modules/rtp_rtcp/source/rtp_rtcp_impl.h

/*
 *  Copyright (c) 2012 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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_

#include <memory>
#include <set>
#include <utility>
#include <vector>

#include "webrtc/api/optional.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/rtc_base/gtest_prod_util.h"

namespace webrtc {

class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
 public:
  explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);

  // Returns the number of milliseconds until the module want a worker thread to
  // call Process.
  int64_t TimeUntilNextProcess() override;

  // Process any pending tasks such as timeouts.
  void Process() override;

  // Receiver part.

  // Called when we receive an RTCP packet.
  void IncomingRtcpPacket(const uint8_t* incoming_packet,
                          size_t incoming_packet_length) override;

  void SetRemoteSSRC(uint32_t ssrc) override;

  // Sender part.

  int32_t RegisterSendPayload(const CodecInst& voice_codec) override;

  int32_t RegisterSendPayload(const VideoCodec& video_codec) override;

  void RegisterVideoSendPayload(int payload_type,
                                const char* payload_name) override;

  int32_t DeRegisterSendPayload(int8_t payload_type) override;

  // Register RTP header extension.
  int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
                                         uint8_t id) override;

  int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) override;

  bool HasBweExtensions() const override;

  // Get start timestamp.
  uint32_t StartTimestamp() const override;

  // Configure start timestamp, default is a random number.
  void SetStartTimestamp(uint32_t timestamp) override;

  uint16_t SequenceNumber() const override;

  // Set SequenceNumber, default is a random number.
  void SetSequenceNumber(uint16_t seq) override;

  void SetRtpState(const RtpState& rtp_state) override;
  void SetRtxState(const RtpState& rtp_state) override;
  RtpState GetRtpState() const override;
  RtpState GetRtxState() const override;

  uint32_t SSRC() const override;

  // Configure SSRC, default is a random number.
  void SetSSRC(uint32_t ssrc) override;

  void SetCsrcs(const std::vector<uint32_t>& csrcs) override;

  RTCPSender::FeedbackState GetFeedbackState();

  void SetRtxSendStatus(int mode) override;
  int RtxSendStatus() const override;

  void SetRtxSsrc(uint32_t ssrc) override;

  void SetRtxSendPayloadType(int payload_type,
                             int associated_payload_type) override;

  rtc::Optional<uint32_t> FlexfecSsrc() const override;

  // Sends kRtcpByeCode when going from true to false.
  int32_t SetSendingStatus(bool sending) override;

  bool Sending() const override;

  // Drops or relays media packets.
  void SetSendingMediaStatus(bool sending) override;

  bool SendingMedia() const override;

  // Used by the codec module to deliver a video or audio frame for
  // packetization.
  bool SendOutgoingData(FrameType frame_type,
                        int8_t payload_type,
                        uint32_t time_stamp,
                        int64_t capture_time_ms,
                        const uint8_t* payload_data,
                        size_t payload_size,
                        const RTPFragmentationHeader* fragmentation,
                        const RTPVideoHeader* rtp_video_header,
                        uint32_t* transport_frame_id_out) override;

  bool TimeToSendPacket(uint32_t ssrc,
                        uint16_t sequence_number,
                        int64_t capture_time_ms,
                        bool retransmission,
                        const PacedPacketInfo& pacing_info) override;

  // Returns the number of padding bytes actually sent, which can be more or
  // less than |bytes|.
  size_t TimeToSendPadding(size_t bytes,
                           const PacedPacketInfo& pacing_info) override;

  // RTCP part.

  // Get RTCP status.
  RtcpMode RTCP() const override;

  // Configure RTCP status i.e on/off.
  void SetRTCPStatus(RtcpMode method) override;

  // Set RTCP CName.
  int32_t SetCNAME(const char* c_name) override;

  // Get remote CName.
  int32_t RemoteCNAME(uint32_t remote_ssrc,
                      char c_name[RTCP_CNAME_SIZE]) const override;

  // Get remote NTP.
  int32_t RemoteNTP(uint32_t* received_ntp_secs,
                    uint32_t* received_ntp_frac,
                    uint32_t* rtcp_arrival_time_secs,
                    uint32_t* rtcp_arrival_time_frac,
                    uint32_t* rtcp_timestamp) const override;

  int32_t AddMixedCNAME(uint32_t ssrc, const char* c_name) override;

  int32_t RemoveMixedCNAME(uint32_t ssrc) override;

  // Get RoundTripTime.
  int32_t RTT(uint32_t remote_ssrc,
              int64_t* rtt,
              int64_t* avg_rtt,
              int64_t* min_rtt,
              int64_t* max_rtt) const override;

  // Force a send of an RTCP packet.
  // Normal SR and RR are triggered via the process function.
  int32_t SendRTCP(RTCPPacketType rtcpPacketType) override;

  int32_t SendCompoundRTCP(
      const std::set<RTCPPacketType>& rtcpPacketTypes) override;

  // Statistics of the amount of data sent and received.
  int32_t DataCountersRTP(size_t* bytes_sent,
                          uint32_t* packets_sent) const override;

  void GetSendStreamDataCounters(
      StreamDataCounters* rtp_counters,
      StreamDataCounters* rtx_counters) const override;

  void GetRtpPacketLossStats(
      bool outgoing,
      uint32_t ssrc,
      struct RtpPacketLossStats* loss_stats) const override;

  // Get received RTCP report, report block.
  int32_t RemoteRTCPStat(
      std::vector<RTCPReportBlock>* receive_blocks) const override;

  // (REMB) Receiver Estimated Max Bitrate.
  bool REMB() const override;

  void SetREMBStatus(bool enable) override;

  void SetREMBData(uint32_t bitrate,
                   const std::vector<uint32_t>& ssrcs) override;

  // (TMMBR) Temporary Max Media Bit Rate.
  bool TMMBR() const override;

  void SetTMMBRStatus(bool enable) override;

  void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) override;

  size_t MaxRtpPacketSize() const override;

  void SetMaxRtpPacketSize(size_t max_packet_size) override;

  // (NACK) Negative acknowledgment part.

  int SelectiveRetransmissions() const override;

  int SetSelectiveRetransmissions(uint8_t settings) override;

  // Send a Negative acknowledgment packet.
  // TODO(philipel): Deprecate SendNACK and use SendNack instead.
  int32_t SendNACK(const uint16_t* nack_list, uint16_t size) override;

  void SendNack(const std::vector<uint16_t>& sequence_numbers) override;

  // Store the sent packets, needed to answer to a negative acknowledgment
  // requests.
  void SetStorePacketsStatus(bool enable, uint16_t number_to_store) override;

  bool StorePackets() const override;

  // Called on receipt of RTCP report block from remote side.
  void RegisterRtcpStatisticsCallback(
      RtcpStatisticsCallback* callback) override;
  RtcpStatisticsCallback* GetRtcpStatisticsCallback() override;

  bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) override;
  // (APP) Application specific data.
  int32_t SetRTCPApplicationSpecificData(uint8_t sub_type,
                                         uint32_t name,
                                         const uint8_t* data,
                                         uint16_t length) override;

  // (XR) VOIP metric.
  int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) override;

  // (XR) Receiver reference time report.
  void SetRtcpXrRrtrStatus(bool enable) override;

  bool RtcpXrRrtrStatus() const override;

  // Audio part.

  // Send a TelephoneEvent tone using RFC 2833 (4733).
  int32_t SendTelephoneEventOutband(uint8_t key,
                                    uint16_t time_ms,
                                    uint8_t level) override;

  // Store the audio level in d_bov for header-extension-for-audio-level-
  // indication.
  int32_t SetAudioLevel(uint8_t level_d_bov) override;

  // Video part.

  // Set method for requesting a new key frame.
  int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) override;

  // Send a request for a keyframe.
  int32_t RequestKeyFrame() override;

  void SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type) override;

  bool SetFecParameters(const FecProtectionParams& delta_params,
                        const FecProtectionParams& key_params) override;

  bool LastReceivedNTP(uint32_t* NTPsecs,
                       uint32_t* NTPfrac,
                       uint32_t* remote_sr) const;

  std::vector<rtcp::TmmbItem> BoundingSet(bool* tmmbr_owner);

  void BitrateSent(uint32_t* total_rate,
                   uint32_t* video_rate,
                   uint32_t* fec_rate,
                   uint32_t* nackRate) const override;

  void RegisterSendChannelRtpStatisticsCallback(
      StreamDataCountersCallback* callback) override;
  StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback()
      const override;

  void OnReceivedNack(
      const std::vector<uint16_t>& nack_sequence_numbers) override;
  void OnReceivedRtcpReportBlocks(
      const ReportBlockList& report_blocks) override;
  void OnRequestSendReport() override;

  void SetVideoBitrateAllocation(const BitrateAllocation& bitrate) override;

 protected:
  bool UpdateRTCPReceiveInformationTimers();

  RTPSender* rtp_sender() { return rtp_sender_.get(); }
  const RTPSender* rtp_sender() const { return rtp_sender_.get(); }

  RTCPSender* rtcp_sender() { return &rtcp_sender_; }
  const RTCPSender* rtcp_sender() const { return &rtcp_sender_; }

  RTCPReceiver* rtcp_receiver() { return &rtcp_receiver_; }
  const RTCPReceiver* rtcp_receiver() const { return &rtcp_receiver_; }

  const Clock* clock() const { return clock_; }

 private:
  FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
  FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
  int64_t RtcpReportInterval();
  void SetRtcpReceiverSsrcs(uint32_t main_ssrc);

  void set_rtt_ms(int64_t rtt_ms);
  int64_t rtt_ms() const;

  bool TimeToSendFullNackList(int64_t now) const;

  std::unique_ptr<RTPSender> rtp_sender_;
  RTCPSender rtcp_sender_;
  RTCPReceiver rtcp_receiver_;

  const Clock* const clock_;

  const bool audio_;

  const RtpKeepAliveConfig keepalive_config_;
  int64_t last_bitrate_process_time_;
  int64_t last_rtt_process_time_;
  int64_t next_process_time_;
  int64_t next_keepalive_time_;
  uint16_t packet_overhead_;

  // Send side
  int64_t nack_last_time_sent_full_;
  uint32_t nack_last_time_sent_full_prev_;
  uint16_t nack_last_seq_number_sent_;

  KeyFrameRequestMethod key_frame_req_method_;

  RemoteBitrateEstimator* remote_bitrate_;

  RtcpRttStats* rtt_stats_;

  PacketLossStats send_loss_stats_;
  PacketLossStats receive_loss_stats_;

  // The processed RTT from RtcpRttStats.
  rtc::CriticalSection critical_section_rtt_;
  int64_t rtt_ms_;
};

}  // namespace webrtc

#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_