Revert "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific t…

webrtc/modules/audio_coding/main/acm2/acm_dump.cc

Index: webrtc/modules/audio_coding/main/acm2/acm_dump.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dump.cc b/webrtc/modules/audio_coding/main/acm2/acm_dump.cc
new file mode 100644
index 0000000000000000000000000000000000000000..9c624d97b67eb728c6f54729ee8a37e6359f2631
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/acm2/acm_dump.cc
@@ -0,0 +1,240 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/modules/audio_coding/main/acm2/acm_dump.h"
+
+#include <deque>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/file_wrapper.h"
+
+#ifdef RTC_AUDIOCODING_DEBUG_DUMP
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/dump.pb.h"
+#else
+#include "webrtc/audio_coding/dump.pb.h"
+#endif
+#endif
+
+namespace webrtc {
+
+// Noop implementation if flag is not set
+#ifndef RTC_AUDIOCODING_DEBUG_DUMP
+class AcmDumpImpl final : public AcmDump {
+ public:
+  void StartLogging(const std::string& file_name, int duration_ms) override{};
+  void LogRtpPacket(bool incoming,
+                    const uint8_t* packet,
+                    size_t length) override{};
+  void LogDebugEvent(DebugEvent event_type,
+                     const std::string& event_message) override{};
+  void LogDebugEvent(DebugEvent event_type) override{};
+};
+#else
+
+class AcmDumpImpl final : public AcmDump {
+ public:
+  AcmDumpImpl();
+
+  void StartLogging(const std::string& file_name, int duration_ms) override;
+  void LogRtpPacket(bool incoming,
+                    const uint8_t* packet,
+                    size_t length) override;
+  void LogDebugEvent(DebugEvent event_type,
+                     const std::string& event_message) override;
+  void LogDebugEvent(DebugEvent event_type) override;
+
+ private:
+  // This function is identical to LogDebugEvent, but requires holding the lock.
+  void LogDebugEventLocked(DebugEvent event_type,
+                           const std::string& event_message)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  // Stops logging and clears the stored data and buffers.
+  void Clear() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  // Adds a new event to the logfile if logging is active, or adds it to the
+  // list of recent log events otherwise.
+  void HandleEvent(ACMDumpEvent* event) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  // Writes the event to the file. Note that this will destroy the state of the
+  // input argument.
+  void StoreToFile(ACMDumpEvent* event) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  // Adds the event to the list of recent events, and removes any events that
+  // are too old and no longer fall in the time window.
+  void AddRecentEvent(const ACMDumpEvent& event)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Amount of time in microseconds to record log events, before starting the
+  // actual log.
+  const int recent_log_duration_us = 10000000;
+
+  rtc::scoped_ptr<webrtc::CriticalSectionWrapper> crit_;
+  rtc::scoped_ptr<webrtc::FileWrapper> file_ GUARDED_BY(crit_);
+  rtc::scoped_ptr<ACMDumpEventStream> stream_ GUARDED_BY(crit_);
+  std::deque<ACMDumpEvent> 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 webrtc::Clock* const clock_;
+};
+
+namespace {
+
+// Convert from AcmDump's debug event enum (runtime format) to the corresponding
+// protobuf enum (serialized format).
+ACMDumpDebugEvent_EventType convertDebugEvent(AcmDump::DebugEvent event_type) {
+  switch (event_type) {
+    case AcmDump::DebugEvent::kLogStart:
+      return ACMDumpDebugEvent::LOG_START;
+    case AcmDump::DebugEvent::kLogEnd:
+      return ACMDumpDebugEvent::LOG_END;
+    case AcmDump::DebugEvent::kAudioPlayout:
+      return ACMDumpDebugEvent::AUDIO_PLAYOUT;
+  }
+  return ACMDumpDebugEvent::UNKNOWN_EVENT;
+}
+
+}  // Anonymous namespace.
+
+// AcmDumpImpl member functions.
+AcmDumpImpl::AcmDumpImpl()
+    : crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
+      file_(webrtc::FileWrapper::Create()),
+      stream_(new webrtc::ACMDumpEventStream()),
+      currently_logging_(false),
+      start_time_us_(0),
+      duration_us_(0),
+      clock_(webrtc::Clock::GetRealTimeClock()) {
+}
+
+void AcmDumpImpl::StartLogging(const std::string& file_name, int duration_ms) {
+  CriticalSectionScoped lock(crit_.get());
+  Clear();
+  if (file_->OpenFile(file_name.c_str(), false) != 0) {
+    return;
+  }
+  // Add LOG_START event to the recent event list. This call will also remove
+  // any events that are too old from the recent event list.
+  LogDebugEventLocked(DebugEvent::kLogStart, "");
+  currently_logging_ = true;
+  start_time_us_ = clock_->TimeInMicroseconds();
+  duration_us_ = static_cast<int64_t>(duration_ms) * 1000;
+  // Write all the recent events to the log file.
+  for (auto&& event : recent_log_events_) {
+    StoreToFile(&event);
+  }
+  recent_log_events_.clear();
+}
+
+void AcmDumpImpl::LogRtpPacket(bool incoming,
+                               const uint8_t* packet,
+                               size_t length) {
+  CriticalSectionScoped lock(crit_.get());
+  ACMDumpEvent rtp_event;
+  const int64_t timestamp = clock_->TimeInMicroseconds();
+  rtp_event.set_timestamp_us(timestamp);
+  rtp_event.set_type(webrtc::ACMDumpEvent::RTP_EVENT);
+  rtp_event.mutable_packet()->set_direction(
+      incoming ? ACMDumpRTPPacket::INCOMING : ACMDumpRTPPacket::OUTGOING);
+  rtp_event.mutable_packet()->set_rtp_data(packet, length);
+  HandleEvent(&rtp_event);
+}
+
+void AcmDumpImpl::LogDebugEvent(DebugEvent event_type,
+                                const std::string& event_message) {
+  CriticalSectionScoped lock(crit_.get());
+  LogDebugEventLocked(event_type, event_message);
+}
+
+void AcmDumpImpl::LogDebugEvent(DebugEvent event_type) {
+  CriticalSectionScoped lock(crit_.get());
+  LogDebugEventLocked(event_type, "");
+}
+
+void AcmDumpImpl::LogDebugEventLocked(DebugEvent event_type,
+                                      const std::string& event_message) {
+  ACMDumpEvent event;
+  int64_t timestamp = clock_->TimeInMicroseconds();
+  event.set_timestamp_us(timestamp);
+  event.set_type(webrtc::ACMDumpEvent::DEBUG_EVENT);
+  auto debug_event = event.mutable_debug_event();
+  debug_event->set_type(convertDebugEvent(event_type));
+  debug_event->set_message(event_message);
+  HandleEvent(&event);
+}
+
+void AcmDumpImpl::Clear() {
+  if (file_->Open()) {
+    file_->CloseFile();
+  }
+  currently_logging_ = false;
+  stream_->Clear();
+}
+
+void AcmDumpImpl::HandleEvent(ACMDumpEvent* event) {
+  if (currently_logging_) {
+    if (clock_->TimeInMicroseconds() < start_time_us_ + duration_us_) {
+      StoreToFile(event);
+    } else {
+      LogDebugEventLocked(DebugEvent::kLogEnd, "");
+      Clear();
+      AddRecentEvent(*event);
+    }
+  } else {
+    AddRecentEvent(*event);
+  }
+}
+
+void AcmDumpImpl::StoreToFile(ACMDumpEvent* event) {
+  // Reuse the same object at every log event.
+  if (stream_->stream_size() < 1) {
+    stream_->add_stream();
+  }
+  DCHECK_EQ(stream_->stream_size(), 1);
+  stream_->mutable_stream(0)->Swap(event);
+
+  std::string dump_buffer;
+  stream_->SerializeToString(&dump_buffer);
+  file_->Write(dump_buffer.data(), dump_buffer.size());
+}
+
+void AcmDumpImpl::AddRecentEvent(const ACMDumpEvent& event) {
+  recent_log_events_.push_back(event);
+  while (recent_log_events_.front().timestamp_us() <
+         event.timestamp_us() - recent_log_duration_us) {
+    recent_log_events_.pop_front();
+  }
+}
+
+bool AcmDump::ParseAcmDump(const std::string& file_name,
+                           ACMDumpEventStream* result) {
+  char tmp_buffer[1024];
+  int bytes_read = 0;
+  rtc::scoped_ptr<FileWrapper> dump_file(FileWrapper::Create());
+  if (dump_file->OpenFile(file_name.c_str(), true) != 0) {
+    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  // RTC_AUDIOCODING_DEBUG_DUMP
+
+// AcmDump member functions.
+rtc::scoped_ptr<AcmDump> AcmDump::Create() {
+  return rtc::scoped_ptr<AcmDump>(new AcmDumpImpl());
+}
+}  // namespace webrtc