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

#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