text: /*
   *  Copyright (c) 2013 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_VIDEO_SEND_STREAM_H_
  #define CALL_VIDEO_SEND_STREAM_H_
  
  #include <stdint.h>
  
  #include <map>
  #include <string>
  #include <vector>
  
  #include "absl/types/optional.h"
  #include "api/adaptation/resource.h"
  #include "api/call/transport.h"
  #include "api/crypto/crypto_options.h"
  #include "api/frame_transformer_interface.h"
  #include "api/rtp_parameters.h"
  #include "api/scoped_refptr.h"
  #include "api/video/video_content_type.h"
  #include "api/video/video_frame.h"
  #include "api/video/video_sink_interface.h"
  #include "api/video/video_source_interface.h"
  #include "api/video/video_stream_encoder_settings.h"
  #include "api/video_codecs/video_encoder_config.h"
  #include "call/rtp_config.h"
  #include "common_video/frame_counts.h"
  #include "common_video/include/quality_limitation_reason.h"
  #include "modules/rtp_rtcp/include/report_block_data.h"
  #include "modules/rtp_rtcp/include/rtcp_statistics.h"
  #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
  
  namespace webrtc {
  
  class FrameEncryptorInterface;
  
  class VideoSendStream {
   public:
    // Multiple StreamStats objects are present if simulcast is used (multiple
    // kMedia streams) or if RTX or FlexFEC is negotiated. Multiple SVC layers, on
    // the other hand, does not cause additional StreamStats.
    struct StreamStats {
      enum class StreamType {
        // A media stream is an RTP stream for audio or video. Retransmissions and
        // FEC is either sent over the same SSRC or negotiated to be sent over
        // separate SSRCs, in which case separate StreamStats objects exist with
        // references to this media stream's SSRC.
        kMedia,
        // RTX streams are streams dedicated to retransmissions. They have a
        // dependency on a single kMedia stream: `referenced_media_ssrc`.
        kRtx,
        // FlexFEC streams are streams dedicated to FlexFEC. They have a
        // dependency on a single kMedia stream: `referenced_media_ssrc`.
        kFlexfec,
      };
  
      StreamStats();
      ~StreamStats();
  
      std::string ToString() const;
  
      StreamType type = StreamType::kMedia;
      // If `type` is kRtx or kFlexfec this value is present. The referenced SSRC
      // is the kMedia stream that this stream is performing retransmissions or
      // FEC for. If `type` is kMedia, this value is null.
      absl::optional<uint32_t> referenced_media_ssrc;
      FrameCounts frame_counts;
      int width = 0;
      int height = 0;
      // TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
      int total_bitrate_bps = 0;
      int retransmit_bitrate_bps = 0;
      int avg_delay_ms = 0;
      int max_delay_ms = 0;
      uint64_t total_packet_send_delay_ms = 0;
      StreamDataCounters rtp_stats;
      RtcpPacketTypeCounter rtcp_packet_type_counts;
      // A snapshot of the most recent Report Block with additional data of
      // interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
      absl::optional<ReportBlockData> report_block_data;
      double encode_frame_rate = 0.0;
      int frames_encoded = 0;
      absl::optional<uint64_t> qp_sum;
      uint64_t total_encode_time_ms = 0;
      uint64_t total_encoded_bytes_target = 0;
      uint32_t huge_frames_sent = 0;
    };
  
    struct Stats {
      Stats();
      ~Stats();
      std::string ToString(int64_t time_ms) const;
      std::string encoder_implementation_name = "unknown";
      double input_frame_rate = 0;
      int encode_frame_rate = 0;
      int avg_encode_time_ms = 0;
      int encode_usage_percent = 0;
      uint32_t frames_encoded = 0;
      // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodetime
      uint64_t total_encode_time_ms = 0;
      // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodedbytestarget
      uint64_t total_encoded_bytes_target = 0;
      uint32_t frames = 0;
      uint32_t frames_dropped_by_capturer = 0;
      uint32_t frames_dropped_by_encoder_queue = 0;
      uint32_t frames_dropped_by_rate_limiter = 0;
      uint32_t frames_dropped_by_congestion_window = 0;
      uint32_t frames_dropped_by_encoder = 0;
      // Bitrate the encoder is currently configured to use due to bandwidth
      // limitations.
      int target_media_bitrate_bps = 0;
      // Bitrate the encoder is actually producing.
      int media_bitrate_bps = 0;
      bool suspended = false;
      bool bw_limited_resolution = false;
      bool cpu_limited_resolution = false;
      bool bw_limited_framerate = false;
      bool cpu_limited_framerate = false;
      // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationreason
      QualityLimitationReason quality_limitation_reason =
          QualityLimitationReason::kNone;
      // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationdurations
      std::map<QualityLimitationReason, int64_t> quality_limitation_durations_ms;
      // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges
      uint32_t quality_limitation_resolution_changes = 0;
      // Total number of times resolution as been requested to be changed due to
      // CPU/quality adaptation.
      int number_of_cpu_adapt_changes = 0;
      int number_of_quality_adapt_changes = 0;
      bool has_entered_low_resolution = false;
      std::map<uint32_t, StreamStats> substreams;
      webrtc::VideoContentType content_type =
          webrtc::VideoContentType::UNSPECIFIED;
      uint32_t frames_sent = 0;
      uint32_t huge_frames_sent = 0;
    };
  
    struct Config {
     public:
      Config() = delete;
      Config(Config&&);
      explicit Config(Transport* send_transport);
  
      Config& operator=(Config&&);
      Config& operator=(const Config&) = delete;
  
      ~Config();
  
      // Mostly used by tests.  Avoid creating copies if you can.
      Config Copy() const { return Config(*this); }
  
      std::string ToString() const;
  
      RtpConfig rtp;
  
      VideoStreamEncoderSettings encoder_settings;
  
      // Time interval between RTCP report for video
      int rtcp_report_interval_ms = 1000;
  
      // Transport for outgoing packets.
      Transport* send_transport = nullptr;
  
      // Expected delay needed by the renderer, i.e. the frame will be delivered
      // this many milliseconds, if possible, earlier than expected render time.
      // Only valid if `local_renderer` is set.
      int render_delay_ms = 0;
  
      // Target delay in milliseconds. A positive value indicates this stream is
      // used for streaming instead of a real-time call.
      int target_delay_ms = 0;
  
      // True if the stream should be suspended when the available bitrate fall
      // below the minimum configured bitrate. If this variable is false, the
      // stream may send at a rate higher than the estimated available bitrate.
      bool suspend_below_min_bitrate = false;
  
      // Enables periodic bandwidth probing in application-limited region.
      bool periodic_alr_bandwidth_probing = false;
  
      // An optional custom frame encryptor that allows the entire frame to be
      // encrypted in whatever way the caller chooses. This is not required by
      // default.
      rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor;
  
      // Per PeerConnection cryptography options.
      CryptoOptions crypto_options;
  
      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer;
  
     private:
      // Access to the copy constructor is private to force use of the Copy()
      // method for those exceptional cases where we do use it.
      Config(const Config&);
    };
  
    // Updates the sending state for all simulcast layers that the video send
    // stream owns. This can mean updating the activity one or for multiple
    // layers. The ordering of active layers is the order in which the
    // rtp modules are stored in the VideoSendStream.
    // Note: This starts stream activity if it is inactive and one of the layers
    // is active. This stops stream activity if it is active and all layers are
    // inactive.
    virtual void UpdateActiveSimulcastLayers(std::vector<bool> active_layers) = 0;
  
    // Starts stream activity.
    // When a stream is active, it can receive, process and deliver packets.
    virtual void Start() = 0;
    // Stops stream activity.
    // When a stream is stopped, it can't receive, process or deliver packets.
    virtual void Stop() = 0;
  
    // Accessor for determining if the stream is active. This is an inexpensive
    // call that must be made on the same thread as `Start()` and `Stop()` methods
    // are called on and will return `true` iff activity has been started either
    // via `Start()` or `UpdateActiveSimulcastLayers()`. If activity is either
    // stopped or is in the process of being stopped as a result of a call to
    // either `Stop()` or `UpdateActiveSimulcastLayers()` where all layers were
    // deactivated, the return value will be `false`.
    virtual bool started() = 0;
  
    // If the resource is overusing, the VideoSendStream will try to reduce
    // resolution or frame rate until no resource is overusing.
    // TODO(https://crbug.com/webrtc/11565): When the ResourceAdaptationProcessor
    // is moved to Call this method could be deleted altogether in favor of
    // Call-level APIs only.
    virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) = 0;
    virtual std::vector<rtc::scoped_refptr<Resource>>
    GetAdaptationResources() = 0;
  
    virtual void SetSource(
        rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
        const DegradationPreference& degradation_preference) = 0;
  
    // Set which streams to send. Must have at least as many SSRCs as configured
    // in the config. Encoder settings are passed on to the encoder instance along
    // with the VideoStream settings.
    virtual void ReconfigureVideoEncoder(VideoEncoderConfig config) = 0;
  
    virtual Stats GetStats() = 0;
  
   protected:
    virtual ~VideoSendStream() {}
  };
  
  }  // namespace webrtc
  
  #endif  // CALL_VIDEO_SEND_STREAM_H_
  