Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
The breaking tests in Chromium have been temporarily disabled, they will be fixed and reenabled soon.
Original CLs: https://codereview.webrtc.org/1748403002/, https://codereview.webrtc.org/2107253002/ and https://codereview.webrtc.org/2106103003/.
TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org,tkchin@webrtc.org
BUG=webrtc:4741, webrtc:5603, chromium:609749
Review-Url: https://codereview.webrtc.org/2110113003
Cr-Commit-Position: refs/heads/master@{#13379}
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;
}
