Reland "Remove WEBRTC_TRACE."

This is a reland of 2209b90
Original change's description:
> Remove WEBRTC_TRACE.
> 
> Bug: webrtc:5118
> Change-Id: I0af0f8845ee016fa61d7cecc526e2a672ec8732d
> Reviewed-on: https://webrtc-review.googlesource.com/5382
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20114}

Bug: webrtc:5118
Change-Id: I2d93fd40fcaa251c363bdcfb1c04b834a3a7f0e9
Reviewed-on: https://webrtc-review.googlesource.com/6000
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20132}

Author: Fredrik Solenberg

call/call.cc

@@ -57,7 +57,6 @@
 #include "system_wrappers/include/cpu_info.h"
 #include "system_wrappers/include/metrics.h"
 #include "system_wrappers/include/rw_lock_wrapper.h"
-#include "system_wrappers/include/trace.h"
 #include "video/call_stats.h"
 #include "video/send_delay_stats.h"
 #include "video/stats_counter.h"
@@ -440,7 +439,6 @@ Call::Call(const Call::Config& config,
     RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
                   config.bitrate_config.start_bitrate_bps);
   }
-  Trace::CreateTrace();
   transport_send->send_side_cc()->RegisterNetworkObserver(this);
   transport_send_ = std::move(transport_send);
   transport_send_->send_side_cc()->SignalNetworkState(kNetworkDown);
@@ -500,8 +498,6 @@ Call::~Call() {
   }
   UpdateReceiveHistograms();
   UpdateHistograms();
-
-  Trace::ReturnTrace();
 }
 
 rtc::Optional<RtpPacketReceived> Call::ParseRtpPacket(

common_types.h

@@ -78,64 +78,6 @@ class OutStream : public RewindableStream {
   virtual bool Write(const void* buf, size_t len) = 0;
 };
 
-enum TraceModule {
-  kTraceUndefined = 0,
-  // not a module, triggered from the engine code
-  kTraceVoice = 0x0001,
-  // not a module, triggered from the engine code
-  kTraceVideo = 0x0002,
-  // not a module, triggered from the utility code
-  kTraceUtility = 0x0003,
-  kTraceRtpRtcp = 0x0004,
-  kTraceTransport = 0x0005,
-  kTraceSrtp = 0x0006,
-  kTraceAudioCoding = 0x0007,
-  kTraceAudioMixerServer = 0x0008,
-  kTraceAudioMixerClient = 0x0009,
-  kTraceFile = 0x000a,
-  kTraceAudioProcessing = 0x000b,
-  kTraceVideoCoding = 0x0010,
-  kTraceVideoMixer = 0x0011,
-  kTraceAudioDevice = 0x0012,
-  kTraceVideoRenderer = 0x0014,
-  kTraceVideoCapture = 0x0015,
-  kTraceRemoteBitrateEstimator = 0x0017,
-};
-
-enum TraceLevel {
-  kTraceNone = 0x0000,  // no trace
-  kTraceStateInfo = 0x0001,
-  kTraceWarning = 0x0002,
-  kTraceError = 0x0004,
-  kTraceCritical = 0x0008,
-  kTraceApiCall = 0x0010,
-  kTraceDefault = 0x00ff,
-
-  kTraceModuleCall = 0x0020,
-  kTraceMemory = 0x0100,  // memory info
-  kTraceTimer = 0x0200,   // timing info
-  kTraceStream = 0x0400,  // "continuous" stream of data
-
-  // used for debug purposes
-  kTraceDebug = 0x0800,  // debug
-  kTraceInfo = 0x1000,   // debug info
-
-  // Non-verbose level used by LS_INFO of logging.h. Do not use directly.
-  kTraceTerseInfo = 0x2000,
-
-  kTraceAll = 0xffff
-};
-
-// External Trace API
-class TraceCallback {
- public:
-  virtual void Print(TraceLevel level, const char* message, int length) = 0;
-
- protected:
-  virtual ~TraceCallback() {}
-  TraceCallback() {}
-};
-
 enum FileFormats {
   kFileFormatWavFile = 1,
   kFileFormatCompressedFile = 2,

examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java

@@ -624,9 +624,8 @@ private void createPeerConnectionInternal() {
     }
     isInitiator = false;
 
-    // Set default WebRTC tracing and INFO libjingle logging.
+    // Set INFO libjingle logging.
     // NOTE: this _must_ happen while |factory| is alive!
-    Logging.enableTracing("logcat:", EnumSet.of(Logging.TraceLevel.TRACE_DEFAULT));
     Logging.enableLogToDebugOutput(Logging.Severity.LS_INFO);
 
     mediaStream = factory.createLocalMediaStream("ARDAMS");

media/engine/webrtcvoiceengine.cc

@@ -42,20 +42,13 @@
 #include "rtc_base/trace_event.h"
 #include "system_wrappers/include/field_trial.h"
 #include "system_wrappers/include/metrics.h"
-#include "system_wrappers/include/trace.h"
 #include "voice_engine/transmit_mixer.h"
 
 namespace cricket {
 namespace {
 
 constexpr size_t kMaxUnsignaledRecvStreams = 4;
 
-const int kDefaultTraceFilter = webrtc::kTraceNone | webrtc::kTraceTerseInfo |
-                                webrtc::kTraceWarning | webrtc::kTraceError |
-                                webrtc::kTraceCritical;
-const int kElevatedTraceFilter = kDefaultTraceFilter | webrtc::kTraceStateInfo |
-                                 webrtc::kTraceInfo;
-
 constexpr int kNackRtpHistoryMs = 5000;
 
 // Check to verify that the define for the intelligibility enhancer is properly
@@ -254,7 +247,6 @@ WebRtcVoiceEngine::~WebRtcVoiceEngine() {
   if (initialized_) {
     StopAecDump();
     voe_wrapper_->base()->Terminate();
-    webrtc::Trace::SetTraceCallback(nullptr);
   }
 }
 
@@ -287,12 +279,8 @@ void WebRtcVoiceEngine::Init() {
 
   channel_config_.enable_voice_pacing = true;
 
-  // Temporarily turn logging level up for the Init() call.
-  webrtc::Trace::SetTraceCallback(this);
-  webrtc::Trace::set_level_filter(kElevatedTraceFilter);
   RTC_CHECK_EQ(0,
                voe_wrapper_->base()->Init(adm_.get(), apm(), decoder_factory_));
-  webrtc::Trace::set_level_filter(kDefaultTraceFilter);
 
   // No ADM supplied? Get the default one from VoE.
   if (!adm_) {
@@ -676,30 +664,6 @@ RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
   return capabilities;
 }
 
-void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
-                              int length) {
-  // Note: This callback can happen on any thread!
-  rtc::LoggingSeverity sev = rtc::LS_VERBOSE;
-  if (level == webrtc::kTraceError || level == webrtc::kTraceCritical)
-    sev = rtc::LS_ERROR;
-  else if (level == webrtc::kTraceWarning)
-    sev = rtc::LS_WARNING;
-  else if (level == webrtc::kTraceStateInfo || level == webrtc::kTraceInfo)
-    sev = rtc::LS_INFO;
-  else if (level == webrtc::kTraceTerseInfo)
-    sev = rtc::LS_INFO;
-
-  // Skip past boilerplate prefix text.
-  if (length < 72) {
-    std::string msg(trace, length);
-    LOG(LS_ERROR) << "Malformed webrtc log message: ";
-    LOG_V(sev) << msg;
-  } else {
-    std::string msg(trace + 71, length - 72);
-    LOG_V(sev) << "webrtc: " << msg;
-  }
-}
-
 void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel* channel) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(channel);

media/engine/webrtcvoiceengine.h

@@ -48,7 +48,7 @@ class WebRtcVoiceMediaChannel;
 
 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
 // It uses the WebRtc VoiceEngine library for audio handling.
-class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
+class WebRtcVoiceEngine final {
   friend class WebRtcVoiceMediaChannel;
  public:
   WebRtcVoiceEngine(
@@ -65,7 +65,7 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
       rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
       rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing,
       VoEWrapper* voe_wrapper);
-  ~WebRtcVoiceEngine() override;
+  ~WebRtcVoiceEngine();
 
   // Does initialization that needs to occur on the worker thread.
   void Init();
@@ -108,9 +108,6 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   // easily at any time.
   bool ApplyOptions(const AudioOptions& options);
 
-  // webrtc::TraceCallback:
-  void Print(webrtc::TraceLevel level, const char* trace, int length) override;
-
   void StartAecDump(const std::string& filename);
   int CreateVoEChannel();
 

modules/audio_coding/test/APITest.cc

@@ -25,7 +25,6 @@
 #include "rtc_base/platform_thread.h"
 #include "rtc_base/timeutils.h"
 #include "system_wrappers/include/event_wrapper.h"
-#include "system_wrappers/include/trace.h"
 #include "test/gtest.h"
 #include "test/testsupport/fileutils.h"
 #include "typedefs.h"  // NOLINT(build/include)
@@ -259,15 +258,8 @@ int16_t APITest::SetUp() {
   // B
   _outFreqHzB = _outFileB.SamplingFrequency();
 
-  //Trace::SetEncryptedTraceFile("ACMAPITestEncrypted.txt");
-
   char print[11];
 
-  // Create a trace file.
-  Trace::CreateTrace();
-  Trace::SetTraceFile(
-      (webrtc::test::OutputPath() + "acm_api_trace.txt").c_str());
-
   printf("\nRandom Test (y/n)?");
   EXPECT_TRUE(fgets(print, 10, stdin) != NULL);
   print[10] = '\0';

modules/audio_coding/test/EncodeDecodeTest.cc

@@ -19,7 +19,6 @@
 #include "modules/audio_coding/codecs/audio_format_conversion.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_coding/test/utility.h"
-#include "system_wrappers/include/trace.h"
 #include "test/gtest.h"
 #include "test/testsupport/fileutils.h"
 
@@ -176,9 +175,6 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
 void Receiver::Teardown() {
   delete[] _playoutBuffer;
   _pcmFile.Close();
-  if (testMode > 1) {
-    Trace::ReturnTrace();
-  }
 }
 
 bool Receiver::IncomingPacket() {
@@ -254,21 +250,13 @@ void Receiver::Run() {
 
 EncodeDecodeTest::EncodeDecodeTest() {
   _testMode = 2;
-  Trace::CreateTrace();
-  Trace::SetTraceFile(
-      (webrtc::test::OutputPath() + "acm_encdec_trace.txt").c_str());
 }
 
 EncodeDecodeTest::EncodeDecodeTest(int testMode) {
   //testMode == 0 for autotest
   //testMode == 1 for testing all codecs/parameters
   //testMode > 1 for specific user-input test (as it was used before)
   _testMode = testMode;
-  if (_testMode != 0) {
-    Trace::CreateTrace();
-    Trace::SetTraceFile(
-        (webrtc::test::OutputPath() + "acm_encdec_trace.txt").c_str());
-  }
 }
 
 void EncodeDecodeTest::Perform() {
@@ -326,11 +314,6 @@ void EncodeDecodeTest::Perform() {
       rtpFile.Close();
     }
   }
-
-  // End tracing.
-  if (_testMode == 1) {
-    Trace::ReturnTrace();
-  }
 }
 
 std::string EncodeDecodeTest::EncodeToFile(int fileType,