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

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..4454c25947d5d1d1d226aea04d75833d362ddf39
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/acm2/acm_dump.cc
@@ -0,0 +1,220 @@
+/*
+ *  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