Revert "Added ACM_dump protobuf, class for reading/writing and unittest."

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
deleted file mode 100644
index 4454c25947d5d1d1d226aea04d75833d362ddf39..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/acm2/acm_dump.cc
+++ /dev/null
@@ -1,220 +0,0 @@
-/*
- *  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 <sstream>
-
-#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"
-
-// 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
-
-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:
-  // Checks if the logging time has expired, and if so stops the logging.
-  void StopIfNecessary() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  // Stops logging and clears the stored data and buffers.
-  void Clear() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  // Returns true if the logging is currently active.
-  bool CurrentlyLogging() const EXCLUSIVE_LOCKS_REQUIRED(crit_) {
-    return active_ &&
-           (clock_->TimeInMicroseconds() <= start_time_us_ + duration_us_);
-  }
-  // 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_);
-
-  rtc::scoped_ptr<webrtc::CriticalSectionWrapper> crit_;
-  rtc::scoped_ptr<webrtc::FileWrapper> file_ GUARDED_BY(crit_);
-  rtc::scoped_ptr<ACMDumpEventStream> stream_ GUARDED_BY(crit_);
-  bool active_ GUARDED_BY(crit_);
-  int64_t start_time_us_ GUARDED_BY(crit_);
-  int64_t duration_us_ GUARDED_BY(crit_);
-  const webrtc::Clock* clock_ GUARDED_BY(crit_);
-};
-
-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()),
-      active_(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 a single object to the stream that is reused at every log event.
-  stream_->add_stream();
-  active_ = true;
-  start_time_us_ = clock_->TimeInMicroseconds();
-  duration_us_ = static_cast<int64_t>(duration_ms) * 1000;
-  // Log the start event.
-  std::stringstream log_msg;
-  log_msg << "Initial timestamp: " << start_time_us_;
-  LogDebugEventLocked(DebugEvent::kLogStart, log_msg.str());
-}
-
-void AcmDumpImpl::LogRtpPacket(bool incoming,
-                               const uint8_t* packet,
-                               size_t length) {
-  CriticalSectionScoped lock(crit_.get());
-  if (!CurrentlyLogging()) {
-    StopIfNecessary();
-    return;
-  }
-  // Reuse the same object at every log event.
-  auto rtp_event = stream_->mutable_stream(0);
-  rtp_event->clear_debug_event();
-  const int64_t timestamp = clock_->TimeInMicroseconds() - start_time_us_;
-  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);
-  std::string dump_buffer;
-  stream_->SerializeToString(&dump_buffer);
-  file_->Write(dump_buffer.data(), dump_buffer.size());
-  file_->Flush();
-}
-
-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::StopIfNecessary() {
-  if (active_) {
-    DCHECK_GT(clock_->TimeInMicroseconds(), start_time_us_ + duration_us_);
-    LogDebugEventLocked(DebugEvent::kLogEnd, "");
-    Clear();
-  }
-}
-
-void AcmDumpImpl::Clear() {
-  if (active_ || file_->Open()) {
-    file_->CloseFile();
-  }
-  active_ = false;
-  stream_->Clear();
-}
-
-void AcmDumpImpl::LogDebugEventLocked(DebugEvent event_type,
-                                      const std::string& event_message) {
-  if (!CurrentlyLogging()) {
-    StopIfNecessary();
-    return;
-  }
-
-  // Reuse the same object at every log event.
-  auto event = stream_->mutable_stream(0);
-  int64_t timestamp = clock_->TimeInMicroseconds() - start_time_us_;
-  event->set_timestamp_us(timestamp);
-  event->set_type(webrtc::ACMDumpEvent::DEBUG_EVENT);
-  event->clear_packet();
-  auto debug_event = event->mutable_debug_event();
-  debug_event->set_type(convertDebugEvent(event_type));
-  debug_event->set_message(event_message);
-  std::string dump_buffer;
-  stream_->SerializeToString(&dump_buffer);
-  file_->Write(dump_buffer.data(), dump_buffer.size());
-}
-
-#endif  // RTC_AUDIOCODING_DEBUG_DUMP
-
-// AcmDump member functions.
-rtc::scoped_ptr<AcmDump> AcmDump::Create() {
-  return rtc::scoped_ptr<AcmDump>(new AcmDumpImpl());
-}
-
-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);
-}
-
-}  // namespace webrtc