Quelle  rtp_video_sender.h   Sprache: C

/*
 *  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.
 */

#ifndef CALL_RTP_VIDEO_SENDER_H_
#define CALL_RTP_VIDEO_SENDER_H_

#include <cstddef>
#include <cstdint>
#include <map>
#include <memory>
#include <optional>
#include <vector>

#include "absl/base/nullability.h"
#include "api/array_view.h"
#include "api/call/bitrate_allocation.h"
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
#include "api/environment/environment.h"
#include "api/fec_controller.h"
#include "api/frame_transformer_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
#include "api/units/frequency.h"
#include "api/video/encoded_image.h"
#include "api/video/video_codec_type.h"
#include "api/video/video_layers_allocation.h"
#include "api/video_codecs/video_encoder.h"
#include "call/rtp_config.h"
#include "call/rtp_payload_params.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "call/rtp_video_sender_interface.h"
#include "common_video/frame_counts.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
#include "modules/rtp_rtcp/source/video_fec_generator.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/thread_annotations.h"

namespace webrtc {

class FrameEncryptorInterface;
class RtpTransportControllerSendInterface;

namespace webrtc_internal_rtp_video_sender {
// RTP state for a single simulcast stream. Internal to the implementation of
// RtpVideoSender.
struct RtpStreamSender {
  RtpStreamSender(std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp,
                  std::unique_ptr<RTPSenderVideo> sender_video,
                  std::unique_ptr<VideoFecGenerator> fec_generator);
  ~RtpStreamSender();

  RtpStreamSender(RtpStreamSender&&) = default;
  RtpStreamSender& operator=(RtpStreamSender&&) = default;

  // Note: Needs pointer stability.
  std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp;
  std::unique_ptr<RTPSenderVideo> sender_video;
  std::unique_ptr<VideoFecGenerator> fec_generator;
};

}  // namespace webrtc_internal_rtp_video_sender

// RtpVideoSender routes outgoing data to the correct sending RTP module, based
// on the simulcast layer in RTPVideoHeader.
class RtpVideoSender : public RtpVideoSenderInterface,
                       public VCMProtectionCallback,
                       public StreamFeedbackObserver {
 public:
  // Rtp modules are assumed to be sorted in simulcast index order.
  RtpVideoSender(
      const Environment& env,
      absl::Nonnull<TaskQueueBase*> transport_queue,
      const std::map<uint32_t, RtpState>& suspended_ssrcs,
      const std::map<uint32_t, RtpPayloadState>& states,
      const RtpConfig& rtp_config,
      int rtcp_report_interval_ms,
      Transport* send_transport,
      const RtpSenderObservers& observers,
      RtpTransportControllerSendInterface* transport,
      RateLimiter* retransmission_limiter,  // move inside RtpTransport
      std::unique_ptr<FecController> fec_controller,
      FrameEncryptorInterface* frame_encryptor,
      const CryptoOptions& crypto_options,  // move inside RtpTransport
      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
  ~RtpVideoSender() override;

  RtpVideoSender(const RtpVideoSender&) = delete;
  RtpVideoSender& operator=(const RtpVideoSender&) = delete;

  void SetSending(bool enabled) RTC_LOCKS_EXCLUDED(mutex_) override;
  bool IsActive() RTC_LOCKS_EXCLUDED(mutex_) override;

  void OnNetworkAvailability(bool network_available)
      RTC_LOCKS_EXCLUDED(mutex_) override;
  std::map<uint32_t, RtpState> GetRtpStates() const
      RTC_LOCKS_EXCLUDED(mutex_) override;
  std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const
      RTC_LOCKS_EXCLUDED(mutex_) override;

  void DeliverRtcp(const uint8_t* packet, size_t length)
      RTC_LOCKS_EXCLUDED(mutex_) override;

  // Implements webrtc::VCMProtectionCallback.
  int ProtectionRequest(const FecProtectionParams* delta_params,
                        const FecProtectionParams* key_params,
                        uint32_t* sent_video_rate_bps,
                        uint32_t* sent_nack_rate_bps,
                        uint32_t* sent_fec_rate_bps)
      RTC_LOCKS_EXCLUDED(mutex_) override;

  // 'retransmission_mode' is either a value of enum RetransmissionMode, or
  // computed with bitwise operators on values of enum RetransmissionMode.
  void SetRetransmissionMode(int retransmission_mode)
      RTC_LOCKS_EXCLUDED(mutex_) override;

  // Implements FecControllerOverride.
  void SetFecAllowed(bool fec_allowed) RTC_LOCKS_EXCLUDED(mutex_) override;

  // Implements EncodedImageCallback.
  // Returns 0 if the packet was routed / sent, -1 otherwise.
  EncodedImageCallback::Result OnEncodedImage(
      const EncodedImage& encoded_image,
      const CodecSpecificInfo* codec_specific_info)
      RTC_LOCKS_EXCLUDED(mutex_) override;

  void OnBitrateAllocationUpdated(const VideoBitrateAllocation& bitrate)
      RTC_LOCKS_EXCLUDED(mutex_) override;
  void OnVideoLayersAllocationUpdated(
      const VideoLayersAllocation& layers) override;
  void OnTransportOverheadChanged(size_t transport_overhead_bytes_per_packet)
      RTC_LOCKS_EXCLUDED(mutex_) override;
  void OnBitrateUpdated(BitrateAllocationUpdate update, int framerate)
      RTC_LOCKS_EXCLUDED(mutex_) override;
  uint32_t GetPayloadBitrateBps() const RTC_LOCKS_EXCLUDED(mutex_) override;
  uint32_t GetProtectionBitrateBps() const RTC_LOCKS_EXCLUDED(mutex_) override;
  void SetEncodingData(size_t width, size_t height, size_t num_temporal_layers)
      RTC_LOCKS_EXCLUDED(mutex_) override;

  std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
      uint32_t ssrc,
      rtc::ArrayView<const uint16_t> sequence_numbers) const
      RTC_LOCKS_EXCLUDED(mutex_) override;

  // From StreamFeedbackObserver.
  void OnPacketFeedbackVector(
      std::vector<StreamPacketInfo> packet_feedback_vector)
      RTC_LOCKS_EXCLUDED(mutex_) override;

 private:
  bool IsActiveLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
  void SetActiveModulesLocked(bool sending)
      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
  void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
  void ConfigureProtection();
  void ConfigureSsrcs(const std::map<uint32_t, RtpState>& suspended_ssrcs);
  bool NackEnabled() const;
  DataRate GetPostEncodeOverhead() const;
  DataRate CalculateOverheadRate(DataRate data_rate,
                                 DataSize packet_size,
                                 DataSize overhead_per_packet,
                                 Frequency framerate) const;
  void SetModuleIsActive(bool sending, RtpRtcpInterface& rtp_module)
      RTC_RUN_ON(transport_checker_);

  const Environment env_;
  const bool use_frame_rate_for_overhead_;
  const bool has_packet_feedback_;

  // Semantically equivalent to checking for `transport_->GetWorkerQueue()`
  // but some tests need to be updated to call from the correct context.
  RTC_NO_UNIQUE_ADDRESS SequenceChecker transport_checker_;
  TaskQueueBase& transport_queue_;

  // TODO(bugs.webrtc.org/13517): Remove mutex_ once RtpVideoSender runs on the
  // transport task queue.
  mutable Mutex mutex_;
  bool active_ RTC_GUARDED_BY(mutex_);

  const std::unique_ptr<FecController> fec_controller_;
  bool fec_allowed_ RTC_GUARDED_BY(mutex_);

  // Rtp modules are assumed to be sorted in simulcast index order.
  const std::vector<webrtc_internal_rtp_video_sender::RtpStreamSender>
      rtp_streams_;
  const RtpConfig rtp_config_;
  const std::optional<VideoCodecType> codec_type_;
  RtpTransportControllerSendInterface* const transport_;

  // When using the generic descriptor we want all simulcast streams to share
  // one frame id space (so that the SFU can switch stream without having to
  // rewrite the frame id), therefore `shared_frame_id` has to live in a place
  // where we are aware of all the different streams.
  int64_t shared_frame_id_ = 0;
  const bool independent_frame_ids_;
  std::vector<RtpPayloadParams> params_ RTC_GUARDED_BY(mutex_);

  size_t transport_overhead_bytes_per_packet_ RTC_GUARDED_BY(mutex_);
  uint32_t protection_bitrate_bps_;
  uint32_t encoder_target_rate_bps_;

  std::vector<bool> loss_mask_vector_ RTC_GUARDED_BY(mutex_);

  std::vector<FrameCounts> frame_counts_ RTC_GUARDED_BY(mutex_);
  FrameCountObserver* const frame_count_observer_;

  // Effectively const map from SSRC to RtpRtcp, for all media SSRCs.
  // This map is set at construction time and never changed, but it's
  // non-trivial to make it properly const.
  std::map<uint32_t, RtpRtcpInterface*> ssrc_to_rtp_module_;

  ScopedTaskSafety safety_;
};

}  // namespace webrtc

#endif  // CALL_RTP_VIDEO_SENDER_H_