Delete all log messages depending on system_wrappers.

webrtc/logging/rtc_event_log/rtc_event_log.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 16 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
 
 #ifdef ENABLE_RTC_EVENT_LOG
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #else
 #include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #endif
 #endif
 
@@ skipping 152 matching lines @@
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   RtcEventLogHelperThread::ControlMessage message;
   message.message_type = RtcEventLogHelperThread::ControlMessage::START_FILE;
   message.max_size_bytes = max_size_bytes <= 0
                                ? std::numeric_limits<int64_t>::max()
                                : max_size_bytes;
   message.start_time = rtc::TimeMicros();
   message.stop_time = std::numeric_limits<int64_t>::max();
   message.file.reset(FileWrapper::Create());
   if (!message.file->OpenFile(file_name.c_str(), false)) {
-    LOG(LS_ERROR) << "Can't open file. WebRTC event log not started.";
     return false;
   }
   if (!message_queue_.Insert(&message)) {
-    LOG(LS_ERROR) << "Message queue full. Can't start logging.";
     return false;
   }
   helper_thread_.SignalNewEvent();
-  LOG(LS_INFO) << "Starting WebRTC event log.";
   return true;
 }
 
 bool RtcEventLogImpl::StartLogging(rtc::PlatformFile platform_file,
                                    int64_t max_size_bytes) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   RtcEventLogHelperThread::ControlMessage message;
   message.message_type = RtcEventLogHelperThread::ControlMessage::START_FILE;
   message.max_size_bytes = max_size_bytes <= 0
                                ? std::numeric_limits<int64_t>::max()
                                : max_size_bytes;
   message.start_time = rtc::TimeMicros();
   message.stop_time = std::numeric_limits<int64_t>::max();
   message.file.reset(FileWrapper::Create());
   FILE* file_handle = rtc::FdopenPlatformFileForWriting(platform_file);
   if (!file_handle) {
-    LOG(LS_ERROR) << "Can't open file. WebRTC event log not started.";
     // Even though we failed to open a FILE*, the platform_file is still open
     // and needs to be closed.
     if (!rtc::ClosePlatformFile(platform_file)) {
-      LOG(LS_ERROR) << "Can't close file.";
     }
     return false;
   }
   if (!message.file->OpenFromFileHandle(file_handle)) {
-    LOG(LS_ERROR) << "Can't open file. WebRTC event log not started.";
     return false;
   }
   if (!message_queue_.Insert(&message)) {
-    LOG(LS_ERROR) << "Message queue full. Can't start logging.";
     return false;
   }
   helper_thread_.SignalNewEvent();
-  LOG(LS_INFO) << "Starting WebRTC event log.";
   return true;
 }
 
 void RtcEventLogImpl::StopLogging() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   RtcEventLogHelperThread::ControlMessage message;
   message.message_type = RtcEventLogHelperThread::ControlMessage::STOP_FILE;
   message.stop_time = rtc::TimeMicros();
   while (!message_queue_.Insert(&message)) {
     // TODO(terelius): We would like to have a blocking Insert function in the
     // SwapQueue, but for the time being we will just clear any previous
     // messages.
     // Since StopLogging waits for the thread, it is essential that we don't
     // clear any STOP_FILE messages. To ensure that there is only one call at a
     // time, we require that all calls to StopLogging are made on the same
     // thread.
-    LOG(LS_ERROR) << "Message queue full. Clearing queue to stop logging.";
     message_queue_.Clear();
   }
-  LOG(LS_INFO) << "Stopping WebRTC event log.";
   helper_thread_.WaitForFileFinished();
 }
 
 void RtcEventLogImpl::LogVideoReceiveStreamConfig(
     const VideoReceiveStream::Config& config) {
   std::unique_ptr<rtclog::Event> event(new rtclog::Event());
   event->set_timestamp_us(rtc::TimeMicros());
   event->set_type(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
 
   rtclog::VideoReceiveConfig* receiver_config =
@@ skipping 280 matching lines @@
   auto probe_result = event->mutable_probe_result();
   probe_result->set_id(id);
   probe_result->set_result(result);
   if (result == rtclog::BweProbeResult::SUCCESS)
     probe_result->set_bitrate_bps(bitrate_bps);
   StoreEvent(&event);
 }
 
 void RtcEventLogImpl::StoreEvent(std::unique_ptr<rtclog::Event>* event) {
   if (!event_queue_.Insert(event)) {
-    LOG(LS_ERROR) << "WebRTC event log queue full. Dropping event.";
   }
   helper_thread_.SignalNewEvent();
 }
 
 bool RtcEventLog::ParseRtcEventLog(const std::string& file_name,
                                    rtclog::EventStream* result) {
   char tmp_buffer[1024];
   int bytes_read = 0;
   std::unique_ptr<FileWrapper> dump_file(FileWrapper::Create());
   if (!dump_file->OpenFile(file_name.c_str(), true)) {
     return false;
   }
   std::string dump_buffer;
   while ((bytes_read = dump_file->Read(tmp_buffer, sizeof(tmp_buffer))) > 0) {
     dump_buffer.append(tmp_buffer, bytes_read);
   }
   dump_file->CloseFile();
   return result->ParseFromString(dump_buffer);
 }
 
 #endif  // ENABLE_RTC_EVENT_LOG
 
 bool RtcEventLogNullImpl::StartLogging(rtc::PlatformFile platform_file,
                                        int64_t max_size_bytes) {
   // The platform_file is open and needs to be closed.
   if (!rtc::ClosePlatformFile(platform_file)) {
-    LOG(LS_ERROR) << "Can't close file.";
   }
   return false;
 }
 
 // RtcEventLog member functions.
 std::unique_ptr<RtcEventLog> RtcEventLog::Create() {
 #ifdef ENABLE_RTC_EVENT_LOG
   return std::unique_ptr<RtcEventLog>(new RtcEventLogImpl());
 #else
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 #endif  // ENABLE_RTC_EVENT_LOG
 }
 
 std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() {
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 }
 
 }  // namespace webrtc