/*
 *  Copyright (c) 2016 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.
 */

#ifndef LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_
#define LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_

#include <memory>

#include "logging/rtc_event_log/events/rtc_event.h"
#include "logging/rtc_event_log/events/rtc_event_alr_state.h"
#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
#include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h"
#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
#include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
#include "logging/rtc_event_log/events/rtc_event_probe_result_success.h"
#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h"
#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
#include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
#include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "rtc_base/random.h"

namespace webrtc {

namespace test {

class EventGenerator {
 public:
  explicit EventGenerator(uint64_t seed) : prng_(seed) {}

  std::unique_ptr<RtcEventAlrState> NewAlrState();

  std::unique_ptr<RtcEventAudioPlayout> NewAudioPlayout(uint32_t ssrc);

  std::unique_ptr<RtcEventAudioNetworkAdaptation> NewAudioNetworkAdaptation();

  std::unique_ptr<RtcEventBweUpdateDelayBased> NewBweUpdateDelayBased();

  std::unique_ptr<RtcEventBweUpdateLossBased> NewBweUpdateLossBased();

  std::unique_ptr<RtcEventProbeClusterCreated> NewProbeClusterCreated();

  std::unique_ptr<RtcEventProbeResultFailure> NewProbeResultFailure();

  std::unique_ptr<RtcEventProbeResultSuccess> NewProbeResultSuccess();

  std::unique_ptr<RtcEventIceCandidatePairConfig> NewIceCandidatePairConfig();

  std::unique_ptr<RtcEventIceCandidatePair> NewIceCandidatePair();

  std::unique_ptr<RtcEventRtcpPacketIncoming> NewRtcpPacketIncoming();

  std::unique_ptr<RtcEventRtcpPacketOutgoing> NewRtcpPacketOutgoing();

  void RandomizeRtpPacket(size_t payload_size,
                          size_t padding_size,
                          uint32_t ssrc,
                          const RtpHeaderExtensionMap& extension_map,
                          RtpPacket* rtp_packet);

  std::unique_ptr<RtcEventRtpPacketIncoming> NewRtpPacketIncoming(
      uint32_t ssrc,
      const RtpHeaderExtensionMap& extension_map);

  std::unique_ptr<RtcEventRtpPacketOutgoing> NewRtpPacketOutgoing(
      uint32_t ssrc,
      const RtpHeaderExtensionMap& extension_map);

  RtpHeaderExtensionMap NewRtpHeaderExtensionMap();

  std::unique_ptr<RtcEventAudioReceiveStreamConfig> NewAudioReceiveStreamConfig(
      uint32_t ssrc,
      const RtpHeaderExtensionMap& extensions);

  std::unique_ptr<RtcEventAudioSendStreamConfig> NewAudioSendStreamConfig(
      uint32_t ssrc,
      const RtpHeaderExtensionMap& extensions);

  std::unique_ptr<RtcEventVideoReceiveStreamConfig> NewVideoReceiveStreamConfig(
      uint32_t ssrc,
      const RtpHeaderExtensionMap& extensions);

  std::unique_ptr<RtcEventVideoSendStreamConfig> NewVideoSendStreamConfig(
      uint32_t ssrc,
      const RtpHeaderExtensionMap& extensions);

 private:
  rtcp::ReportBlock NewReportBlock();
  rtcp::SenderReport NewSenderReport();
  rtcp::ReceiverReport NewReceiverReport();

  Random prng_;
};

bool VerifyLoggedAlrStateEvent(const RtcEventAlrState& original_event,
                               const LoggedAlrStateEvent& logged_event);

bool VerifyLoggedAudioPlayoutEvent(const RtcEventAudioPlayout& original_event,
                                   const LoggedAudioPlayoutEvent& logged_event);

bool VerifyLoggedAudioNetworkAdaptationEvent(
    const RtcEventAudioNetworkAdaptation& original_event,
    const LoggedAudioNetworkAdaptationEvent& logged_event);

bool VerifyLoggedBweDelayBasedUpdate(
    const RtcEventBweUpdateDelayBased& original_event,
    const LoggedBweDelayBasedUpdate& logged_event);

bool VerifyLoggedBweLossBasedUpdate(
    const RtcEventBweUpdateLossBased& original_event,
    const LoggedBweLossBasedUpdate& logged_event);

bool VerifyLoggedBweProbeClusterCreatedEvent(
    const RtcEventProbeClusterCreated& original_event,
    const LoggedBweProbeClusterCreatedEvent& logged_event);

bool VerifyLoggedBweProbeFailureEvent(
    const RtcEventProbeResultFailure& original_event,
    const LoggedBweProbeFailureEvent& logged_event);

bool VerifyLoggedBweProbeSuccessEvent(
    const RtcEventProbeResultSuccess& original_event,
    const LoggedBweProbeSuccessEvent& logged_event);

bool VerifyLoggedIceCandidatePairConfig(
    const RtcEventIceCandidatePairConfig& original_event,
    const LoggedIceCandidatePairConfig& logged_event);

bool VerifyLoggedIceCandidatePairEvent(
    const RtcEventIceCandidatePair& original_event,
    const LoggedIceCandidatePairEvent& logged_event);

bool VerifyLoggedRtpPacketIncoming(
    const RtcEventRtpPacketIncoming& original_event,
    const LoggedRtpPacketIncoming& logged_event);

bool VerifyLoggedRtpPacketOutgoing(
    const RtcEventRtpPacketOutgoing& original_event,
    const LoggedRtpPacketOutgoing& logged_event);

bool VerifyLoggedRtcpPacketIncoming(
    const RtcEventRtcpPacketIncoming& original_event,
    const LoggedRtcpPacketIncoming& logged_event);

bool VerifyLoggedRtcpPacketOutgoing(
    const RtcEventRtcpPacketOutgoing& original_event,
    const LoggedRtcpPacketOutgoing& logged_event);

bool VerifyLoggedStartEvent(int64_t start_time_us,
                            const LoggedStartEvent& logged_event);
bool VerifyLoggedStopEvent(int64_t stop_time_us,
                           const LoggedStopEvent& logged_event);

bool VerifyLoggedAudioRecvConfig(
    const RtcEventAudioReceiveStreamConfig& original_event,
    const LoggedAudioRecvConfig& logged_event);

bool VerifyLoggedAudioSendConfig(
    const RtcEventAudioSendStreamConfig& original_event,
    const LoggedAudioSendConfig& logged_event);

bool VerifyLoggedVideoRecvConfig(
    const RtcEventVideoReceiveStreamConfig& original_event,
    const LoggedVideoRecvConfig& logged_event);

bool VerifyLoggedVideoSendConfig(
    const RtcEventVideoSendStreamConfig& original_event,
    const LoggedVideoSendConfig& logged_event);

}  // namespace test
}  // namespace webrtc

#endif  // LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_