Added functions on libjingle API to start and stop the recording of an RtcEventLog.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1374253002
Cr-Commit-Position: refs/heads/master@{#10297}
diff --git a/talk/app/webrtc/peerconnectionfactory.cc b/talk/app/webrtc/peerconnectionfactory.cc
index 6619f31..54a32bc 100644
--- a/talk/app/webrtc/peerconnectionfactory.cc
+++ b/talk/app/webrtc/peerconnectionfactory.cc
@@ -218,6 +218,16 @@
return channel_manager_->StartAecDump(file);
}
+bool PeerConnectionFactory::StartRtcEventLog(rtc::PlatformFile file) {
+ RTC_DCHECK(signaling_thread_->IsCurrent());
+ return channel_manager_->StartRtcEventLog(file);
+}
+
+void PeerConnectionFactory::StopRtcEventLog() {
+ RTC_DCHECK(signaling_thread_->IsCurrent());
+ channel_manager_->StopRtcEventLog();
+}
+
rtc::scoped_refptr<PeerConnectionInterface>
PeerConnectionFactory::CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
diff --git a/talk/app/webrtc/peerconnectionfactory.h b/talk/app/webrtc/peerconnectionfactory.h
index 8689199..f8e0a40 100644
--- a/talk/app/webrtc/peerconnectionfactory.h
+++ b/talk/app/webrtc/peerconnectionfactory.h
@@ -80,6 +80,8 @@
AudioSourceInterface* audio_source) override;
bool StartAecDump(rtc::PlatformFile file) override;
+ bool StartRtcEventLog(rtc::PlatformFile file) override;
+ void StopRtcEventLog() override;
virtual webrtc::MediaControllerInterface* CreateMediaController() const;
virtual rtc::Thread* signaling_thread();
diff --git a/talk/app/webrtc/peerconnectionfactoryproxy.h b/talk/app/webrtc/peerconnectionfactoryproxy.h
index 4a8cd87..826d577 100644
--- a/talk/app/webrtc/peerconnectionfactoryproxy.h
+++ b/talk/app/webrtc/peerconnectionfactoryproxy.h
@@ -62,6 +62,8 @@
PROXY_METHOD2(rtc::scoped_refptr<AudioTrackInterface>,
CreateAudioTrack, const std::string&, AudioSourceInterface*)
PROXY_METHOD1(bool, StartAecDump, rtc::PlatformFile)
+ PROXY_METHOD1(bool, StartRtcEventLog, rtc::PlatformFile)
+ PROXY_METHOD0(void, StopRtcEventLog)
private:
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection_ot(
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index d3d0d9e..f61f2f6 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -619,6 +619,22 @@
// http://crbug.com/264611.
virtual bool StartAecDump(rtc::PlatformFile file) = 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.
+ // 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.
+ virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
+
+ // Stops logging the RtcEventLog.
+ virtual void StopRtcEventLog() = 0;
+
protected:
// Dtor and ctor protected as objects shouldn't be created or deleted via
// this interface.
diff --git a/talk/media/base/fakemediaengine.h b/talk/media/base/fakemediaengine.h
index 57d0145..5a4f46a 100644
--- a/talk/media/base/fakemediaengine.h
+++ b/talk/media/base/fakemediaengine.h
@@ -785,6 +785,10 @@
bool StartAecDump(rtc::PlatformFile file) { return false; }
+ bool StartRtcEventLog(rtc::PlatformFile file) { return false; }
+
+ void StopRtcEventLog() {}
+
private:
std::vector<FakeVoiceMediaChannel*> channels_;
std::vector<AudioCodec> codecs_;
diff --git a/talk/media/base/mediaengine.h b/talk/media/base/mediaengine.h
index 8468e07..c5b9096 100644
--- a/talk/media/base/mediaengine.h
+++ b/talk/media/base/mediaengine.h
@@ -121,6 +121,12 @@
// Starts AEC dump using existing file.
virtual bool StartAecDump(rtc::PlatformFile file) = 0;
+
+ // Starts RtcEventLog using existing file.
+ virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
+
+ // Stops recording an RtcEventLog.
+ virtual void StopRtcEventLog() = 0;
};
@@ -219,6 +225,12 @@
return voice_.StartAecDump(file);
}
+ virtual bool StartRtcEventLog(rtc::PlatformFile file) {
+ return voice_.StartRtcEventLog(file);
+ }
+
+ virtual void StopRtcEventLog() { voice_.StopRtcEventLog(); }
+
protected:
VOICE voice_;
VIDEO video_;
@@ -251,6 +263,8 @@
}
void SetLogging(int min_sev, const char* filter) {}
bool StartAecDump(rtc::PlatformFile file) { return false; }
+ bool StartRtcEventLog(rtc::PlatformFile file) { return false; }
+ void StopRtcEventLog() {}
private:
std::vector<AudioCodec> codecs_;
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index caaf87e..2a3df2d 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -50,6 +50,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
+#include "webrtc/call/rtc_event_log.h"
#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/system_wrappers/interface/field_trial.h"
@@ -1306,6 +1307,14 @@
}
}
+bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) {
+ return voe_wrapper_->codec()->GetEventLog()->StartLogging(file);
+}
+
+void WebRtcVoiceEngine::StopRtcEventLog() {
+ voe_wrapper_->codec()->GetEventLog()->StopLogging();
+}
+
int WebRtcVoiceEngine::CreateVoiceChannel(VoEWrapper* voice_engine_wrapper) {
return voice_engine_wrapper->base()->CreateChannel(voe_config_);
}
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index cba9456..cd792bc 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -108,6 +108,13 @@
// Starts AEC dump using existing file.
bool StartAecDump(rtc::PlatformFile file);
+ // Starts recording an RtcEventLog using an existing file until 10 minutes
+ // pass or the StopRtcEventLog function is called.
+ bool StartRtcEventLog(rtc::PlatformFile file);
+
+ // Stops recording the RtcEventLog.
+ void StopRtcEventLog();
+
// Create a VoiceEngine Channel.
int CreateMediaVoiceChannel();
diff --git a/talk/session/media/channelmanager.cc b/talk/session/media/channelmanager.cc
index 5930dfd..5caf3c4 100644
--- a/talk/session/media/channelmanager.cc
+++ b/talk/session/media/channelmanager.cc
@@ -630,4 +630,14 @@
Bind(&MediaEngineInterface::StartAecDump, media_engine_.get(), file));
}
+bool ChannelManager::StartRtcEventLog(rtc::PlatformFile file) {
+ return worker_thread_->Invoke<bool>(
+ Bind(&MediaEngineInterface::StartRtcEventLog, media_engine_.get(), file));
+}
+
+void ChannelManager::StopRtcEventLog() {
+ worker_thread_->Invoke<void>(
+ Bind(&MediaEngineInterface::StopRtcEventLog, media_engine_.get()));
+}
+
} // namespace cricket
diff --git a/talk/session/media/channelmanager.h b/talk/session/media/channelmanager.h
index c17a60f..2417662 100644
--- a/talk/session/media/channelmanager.h
+++ b/talk/session/media/channelmanager.h
@@ -170,6 +170,12 @@
// Starts AEC dump using existing file.
bool StartAecDump(rtc::PlatformFile file);
+ // Starts RtcEventLog using existing file.
+ bool StartRtcEventLog(rtc::PlatformFile file);
+
+ // Stops logging RtcEventLog.
+ void StopRtcEventLog();
+
sigslot::signal2<VideoCapturer*, CaptureState> SignalVideoCaptureStateChange;
protected:
diff --git a/webrtc/call/rtc_event_log.cc b/webrtc/call/rtc_event_log.cc
index 97885cc..6a15912 100644
--- a/webrtc/call/rtc_event_log.cc
+++ b/webrtc/call/rtc_event_log.cc
@@ -37,6 +37,7 @@
class RtcEventLogImpl final : public RtcEventLog {
public:
void StartLogging(const std::string& file_name, int duration_ms) override {}
+ bool StartLogging(rtc::PlatformFile log_file) override { return false; }
void StopLogging(void) override {}
void LogVideoReceiveStreamConfig(
const VideoReceiveStream::Config& config) override {}
@@ -57,9 +58,8 @@
class RtcEventLogImpl final : public RtcEventLog {
public:
- RtcEventLogImpl();
-
void StartLogging(const std::string& file_name, int duration_ms) override;
+ bool StartLogging(rtc::PlatformFile log_file) override;
void StopLogging() override;
void LogVideoReceiveStreamConfig(
const VideoReceiveStream::Config& config) override;
@@ -75,6 +75,9 @@
void LogAudioPlayout(uint32_t ssrc) override;
private:
+ // Starts logging. This function assumes the file_ has been opened succesfully
+ // and that the start_time_us_ and _duration_us_ have been set.
+ void StartLoggingLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Stops logging and clears the stored data and buffers.
void StopLoggingLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Adds a new event to the logfile if logging is active, or adds it to the
@@ -93,13 +96,16 @@
const int recent_log_duration_us = 10000000;
rtc::CriticalSection crit_;
- rtc::scoped_ptr<FileWrapper> file_ GUARDED_BY(crit_);
+ rtc::scoped_ptr<FileWrapper> file_ GUARDED_BY(crit_) =
+ rtc::scoped_ptr<FileWrapper>(FileWrapper::Create());
+ rtc::PlatformFile platform_file_ GUARDED_BY(crit_) =
+ rtc::kInvalidPlatformFileValue;
rtclog::EventStream stream_ GUARDED_BY(crit_);
std::deque<rtclog::Event> recent_log_events_ GUARDED_BY(crit_);
- bool currently_logging_ GUARDED_BY(crit_);
- int64_t start_time_us_ GUARDED_BY(crit_);
- int64_t duration_us_ GUARDED_BY(crit_);
- const Clock* const clock_;
+ bool currently_logging_ GUARDED_BY(crit_) = false;
+ int64_t start_time_us_ GUARDED_BY(crit_) = 0;
+ int64_t duration_us_ GUARDED_BY(crit_) = 0;
+ const Clock* const clock_ = Clock::GetRealTimeClock();
};
namespace {
@@ -143,14 +149,6 @@
} // namespace
// RtcEventLogImpl member functions.
-RtcEventLogImpl::RtcEventLogImpl()
- : file_(FileWrapper::Create()),
- stream_(),
- currently_logging_(false),
- start_time_us_(0),
- duration_us_(0),
- clock_(Clock::GetRealTimeClock()) {
-}
void RtcEventLogImpl::StartLogging(const std::string& file_name,
int duration_ms) {
@@ -161,9 +159,39 @@
if (file_->OpenFile(file_name.c_str(), false) != 0) {
return;
}
- currently_logging_ = true;
start_time_us_ = clock_->TimeInMicroseconds();
duration_us_ = static_cast<int64_t>(duration_ms) * 1000;
+ StartLoggingLocked();
+}
+
+bool RtcEventLogImpl::StartLogging(rtc::PlatformFile log_file) {
+ rtc::CritScope lock(&crit_);
+
+ if (currently_logging_) {
+ StopLoggingLocked();
+ }
+ RTC_DCHECK(platform_file_ == rtc::kInvalidPlatformFileValue);
+
+ FILE* file_stream = rtc::FdopenPlatformFileForWriting(log_file);
+ if (!file_stream) {
+ rtc::ClosePlatformFile(log_file);
+ return false;
+ }
+
+ if (file_->OpenFromFileHandle(file_stream, true, false) != 0) {
+ rtc::ClosePlatformFile(log_file);
+ return false;
+ }
+ platform_file_ = log_file;
+ // Set the start time and duration to keep logging for 10 minutes.
+ start_time_us_ = clock_->TimeInMicroseconds();
+ duration_us_ = 10 * 60 * 1000000;
+ StartLoggingLocked();
+ return true;
+}
+
+void RtcEventLogImpl::StartLoggingLocked() {
+ currently_logging_ = true;
// Write all the recent events to the log file, ignoring any old events.
for (auto& event : recent_log_events_) {
if (event.timestamp_us() >= start_time_us_ - recent_log_duration_us) {
@@ -339,6 +367,10 @@
RTC_DCHECK(file_->Open());
StoreToFile(&event);
file_->CloseFile();
+ if (platform_file_ != rtc::kInvalidPlatformFileValue) {
+ rtc::ClosePlatformFile(platform_file_);
+ platform_file_ = rtc::kInvalidPlatformFileValue;
+ }
}
RTC_DCHECK(!file_->Open());
stream_.Clear();
diff --git a/webrtc/call/rtc_event_log.h b/webrtc/call/rtc_event_log.h
index a4b7cad..02bbdc6 100644
--- a/webrtc/call/rtc_event_log.h
+++ b/webrtc/call/rtc_event_log.h
@@ -13,6 +13,7 @@
#include <string>
+#include "webrtc/base/platform_file.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
@@ -41,6 +42,11 @@
// If the file cannot be opened, the RtcEventLog will not start logging.
virtual void StartLogging(const std::string& file_name, int duration_ms) = 0;
+ // Starts logging until either the 10 minute timer runs out or the StopLogging
+ // function is called. The RtcEventLog takes ownership of the supplied
+ // rtc::PlatformFile.
+ virtual bool StartLogging(rtc::PlatformFile log_file) = 0;
+
virtual void StopLogging() = 0;
// Logs configuration information for webrtc::VideoReceiveStream
