(*channel_proxy_, GetAudioDecoderFactory())
.WillOnce(ReturnRef(decoder_factory_));
+ testing::Expectation expect_set =
+ EXPECT_CALL(*channel_proxy_, SetRtcEventLog(&event_log_))
+ .Times(1);
+ EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
+ .Times(1)
+ .After(expect_set);
return channel_proxy_;
}));
stream_config_.voe_channel_id = kChannelId;
@@ -130,6 +138,7 @@
MockRemoteBitrateEstimator* remote_bitrate_estimator() {
return &remote_bitrate_estimator_;
}
+ MockRtcEventLog* event_log() { return &event_log_; }
AudioReceiveStream::Config& config() { return stream_config_; }
rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
MockVoiceEngine& voice_engine() { return voice_engine_; }
@@ -171,6 +180,7 @@
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
MockCongestionController congestion_controller_;
MockRemoteBitrateEstimator remote_bitrate_estimator_;
+ MockRtcEventLog event_log_;
testing::StrictMock<MockVoiceEngine> voice_engine_;
rtc::scoped_refptr<AudioState> audio_state_;
AudioReceiveStream::Config stream_config_;
@@ -248,7 +258,8 @@
TEST(AudioReceiveStreamTest, ConstructDestruct) {
ConfigHelper helper;
internal::AudioReceiveStream recv_stream(
- helper.congestion_controller(), helper.config(), helper.audio_state());
+ helper.congestion_controller(), helper.config(), helper.audio_state(),
+ helper.event_log());
}
MATCHER_P(VerifyHeaderExtension, expected_extension, "") {
@@ -267,7 +278,8 @@
helper.config().rtp.transport_cc = true;
helper.SetupMockForBweFeedback(true);
internal::AudioReceiveStream recv_stream(
- helper.congestion_controller(), helper.config(), helper.audio_state());
+ helper.congestion_controller(), helper.config(), helper.audio_state(),
+ helper.event_log());
const int kTransportSequenceNumberValue = 1234;
std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
@@ -295,7 +307,8 @@
helper.config().rtp.transport_cc = true;
helper.SetupMockForBweFeedback(true);
internal::AudioReceiveStream recv_stream(
- helper.congestion_controller(), helper.config(), helper.audio_state());
+ helper.congestion_controller(), helper.config(), helper.audio_state(),
+ helper.event_log());
std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport();
EXPECT_CALL(*helper.channel_proxy(),
@@ -307,7 +320,8 @@
TEST(AudioReceiveStreamTest, GetStats) {
ConfigHelper helper;
internal::AudioReceiveStream recv_stream(
- helper.congestion_controller(), helper.config(), helper.audio_state());
+ helper.congestion_controller(), helper.config(), helper.audio_state(),
+ helper.event_log());
helper.SetupMockForGetStats();
AudioReceiveStream::Stats stats = recv_stream.GetStats();
EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
@@ -349,7 +363,8 @@
TEST(AudioReceiveStreamTest, SetGain) {
ConfigHelper helper;
internal::AudioReceiveStream recv_stream(
- helper.congestion_controller(), helper.config(), helper.audio_state());
+ helper.congestion_controller(), helper.config(), helper.audio_state(),
+ helper.event_log());
EXPECT_CALL(*helper.channel_proxy(),
SetChannelOutputVolumeScaling(FloatEq(0.765f)));
recv_stream.SetGain(0.765f);
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index b953554..ea70d30 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -17,6 +17,7 @@
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
+#include "webrtc/call/mock/mock_rtc_event_log.h"
#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
@@ -55,7 +56,8 @@
stream_config_(nullptr),
congestion_controller_(&simulated_clock_,
&bitrate_observer_,
- &remote_bitrate_observer_) {
+ &remote_bitrate_observer_,
+ &event_log_) {
using testing::Invoke;
using testing::StrEq;
@@ -167,6 +169,7 @@
testing::NiceMock<MockCongestionObserver> bitrate_observer_;
testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
CongestionController congestion_controller_;
+ MockRtcEventLog event_log_;
};
} // namespace
diff --git a/webrtc/call.h b/webrtc/call.h
index f89af93..ff20a1e 100644
--- a/webrtc/call.h
+++ b/webrtc/call.h
@@ -18,6 +18,7 @@
#include "webrtc/audio_send_stream.h"
#include "webrtc/audio_state.h"
#include "webrtc/base/networkroute.h"
+#include "webrtc/base/platform_file.h"
#include "webrtc/base/socket.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
@@ -147,6 +148,10 @@
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/call.cc b/webrtc/call/call.cc
index a6a7978..cbf9da7 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -35,6 +35,7 @@
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/cpu_info.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/metrics.h"
@@ -107,6 +108,13 @@
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);
@@ -162,7 +170,7 @@
VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
- RtcEventLog* event_log_ = nullptr;
+ std::unique_ptr<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
@@ -210,6 +218,7 @@
video_network_state_(kNetworkUp),
receive_crit_(RWLockWrapper::CreateRWLock()),
send_crit_(RWLockWrapper::CreateRWLock()),
+ event_log_(RtcEventLog::Create(webrtc::Clock::GetRealTimeClock())),
received_video_bytes_(0),
received_audio_bytes_(0),
received_rtcp_bytes_(0),
@@ -221,7 +230,8 @@
min_allocated_send_bitrate_bps_(0),
num_bitrate_updates_(0),
remb_(clock_),
- congestion_controller_(new CongestionController(clock_, this, &remb_)),
+ congestion_controller_(
+ new CongestionController(clock_, this, &remb_, event_log_.get())),
video_send_delay_stats_(new SendDelayStats(clock_)) {
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
@@ -231,10 +241,6 @@
RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
config.bitrate_config.start_bitrate_bps);
}
- if (config.audio_state.get()) {
- ScopedVoEInterface<VoECodec> voe_codec(voice_engine());
- event_log_ = voe_codec->GetEventLog();
- }
Trace::CreateTrace();
call_stats_->RegisterStatsObserver(congestion_controller_.get());
@@ -243,7 +249,6 @@
config_.bitrate_config.min_bitrate_bps,
config_.bitrate_config.start_bitrate_bps,
config_.bitrate_config.max_bitrate_bps);
- congestion_controller_->GetBitrateController()->SetEventLog(event_log_);
module_process_thread_->Start();
module_process_thread_->RegisterModule(call_stats_.get());
@@ -371,8 +376,9 @@
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);
+ AudioReceiveStream* receive_stream =
+ new AudioReceiveStream(congestion_controller_.get(), config,
+ config_.audio_state, event_log_.get());
{
WriteLockScoped write_lock(*receive_crit_);
RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
@@ -421,8 +427,8 @@
VideoSendStream* send_stream = new VideoSendStream(
num_cpu_cores_, module_process_thread_.get(), call_stats_.get(),
congestion_controller_.get(), bitrate_allocator_.get(),
- video_send_delay_stats_.get(), &remb_, event_log_, config, encoder_config,
- suspended_video_send_ssrcs_);
+ video_send_delay_stats_.get(), &remb_, event_log_.get(), config,
+ encoder_config, suspended_video_send_ssrcs_);
{
WriteLockScoped write_lock(*send_crit_);
for (uint32_t ssrc : config.rtp.ssrcs) {
@@ -433,8 +439,7 @@
}
send_stream->SignalNetworkState(video_network_state_);
UpdateAggregateNetworkState();
- if (event_log_)
- event_log_->LogVideoSendStreamConfig(config);
+ event_log_->LogVideoSendStreamConfig(config);
return send_stream;
}
@@ -493,13 +498,11 @@
if (it != config.rtp.rtx.end())
video_receive_ssrcs_[it->second.ssrc] = receive_stream;
video_receive_streams_.insert(receive_stream);
-
ConfigureSync(config.sync_group);
}
receive_stream->SignalNetworkState(video_network_state_);
UpdateAggregateNetworkState();
- if (event_log_)
- event_log_->LogVideoReceiveStreamConfig(config);
+ event_log_->LogVideoReceiveStreamConfig(config);
return receive_stream;
}
@@ -827,7 +830,7 @@
auto status = it->second->DeliverRtp(packet, length, packet_time)
? DELIVERY_OK
: DELIVERY_PACKET_ERROR;
- if (status == DELIVERY_OK && event_log_)
+ if (status == DELIVERY_OK)
event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
return status;
}
@@ -839,7 +842,7 @@
auto status = it->second->DeliverRtp(packet, length, packet_time)
? DELIVERY_OK
: DELIVERY_PACKET_ERROR;
- if (status == DELIVERY_OK && event_log_)
+ if (status == DELIVERY_OK)
event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
return status;
}
diff --git a/webrtc/call/rtc_event_log.cc b/webrtc/call/rtc_event_log.cc
index 7ea1e16..840b210 100644
--- a/webrtc/call/rtc_event_log.cc
+++ b/webrtc/call/rtc_event_log.cc
@@ -38,9 +38,7 @@
namespace webrtc {
-#ifndef ENABLE_RTC_EVENT_LOG
-
-// No-op implementation if flag is not set.
+// No-op implementation is used if flag is not set, or in tests.
class RtcEventLogNullImpl final : public RtcEventLog {
public:
bool StartLogging(const std::string& file_name,
@@ -74,7 +72,7 @@
int32_t total_packets) override {}
};
-#else // ENABLE_RTC_EVENT_LOG is defined
+#ifdef ENABLE_RTC_EVENT_LOG
class RtcEventLogImpl final : public RtcEventLog {
public:
@@ -465,4 +463,8 @@
#endif // ENABLE_RTC_EVENT_LOG
}
+std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() {
+ return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
+}
+
} // namespace webrtc
diff --git a/webrtc/call/rtc_event_log.h b/webrtc/call/rtc_event_log.h
index bea57b0..7c72dd5 100644
--- a/webrtc/call/rtc_event_log.h
+++ b/webrtc/call/rtc_event_log.h
@@ -40,6 +40,9 @@
// Factory method to create an RtcEventLog object.
static std::unique_ptr<RtcEventLog> Create(const Clock* clock);
+ // Create an RtcEventLog object that does nothing.
+ static std::unique_ptr<RtcEventLog> CreateNull();
+
// Starts logging a maximum of max_size_bytes bytes to the specified file.
// If the file already exists it will be overwritten.
// If max_size_bytes <= 0, logging will be active until StopLogging is called.
diff --git a/webrtc/media/base/mediaengine.h b/webrtc/media/base/mediaengine.h
index 0e2b7ef..dd1f527 100644
--- a/webrtc/media/base/mediaengine.h
+++ b/webrtc/media/base/mediaengine.h
@@ -88,15 +88,6 @@
// Stops recording AEC dump.
virtual void StopAecDump() = 0;
-
- // Starts RtcEventLog using existing file. A maximum file size in bytes can be
- // specified. Logging is stopped just before the size limit is exceeded.
- // If max_size_bytes is set to a value <= 0, no limit will be used.
- virtual bool StartRtcEventLog(rtc::PlatformFile file,
- int64_t max_size_bytes) = 0;
-
- // Stops recording an RtcEventLog.
- virtual void StopRtcEventLog() = 0;
};
@@ -175,13 +166,6 @@
voice_.StopAecDump();
}
- virtual bool StartRtcEventLog(rtc::PlatformFile file,
- int64_t max_size_bytes) {
- return voice_.StartRtcEventLog(file, max_size_bytes);
- }
-
- virtual void StopRtcEventLog() { voice_.StopRtcEventLog(); }
-
protected:
VOICE voice_;
VIDEO video_;
diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
index 914a403..fdf7cf3 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -478,4 +478,11 @@
}
}
+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 f703b15..a2ac079 100644
--- a/webrtc/media/engine/fakewebrtccall.h
+++ b/webrtc/media/engine/fakewebrtccall.h
@@ -231,6 +231,10 @@
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/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
index 4e3ced6..e745f13 100644
--- a/webrtc/media/engine/fakewebrtcvoiceengine.h
+++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
@@ -272,7 +272,6 @@
channels_[channel]->associate_send_channel = accociate_send_channel;
return 0;
}
- webrtc::RtcEventLog* GetEventLog() override { return nullptr; }
// webrtc::VoECodec
WEBRTC_STUB(NumOfCodecs, ());
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 2ddf67d..d329270 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -28,7 +28,6 @@
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/trace_event.h"
-#include "webrtc/call/rtc_event_log.h"
#include "webrtc/common.h"
#include "webrtc/media/base/audiosource.h"
#include "webrtc/media/base/mediaconstants.h"
@@ -1040,27 +1039,6 @@
}
}
-bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file,
- int64_t max_size_bytes) {
- RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog();
- if (event_log) {
- return event_log->StartLogging(file, max_size_bytes);
- }
- LOG_RTCERR0(StartRtcEventLog);
- return false;
-}
-
-void WebRtcVoiceEngine::StopRtcEventLog() {
- RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog();
- if (event_log) {
- event_log->StopLogging();
- return;
- }
- LOG_RTCERR0(StopRtcEventLog);
-}
-
int WebRtcVoiceEngine::CreateVoEChannel() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return voe_wrapper_->base()->CreateChannel(voe_config_);
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index 52983fa..16da9ef 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -109,14 +109,6 @@
// Stops AEC dump.
void StopAecDump();
- // Starts recording an RtcEventLog using an existing file until the log file
- // reaches the maximum filesize or the StopRtcEventLog function is called.
- // If the value of max_size_bytes is <= 0, no limit is used.
- bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes);
-
- // Stops recording the RtcEventLog.
- void StopRtcEventLog();
-
private:
// Every option that is "set" will be applied. Every option not "set" will be
// ignored. This allows us to selectively turn on and off different options
diff --git a/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc b/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
index c99bd5321..8a2464d 100644
--- a/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
+++ b/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
@@ -79,20 +79,25 @@
BitrateController* BitrateController::CreateBitrateController(
Clock* clock,
- BitrateObserver* observer) {
- return new BitrateControllerImpl(clock, observer);
+ BitrateObserver* observer,
+ RtcEventLog* event_log) {
+ return new BitrateControllerImpl(clock, observer, event_log);
}
-BitrateController* BitrateController::CreateBitrateController(Clock* clock) {
- return new BitrateControllerImpl(clock, nullptr);
+BitrateController* BitrateController::CreateBitrateController(
+ Clock* clock,
+ RtcEventLog* event_log) {
+ return CreateBitrateController(clock, nullptr, event_log);
}
BitrateControllerImpl::BitrateControllerImpl(Clock* clock,
- BitrateObserver* observer)
+ BitrateObserver* observer,
+ RtcEventLog* event_log)
: clock_(clock),
observer_(observer),
last_bitrate_update_ms_(clock_->TimeInMilliseconds()),
- bandwidth_estimation_(),
+ event_log_(event_log),
+ bandwidth_estimation_(event_log),
reserved_bitrate_bps_(0),
last_bitrate_bps_(0),
last_fraction_loss_(0),
@@ -143,7 +148,7 @@
int max_bitrate_bps) {
{
rtc::CritScope cs(&critsect_);
- bandwidth_estimation_ = SendSideBandwidthEstimation();
+ bandwidth_estimation_ = SendSideBandwidthEstimation(event_log_);
bandwidth_estimation_.SetBitrates(bitrate_bps, min_bitrate_bps,
max_bitrate_bps);
}
@@ -158,11 +163,6 @@
MaybeTriggerOnNetworkChanged();
}
-void BitrateControllerImpl::SetEventLog(RtcEventLog* event_log) {
- rtc::CritScope cs(&critsect_);
- bandwidth_estimation_.SetEventLog(event_log);
-}
-
void BitrateControllerImpl::OnReceivedEstimatedBitrate(uint32_t bitrate) {
{
rtc::CritScope cs(&critsect_);
diff --git a/webrtc/modules/bitrate_controller/bitrate_controller_impl.h b/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
index 91f0902..114e1a6 100644
--- a/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
+++ b/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
@@ -30,7 +30,9 @@
public:
// TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
// |observer| is left for project that is not yet updated.
- BitrateControllerImpl(Clock* clock, BitrateObserver* observer);
+ BitrateControllerImpl(Clock* clock,
+ BitrateObserver* observer,
+ RtcEventLog* event_log);
virtual ~BitrateControllerImpl() {}
bool AvailableBandwidth(uint32_t* bandwidth) const override;
@@ -54,8 +56,6 @@
void SetReservedBitrate(uint32_t reserved_bitrate_bps) override;
- void SetEventLog(RtcEventLog* event_log) override;
-
// Returns true if the parameters have changed since the last call.
bool GetNetworkParameters(uint32_t* bitrate,
uint8_t* fraction_loss,
@@ -86,6 +86,7 @@
Clock* const clock_;
BitrateObserver* const observer_;
int64_t last_bitrate_update_ms_;
+ RtcEventLog* const event_log_;
rtc::CriticalSection critsect_;
SendSideBandwidthEstimation bandwidth_estimation_ GUARDED_BY(critsect_);
diff --git a/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc b/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
index 4f92a38..f2a14ea 100644
--- a/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
+++ b/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
@@ -13,6 +13,7 @@
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/call/mock/mock_rtc_event_log.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/pacing/mock/mock_paced_sender.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -66,8 +67,8 @@
~BitrateControllerTest() {}
virtual void SetUp() {
- controller_ =
- BitrateController::CreateBitrateController(&clock_, &bitrate_observer_);
+ controller_ = BitrateController::CreateBitrateController(
+ &clock_, &bitrate_observer_, &event_log_);
controller_->SetStartBitrate(kStartBitrateBps);
EXPECT_EQ(kStartBitrateBps, bitrate_observer_.last_bitrate_);
controller_->SetMinMaxBitrate(kMinBitrateBps, kMaxBitrateBps);
@@ -91,6 +92,7 @@
TestBitrateObserver bitrate_observer_;
BitrateController* controller_;
RtcpBandwidthObserver* bandwidth_observer_;
+ webrtc::MockRtcEventLog event_log_;
};
TEST_F(BitrateControllerTest, DefaultMinMaxBitrate) {
@@ -107,6 +109,7 @@
TEST_F(BitrateControllerTest, OneBitrateObserverOneRtcpObserver) {
// First REMB applies immediately.
+ EXPECT_CALL(event_log_, LogBwePacketLossEvent(testing::Gt(0), 0, 0)).Times(8);
int64_t time_ms = 1001;
webrtc::ReportBlockList report_blocks;
report_blocks.push_back(CreateReportBlock(1, 2, 0, 1));
@@ -183,6 +186,7 @@
TEST_F(BitrateControllerTest, OneBitrateObserverTwoRtcpObservers) {
// REMBs during the first 2 seconds apply immediately.
+ EXPECT_CALL(event_log_, LogBwePacketLossEvent(testing::Gt(0), 0, 0)).Times(9);
int64_t time_ms = 1;
webrtc::ReportBlockList report_blocks;
report_blocks.push_back(CreateReportBlock(1, 2, 0, 1));
@@ -278,6 +282,13 @@
}
TEST_F(BitrateControllerTest, OneBitrateObserverMultipleReportBlocks) {
+ testing::Expectation first_calls =
+ EXPECT_CALL(event_log_, LogBwePacketLossEvent(testing::Gt(0), 0, 0))
+ .Times(7);
+ EXPECT_CALL(event_log_,
+ LogBwePacketLossEvent(testing::Gt(0), testing::Gt(0), 0))
+ .Times(2)
+ .After(first_calls);
uint32_t sequence_number[2] = {0, 0xFF00};
const int kStartBitrate = 200000;
const int kMinBitrate = 100000;
diff --git a/webrtc/modules/bitrate_controller/include/bitrate_controller.h b/webrtc/modules/bitrate_controller/include/bitrate_controller.h
index 4165f06..f06314c 100644
--- a/webrtc/modules/bitrate_controller/include/bitrate_controller.h
+++ b/webrtc/modules/bitrate_controller/include/bitrate_controller.h
@@ -55,8 +55,11 @@
// TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
// Remove this method once other other projects does not use it.
static BitrateController* CreateBitrateController(Clock* clock,
- BitrateObserver* observer);
- static BitrateController* CreateBitrateController(Clock* clock);
+ BitrateObserver* observer,
+ RtcEventLog* event_log);
+
+ static BitrateController* CreateBitrateController(Clock* clock,
+ RtcEventLog* event_log);
virtual ~BitrateController() {}
@@ -76,8 +79,6 @@
virtual void UpdateDelayBasedEstimate(uint32_t bitrate_bps) = 0;
- virtual void SetEventLog(RtcEventLog* event_log) = 0;
-
// Gets the available payload bandwidth in bits per second. Note that
// this bandwidth excludes packet headers.
virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
index a1b78a2..e47d491 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
@@ -44,7 +44,7 @@
} // namespace
-SendSideBandwidthEstimation::SendSideBandwidthEstimation()
+SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)
: lost_packets_since_last_loss_update_Q8_(0),
expected_packets_since_last_loss_update_(0),
bitrate_(0),
@@ -63,7 +63,9 @@
bitrate_at_2_seconds_kbps_(0),
uma_update_state_(kNoUpdate),
rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
- event_log_(nullptr) {}
+ event_log_(event_log) {
+ RTC_DCHECK(event_log);
+}
SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
@@ -227,11 +229,9 @@
// rates).
bitrate_ += 1000;
- if (event_log_) {
- event_log_->LogBwePacketLossEvent(
- bitrate_, last_fraction_loss_,
- expected_packets_since_last_loss_update_);
- }
+ event_log_->LogBwePacketLossEvent(
+ bitrate_, last_fraction_loss_,
+ expected_packets_since_last_loss_update_);
} else if (last_fraction_loss_ <= 26) {
// Loss between 2% - 10%: Do nothing.
} else {
@@ -250,11 +250,9 @@
512.0);
has_decreased_since_last_fraction_loss_ = true;
}
- if (event_log_) {
- event_log_->LogBwePacketLossEvent(
- bitrate_, last_fraction_loss_,
- expected_packets_since_last_loss_update_);
- }
+ event_log_->LogBwePacketLossEvent(
+ bitrate_, last_fraction_loss_,
+ expected_packets_since_last_loss_update_);
}
}
bitrate_ = CapBitrateToThresholds(now_ms, bitrate_);
@@ -308,9 +306,4 @@
}
return bitrate;
}
-
-void SendSideBandwidthEstimation::SetEventLog(RtcEventLog* event_log) {
- event_log_ = event_log;
-}
-
} // namespace webrtc
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
index 402d22a..c97714f 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
@@ -26,7 +26,8 @@
class SendSideBandwidthEstimation {
public:
- SendSideBandwidthEstimation();
+ SendSideBandwidthEstimation() = delete;
+ explicit SendSideBandwidthEstimation(RtcEventLog* event_log);
virtual ~SendSideBandwidthEstimation();
void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const;
@@ -53,8 +54,6 @@
void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
int GetMinBitrate() const;
- void SetEventLog(RtcEventLog* event_log);
-
private:
enum UmaState { kNoUpdate, kFirstDone, kDone };
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
index a6fda5b..64cb2fe 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
@@ -12,12 +12,14 @@
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/call/mock/mock_rtc_event_log.h"
#include "webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h"
namespace webrtc {
void TestProbing(bool use_delay_based) {
- SendSideBandwidthEstimation bwe;
+ MockRtcEventLog event_log;
+ SendSideBandwidthEstimation bwe(&event_log);
bwe.SetMinMaxBitrate(100000, 1500000);
bwe.SetSendBitrate(200000);
@@ -62,7 +64,11 @@
}
TEST(SendSideBweTest, DoesntReapplyBitrateDecreaseWithoutFollowingRemb) {
- SendSideBandwidthEstimation bwe;
+ MockRtcEventLog event_log;
+ EXPECT_CALL(event_log,
+ LogBwePacketLossEvent(testing::Gt(0), testing::Gt(0), 0))
+ .Times(3);
+ SendSideBandwidthEstimation bwe(&event_log);
static const int kMinBitrateBps = 100000;
static const int kInitialBitrateBps = 1000000;
bwe.SetMinMaxBitrate(kMinBitrateBps, 1500000);
diff --git a/webrtc/modules/congestion_controller/DEPS b/webrtc/modules/congestion_controller/DEPS
index d72e34d..675f103 100644
--- a/webrtc/modules/congestion_controller/DEPS
+++ b/webrtc/modules/congestion_controller/DEPS
@@ -1,5 +1,6 @@
include_rules = [
"+webrtc/base",
+ "+webrtc/call/mock",
"+webrtc/system_wrappers",
"+webrtc/video",
]
diff --git a/webrtc/modules/congestion_controller/congestion_controller.cc b/webrtc/modules/congestion_controller/congestion_controller.cc
index c836f40..d42ad4a 100644
--- a/webrtc/modules/congestion_controller/congestion_controller.cc
+++ b/webrtc/modules/congestion_controller/congestion_controller.cc
@@ -153,37 +153,17 @@
CongestionController::CongestionController(
Clock* clock,
- BitrateObserver* bitrate_observer,
- RemoteBitrateObserver* remote_bitrate_observer)
- : clock_(clock),
- observer_(nullptr),
- packet_router_(new PacketRouter()),
- pacer_(new PacedSender(clock_, packet_router_.get())),
- remote_bitrate_estimator_(
- new WrappingBitrateEstimator(remote_bitrate_observer, clock_)),
- bitrate_controller_(
- BitrateController::CreateBitrateController(clock_, bitrate_observer)),
- remote_estimator_proxy_(clock_, packet_router_.get()),
- transport_feedback_adapter_(bitrate_controller_.get(), clock_),
- min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps),
- last_reported_bitrate_bps_(0),
- last_reported_fraction_loss_(0),
- last_reported_rtt_(0),
- network_state_(kNetworkUp) {
- Init();
-}
-
-CongestionController::CongestionController(
- Clock* clock,
Observer* observer,
- RemoteBitrateObserver* remote_bitrate_observer)
+ RemoteBitrateObserver* remote_bitrate_observer,
+ RtcEventLog* event_log)
: clock_(clock),
observer_(observer),
packet_router_(new PacketRouter()),
pacer_(new PacedSender(clock_, packet_router_.get())),
remote_bitrate_estimator_(
new WrappingBitrateEstimator(remote_bitrate_observer, clock_)),
- bitrate_controller_(BitrateController::CreateBitrateController(clock_)),
+ bitrate_controller_(
+ BitrateController::CreateBitrateController(clock_, event_log)),
remote_estimator_proxy_(clock_, packet_router_.get()),
transport_feedback_adapter_(bitrate_controller_.get(), clock_),
min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps),
@@ -198,6 +178,7 @@
Clock* clock,
Observer* observer,
RemoteBitrateObserver* remote_bitrate_observer,
+ RtcEventLog* event_log,
std::unique_ptr<PacketRouter> packet_router,
std::unique_ptr<PacedSender> pacer)
: clock_(clock),
@@ -208,7 +189,8 @@
new WrappingBitrateEstimator(remote_bitrate_observer, clock_)),
// Constructed last as this object calls the provided callback on
// construction.
- bitrate_controller_(BitrateController::CreateBitrateController(clock_)),
+ bitrate_controller_(
+ BitrateController::CreateBitrateController(clock_, event_log)),
remote_estimator_proxy_(clock_, packet_router_.get()),
transport_feedback_adapter_(bitrate_controller_.get(), clock_),
min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps),
diff --git a/webrtc/modules/congestion_controller/congestion_controller_unittest.cc b/webrtc/modules/congestion_controller/congestion_controller_unittest.cc
index 1a26582..aa4a5d3 100644
--- a/webrtc/modules/congestion_controller/congestion_controller_unittest.cc
+++ b/webrtc/modules/congestion_controller/congestion_controller_unittest.cc
@@ -10,6 +10,7 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/call/mock/mock_rtc_event_log.h"
#include "webrtc/modules/pacing/mock/mock_paced_sender.h"
#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
@@ -34,9 +35,9 @@
pacer_ = new NiceMock<MockPacedSender>();
std::unique_ptr<PacedSender> pacer(pacer_); // Passes ownership.
std::unique_ptr<PacketRouter> packet_router(new PacketRouter());
- controller_.reset(
- new CongestionController(&clock_, &observer_, &remote_bitrate_observer_,
- std::move(packet_router), std::move(pacer)));
+ controller_.reset(new CongestionController(
+ &clock_, &observer_, &remote_bitrate_observer_, &event_log_,
+ std::move(packet_router), std::move(pacer)));
bandwidth_observer_.reset(
controller_->GetBitrateController()->CreateRtcpBandwidthObserver());
@@ -51,6 +52,7 @@
StrictMock<MockCongestionObserver> observer_;
NiceMock<MockPacedSender>* pacer_;
NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
+ MockRtcEventLog event_log_;
std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
std::unique_ptr<CongestionController> controller_;
const uint32_t kInitialBitrateBps = 60000;
diff --git a/webrtc/modules/congestion_controller/include/congestion_controller.h b/webrtc/modules/congestion_controller/include/congestion_controller.h
index 82906ef..a48f000 100644
--- a/webrtc/modules/congestion_controller/include/congestion_controller.h
+++ b/webrtc/modules/congestion_controller/include/congestion_controller.h
@@ -29,11 +29,11 @@
namespace webrtc {
class BitrateController;
-class BitrateObserver;
class Clock;
class ProcessThread;
class RemoteBitrateEstimator;
class RemoteBitrateObserver;
+class RtcEventLog;
class TransportFeedbackObserver;
class CongestionController : public CallStatsObserver, public Module {
@@ -52,17 +52,14 @@
protected:
virtual ~Observer() {}
};
- // Deprecated
- // TODO(perkj): Remove once no other clients use this ctor.
- CongestionController(Clock* clock,
- BitrateObserver* bitrate_observer,
- RemoteBitrateObserver* remote_bitrate_observer);
- CongestionController(Clock* clock,
- Observer* observer,
- RemoteBitrateObserver* remote_bitrate_observer);
CongestionController(Clock* clock,
Observer* observer,
RemoteBitrateObserver* remote_bitrate_observer,
+ RtcEventLog* event_log);
+ CongestionController(Clock* clock,
+ Observer* observer,
+ RemoteBitrateObserver* remote_bitrate_observer,
+ RtcEventLog* event_log,
std::unique_ptr<PacketRouter> packet_router,
std::unique_ptr<PacedSender> pacer);
virtual ~CongestionController();
diff --git a/webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h b/webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h
index 20955ea..f7d4aaa 100644
--- a/webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h
+++ b/webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h
@@ -31,8 +31,12 @@
public:
MockCongestionController(Clock* clock,
Observer* observer,
- RemoteBitrateObserver* remote_bitrate_observer)
- : CongestionController(clock, observer, remote_bitrate_observer) {}
+ RemoteBitrateObserver* remote_bitrate_observer,
+ RtcEventLog* event_log)
+ : CongestionController(clock,
+ observer,
+ remote_bitrate_observer,
+ event_log) {}
MOCK_METHOD3(SetBweBitrates,
void(int min_bitrate_bps,
int start_bitrate_bps,
diff --git a/webrtc/modules/remote_bitrate_estimator/DEPS b/webrtc/modules/remote_bitrate_estimator/DEPS
index 9a863d9..fc31fa7 100644
--- a/webrtc/modules/remote_bitrate_estimator/DEPS
+++ b/webrtc/modules/remote_bitrate_estimator/DEPS
@@ -7,4 +7,10 @@
"nada\.h": [
"+webrtc/voice_engine",
],
+ "remb\.h": [
+ "+webrtc/call/mock",
+ ],
+ "send_side\.h": [
+ "+webrtc/call/mock",
+ ],
}
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
index e2d3da9..1c878b9 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
@@ -25,7 +25,9 @@
RembBweSender::RembBweSender(int kbps, BitrateObserver* observer, Clock* clock)
: bitrate_controller_(
- BitrateController::CreateBitrateController(clock, observer)),
+ BitrateController::CreateBitrateController(clock,
+ observer,
+ &event_log_)),
feedback_observer_(bitrate_controller_->CreateRtcpBandwidthObserver()),
clock_(clock) {
assert(kbps >= kMinBitrateKbps);
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h b/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h
index 3dc4f38..5840aef 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h
@@ -16,6 +16,7 @@
#include <vector>
#include "webrtc/base/constructormagic.h"
+#include "webrtc/call/mock/mock_rtc_event_log.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
namespace webrtc {
@@ -45,6 +46,7 @@
private:
Clock* clock_;
+ MockRtcEventLog event_log_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RembBweSender);
};
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
index 7a31b54..6d79e09 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
@@ -22,7 +22,9 @@
FullBweSender::FullBweSender(int kbps, BitrateObserver* observer, Clock* clock)
: bitrate_controller_(
- BitrateController::CreateBitrateController(clock, observer)),
+ BitrateController::CreateBitrateController(clock,
+ observer,
+ &event_log_)),
rbe_(new RemoteBitrateEstimatorAbsSendTime(this)),
feedback_observer_(bitrate_controller_->CreateRtcpBandwidthObserver()),
clock_(clock),
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h b/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
index ca7f6db..9eb0c1a 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
@@ -14,6 +14,7 @@
#include <memory>
#include <vector>
+#include "webrtc/call/mock/mock_rtc_event_log.h"
#include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
@@ -46,6 +47,7 @@
bool has_received_ack_;
uint16_t last_acked_seq_num_;
int64_t last_log_time_ms_;
+ MockRtcEventLog event_log_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FullBweSender);
};
diff --git a/webrtc/pc/channelmanager.cc b/webrtc/pc/channelmanager.cc
index fba1040..c2ce1cc 100644
--- a/webrtc/pc/channelmanager.cc
+++ b/webrtc/pc/channelmanager.cc
@@ -389,17 +389,4 @@
Bind(&MediaEngineInterface::StopAecDump, media_engine_.get()));
}
-bool ChannelManager::StartRtcEventLog(rtc::PlatformFile file,
- int64_t max_size_bytes) {
- return worker_thread_->Invoke<bool>(
- RTC_FROM_HERE, Bind(&MediaEngineInterface::StartRtcEventLog,
- media_engine_.get(), file, max_size_bytes));
-}
-
-void ChannelManager::StopRtcEventLog() {
- worker_thread_->Invoke<void>(
- RTC_FROM_HERE,
- Bind(&MediaEngineInterface::StopRtcEventLog, media_engine_.get()));
-}
-
} // namespace cricket
diff --git a/webrtc/pc/channelmanager.h b/webrtc/pc/channelmanager.h
index 34188e6..c6a67df 100644
--- a/webrtc/pc/channelmanager.h
+++ b/webrtc/pc/channelmanager.h
@@ -138,12 +138,6 @@
// Stops recording AEC dump.
void StopAecDump();
- // Starts RtcEventLog using existing file.
- bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes);
-
- // Stops logging RtcEventLog.
- void StopRtcEventLog();
-
private:
typedef std::vector<VoiceChannel*> VoiceChannels;
typedef std::vector<VideoChannel*> VideoChannels;
diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h
index dc2a961..fef8b7d 100644
--- a/webrtc/test/mock_voe_channel_proxy.h
+++ b/webrtc/test/mock_voe_channel_proxy.h
@@ -57,6 +57,7 @@
MOCK_CONST_METHOD0(GetAudioDecoderFactory,
const rtc::scoped_refptr<AudioDecoderFactory>&());
MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
+ MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log));
};
} // namespace test
} // namespace webrtc
diff --git a/webrtc/test/mock_voice_engine.h b/webrtc/test/mock_voice_engine.h
index be72e0d..469028d 100644
--- a/webrtc/test/mock_voice_engine.h
+++ b/webrtc/test/mock_voice_engine.h
@@ -157,7 +157,6 @@
int(int channel, bool& enabled, VadModes& mode, bool& disabledDTX));
MOCK_METHOD2(SetOpusMaxPlaybackRate, int(int channel, int frequency_hz));
MOCK_METHOD2(SetOpusDtx, int(int channel, bool enable_dtx));
- MOCK_METHOD0(GetEventLog, RtcEventLog*());
// VoEExternalMedia
MOCK_METHOD3(RegisterExternalMediaProcessing,
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 943a045..8287668 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -19,6 +19,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/base/timeutils.h"
+#include "webrtc/call/rtc_event_log.h"
#include "webrtc/common.h"
#include "webrtc/config.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
@@ -58,6 +59,87 @@
const int kTelephoneEventAttenuationdB = 10;
+class RtcEventLogProxy final : public webrtc::RtcEventLog {
+ public:
+ RtcEventLogProxy() : event_log_(nullptr) {}
+
+ bool StartLogging(const std::string& file_name,
+ int64_t max_size_bytes) override {
+ RTC_NOTREACHED();
+ return false;
+ }
+
+ bool StartLogging(rtc::PlatformFile log_file,
+ int64_t max_size_bytes) override {
+ RTC_NOTREACHED();
+ return false;
+ }
+
+ void StopLogging() override { RTC_NOTREACHED(); }
+
+ void LogVideoReceiveStreamConfig(
+ const webrtc::VideoReceiveStream::Config& config) override {
+ rtc::CritScope lock(&crit_);
+ if (event_log_) {
+ event_log_->LogVideoReceiveStreamConfig(config);
+ }
+ }
+
+ void LogVideoSendStreamConfig(
+ const webrtc::VideoSendStream::Config& config) override {
+ rtc::CritScope lock(&crit_);
+ if (event_log_) {
+ event_log_->LogVideoSendStreamConfig(config);
+ }
+ }
+
+ void LogRtpHeader(webrtc::PacketDirection direction,
+ webrtc::MediaType media_type,
+ const uint8_t* header,
+ size_t packet_length) override {
+ rtc::CritScope lock(&crit_);
+ if (event_log_) {
+ event_log_->LogRtpHeader(direction, media_type, header, packet_length);
+ }
+ }
+
+ void LogRtcpPacket(webrtc::PacketDirection direction,
+ webrtc::MediaType media_type,
+ const uint8_t* packet,
+ size_t length) override {
+ rtc::CritScope lock(&crit_);
+ if (event_log_) {
+ event_log_->LogRtcpPacket(direction, media_type, packet, length);
+ }
+ }
+
+ void LogAudioPlayout(uint32_t ssrc) override {
+ rtc::CritScope lock(&crit_);
+ if (event_log_) {
+ event_log_->LogAudioPlayout(ssrc);
+ }
+ }
+
+ void LogBwePacketLossEvent(int32_t bitrate,
+ uint8_t fraction_loss,
+ int32_t total_packets) override {
+ rtc::CritScope lock(&crit_);
+ if (event_log_) {
+ event_log_->LogBwePacketLossEvent(bitrate, fraction_loss, total_packets);
+ }
+ }
+
+ void SetEventLog(RtcEventLog* event_log) {
+ rtc::CritScope lock(&crit_);
+ event_log_ = event_log;
+ }
+
+ private:
+ rtc::CriticalSection crit_;
+ RtcEventLog* event_log_ GUARDED_BY(crit_);
+ RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogProxy);
+};
+
class TransportFeedbackProxy : public TransportFeedbackObserver {
public:
TransportFeedbackProxy() : feedback_observer_(nullptr) {
@@ -480,11 +562,9 @@
MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
int32_t id,
AudioFrame* audioFrame) {
- if (event_log_) {
- unsigned int ssrc;
- RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
- event_log_->LogAudioPlayout(ssrc);
- }
+ unsigned int ssrc;
+ RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
+ event_log_proxy_->LogAudioPlayout(ssrc);
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
bool muted;
if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame,
@@ -671,9 +751,8 @@
int32_t Channel::CreateChannel(Channel*& channel,
int32_t channelId,
uint32_t instanceId,
- RtcEventLog* const event_log,
const Config& config) {
- return CreateChannel(channel, channelId, instanceId, event_log, config,
+ return CreateChannel(channel, channelId, instanceId, config,
CreateBuiltinAudioDecoderFactory());
}
@@ -681,15 +760,13 @@
Channel*& channel,
int32_t channelId,
uint32_t instanceId,
- RtcEventLog* const event_log,
const Config& config,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
"Channel::CreateChannel(channelId=%d, instanceId=%d)", channelId,
instanceId);
- channel =
- new Channel(channelId, instanceId, event_log, config, decoder_factory);
+ channel = new Channel(channelId, instanceId, config, decoder_factory);
if (channel == NULL) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
"Channel::CreateChannel() unable to allocate memory for"
@@ -748,12 +825,11 @@
Channel::Channel(int32_t channelId,
uint32_t instanceId,
- RtcEventLog* const event_log,
const Config& config,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)
: _instanceId(instanceId),
_channelId(channelId),
- event_log_(event_log),
+ event_log_proxy_(new RtcEventLogProxy()),
rtp_header_parser_(RtpHeaderParser::Create()),
rtp_payload_registry_(
new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
@@ -856,7 +932,7 @@
seq_num_allocator_proxy_.get();
configuration.transport_feedback_callback = feedback_observer_proxy_.get();
}
- configuration.event_log = event_log;
+ configuration.event_log = &(*event_log_proxy_);
_rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
_rtpRtcpModule->SetSendingMediaStatus(false);
@@ -3008,6 +3084,10 @@
}
}
+void Channel::SetRtcEventLog(RtcEventLog* event_log) {
+ event_log_proxy_->SetEventLog(event_log);
+}
+
int Channel::RegisterExternalMediaProcessing(ProcessingTypes type,
VoEMediaProcess& processObject) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 9b6a2a1..68211a6 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -66,6 +66,7 @@
namespace voe {
class OutputMixer;
+class RtcEventLogProxy;
class RtpPacketSenderProxy;
class Statistics;
class StatisticsProxy;
@@ -173,18 +174,15 @@
static int32_t CreateChannel(Channel*& channel,
int32_t channelId,
uint32_t instanceId,
- RtcEventLog* const event_log,
const Config& config);
static int32_t CreateChannel(
Channel*& channel,
int32_t channelId,
uint32_t instanceId,
- RtcEventLog* const event_log,
const Config& config,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
Channel(int32_t channelId,
uint32_t instanceId,
- RtcEventLog* const event_log,
const Config& config,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
int32_t Init();
@@ -451,6 +449,9 @@
// Disassociate a send channel if it was associated.
void DisassociateSendChannel(int channel_id);
+ // Set a RtcEventLog logging object.
+ void SetRtcEventLog(RtcEventLog* event_log);
+
protected:
void OnIncomingFractionLoss(int fraction_lost);
@@ -486,7 +487,7 @@
ChannelState channel_state_;
- RtcEventLog* const event_log_;
+ std::unique_ptr<voe::RtcEventLogProxy> event_log_proxy_;
std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
diff --git a/webrtc/voice_engine/channel_manager.cc b/webrtc/voice_engine/channel_manager.cc
index 47350b5..5f20b2e 100644
--- a/webrtc/voice_engine/channel_manager.cc
+++ b/webrtc/voice_engine/channel_manager.cc
@@ -47,10 +47,7 @@
: channel(channel), ref_count(1) {}
ChannelManager::ChannelManager(uint32_t instance_id, const Config& config)
- : instance_id_(instance_id),
- last_channel_id_(-1),
- config_(config),
- event_log_(RtcEventLog::Create(Clock::GetRealTimeClock())) {}
+ : instance_id_(instance_id), last_channel_id_(-1), config_(config) {}
ChannelOwner ChannelManager::CreateChannel() {
return CreateChannel(CreateBuiltinAudioDecoderFactory());
@@ -75,8 +72,8 @@
const Config& config,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
Channel* channel;
- Channel::CreateChannel(channel, ++last_channel_id_, instance_id_,
- event_log_.get(), config, decoder_factory);
+ Channel::CreateChannel(channel, ++last_channel_id_, instance_id_, config,
+ decoder_factory);
ChannelOwner channel_owner(channel);
rtc::CritScope crit(&lock_);
@@ -143,10 +140,6 @@
return channels_.size();
}
-RtcEventLog* ChannelManager::GetEventLog() const {
- return event_log_.get();
-}
-
ChannelManager::Iterator::Iterator(ChannelManager* channel_manager)
: iterator_pos_(0) {
channel_manager->GetAllChannels(&channels_);
diff --git a/webrtc/voice_engine/channel_manager.h b/webrtc/voice_engine/channel_manager.h
index 213a48f..c59ee5d 100644
--- a/webrtc/voice_engine/channel_manager.h
+++ b/webrtc/voice_engine/channel_manager.h
@@ -17,7 +17,6 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/call/rtc_event_log.h"
#include "webrtc/system_wrappers/include/atomic32.h"
#include "webrtc/typedefs.h"
@@ -119,9 +118,6 @@
size_t NumOfChannels() const;
- // Returns a pointer to the event log object stored within the ChannelManager.
- RtcEventLog* GetEventLog() const;
-
private:
// Create a channel given a configuration, |config|.
ChannelOwner CreateChannelInternal(
@@ -136,7 +132,6 @@
std::vector<ChannelOwner> channels_;
const Config& config_;
- std::unique_ptr<RtcEventLog> event_log_;
RTC_DISALLOW_COPY_AND_ASSIGN(ChannelManager);
};
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index f60728a..215d931 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -204,6 +204,11 @@
RTC_DCHECK_EQ(0, error);
}
+void ChannelProxy::SetRtcEventLog(RtcEventLog* event_log) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ channel()->SetRtcEventLog(event_log);
+}
+
Channel* ChannelProxy::channel() const {
RTC_DCHECK(channel_owner_.channel());
return channel_owner_.channel();
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index 39eed67..0c26b00 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -24,6 +24,7 @@
class AudioSinkInterface;
class PacketRouter;
+class RtcEventLog;
class RtpPacketSender;
class Transport;
class TransportFeedbackObserver;
@@ -87,6 +88,8 @@
virtual void SetChannelOutputVolumeScaling(float scaling);
+ virtual void SetRtcEventLog(RtcEventLog* event_log);
+
private:
Channel* channel() const;
diff --git a/webrtc/voice_engine/include/voe_codec.h b/webrtc/voice_engine/include/voe_codec.h
index 6c4fb38..5d94ac2 100644
--- a/webrtc/voice_engine/include/voe_codec.h
+++ b/webrtc/voice_engine/include/voe_codec.h
@@ -35,7 +35,6 @@
namespace webrtc {
-class RtcEventLog;
class VoiceEngine;
class WEBRTC_DLLEXPORT VoECodec {
@@ -132,10 +131,6 @@
// success, and -1 if failed.
virtual int SetOpusDtx(int channel, bool enable_dtx) = 0;
- // Get a pointer to the event logging object associated with this Voice
- // Engine. This pointer will remain valid until VoiceEngine is destroyed.
- virtual RtcEventLog* GetEventLog() = 0;
-
protected:
VoECodec() {}
virtual ~VoECodec() {}
diff --git a/webrtc/voice_engine/test/auto_test/standard/codec_test.cc b/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
index 3a3d830..5a500af 100644
--- a/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
@@ -176,30 +176,6 @@
}
}
-#ifdef ENABLE_RTC_EVENT_LOG
-TEST_F(CodecTest, RtcEventLogIntegrationTest) {
- webrtc::RtcEventLog* event_log = voe_codec_->GetEventLog();
- ASSERT_TRUE(event_log);
-
- // Find the name of the current test, in order to use it as a temporary
- // filename.
- auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
- const std::string temp_filename = webrtc::test::OutputPath() +
- test_info->test_case_name() +
- test_info->name();
- // Create a log file.
- event_log->StartLogging(temp_filename, 1000);
- event_log->StopLogging();
-
- // Check if the file has been created.
- FILE* event_file = fopen(temp_filename.c_str(), "r");
- ASSERT_TRUE(event_file);
- fclose(event_file);
- // Remove the temporary file.
- remove(temp_filename.c_str());
-}
-#endif // ENABLE_RTC_EVENT_LOG
-
// TODO(xians, phoglund): Re-enable when issue 372 is resolved.
TEST_F(CodecTest, DISABLED_ManualVerifySendCodecsForAllPacketSizes) {
for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
diff --git a/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc b/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
index d3d2a2d..674a05c 100644
--- a/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
+++ b/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
@@ -21,7 +21,6 @@
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/format_macros.h"
-#include "webrtc/call/rtc_event_log.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/test/channel_transport/channel_transport.h"
@@ -448,7 +447,6 @@
printf("%i. Set bit rate (only take effect on codecs that allow the "
"change) \n", option_index++);
printf("%i. Toggle AECdump recording \n", option_index++);
- printf("%i. Record RtcEventLog file of 30 seconds \n", option_index++);
printf("Select action or %i to stop the call: ", option_index);
int option_selection;
@@ -795,9 +793,6 @@
printf("Debug recording named %s started\n", kDebugFileName);
}
debug_recording_started = !debug_recording_started;
- } else if (option_selection == option_index++) {
- const char* kDebugFileName = "eventlog.rel";
- codec->GetEventLog()->StartLogging(kDebugFileName, 30000);
} else {
break;
}
diff --git a/webrtc/voice_engine/voe_codec_impl.cc b/webrtc/voice_engine/voe_codec_impl.cc
index 5a16592..19891bc 100644
--- a/webrtc/voice_engine/voe_codec_impl.cc
+++ b/webrtc/voice_engine/voe_codec_impl.cc
@@ -376,10 +376,6 @@
return channelPtr->SetOpusDtx(enable_dtx);
}
-RtcEventLog* VoECodecImpl::GetEventLog() {
- return _shared->channel_manager().GetEventLog();
-}
-
#endif // WEBRTC_VOICE_ENGINE_CODEC_API
} // namespace webrtc
diff --git a/webrtc/voice_engine/voe_codec_impl.h b/webrtc/voice_engine/voe_codec_impl.h
index 5095f6e..5519232 100644
--- a/webrtc/voice_engine/voe_codec_impl.h
+++ b/webrtc/voice_engine/voe_codec_impl.h
@@ -58,8 +58,6 @@
int SetOpusDtx(int channel, bool enable_dtx) override;
- RtcEventLog* GetEventLog() override;
-
protected:
VoECodecImpl(voe::SharedData* shared);
~VoECodecImpl() override;
