1 /* 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef CALL_VIDEO_SEND_STREAM_H_ 12 #define CALL_VIDEO_SEND_STREAM_H_ 13 14 #include <stdint.h> 15 16 #include <map> 17 #include <string> 18 #include <vector> 19 20 #include "absl/types/optional.h" 21 #include "api/adaptation/resource.h" 22 #include "api/call/transport.h" 23 #include "api/crypto/crypto_options.h" 24 #include "api/frame_transformer_interface.h" 25 #include "api/rtp_parameters.h" 26 #include "api/scoped_refptr.h" 27 #include "api/video/video_content_type.h" 28 #include "api/video/video_frame.h" 29 #include "api/video/video_sink_interface.h" 30 #include "api/video/video_source_interface.h" 31 #include "api/video/video_stream_encoder_settings.h" 32 #include "api/video_codecs/video_encoder_config.h" 33 #include "call/rtp_config.h" 34 #include "common_video/include/quality_limitation_reason.h" 35 #include "modules/rtp_rtcp/include/report_block_data.h" 36 #include "modules/rtp_rtcp/include/rtcp_statistics.h" 37 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" 38 39 namespace webrtc { 40 41 class FrameEncryptorInterface; 42 43 class VideoSendStream { 44 public: 45 // Multiple StreamStats objects are present if simulcast is used (multiple 46 // kMedia streams) or if RTX or FlexFEC is negotiated. Multiple SVC layers, on 47 // the other hand, does not cause additional StreamStats. 48 struct StreamStats { 49 enum class StreamType { 50 // A media stream is an RTP stream for audio or video. Retransmissions and 51 // FEC is either sent over the same SSRC or negotiated to be sent over 52 // separate SSRCs, in which case separate StreamStats objects exist with 53 // references to this media stream's SSRC. 54 kMedia, 55 // RTX streams are streams dedicated to retransmissions. They have a 56 // dependency on a single kMedia stream: |referenced_media_ssrc|. 57 kRtx, 58 // FlexFEC streams are streams dedicated to FlexFEC. They have a 59 // dependency on a single kMedia stream: |referenced_media_ssrc|. 60 kFlexfec, 61 }; 62 63 StreamStats(); 64 ~StreamStats(); 65 66 std::string ToString() const; 67 68 StreamType type = StreamType::kMedia; 69 // If |type| is kRtx or kFlexfec this value is present. The referenced SSRC 70 // is the kMedia stream that this stream is performing retransmissions or 71 // FEC for. If |type| is kMedia, this value is null. 72 absl::optional<uint32_t> referenced_media_ssrc; 73 FrameCounts frame_counts; 74 int width = 0; 75 int height = 0; 76 // TODO(holmer): Move bitrate_bps out to the webrtc::Call layer. 77 int total_bitrate_bps = 0; 78 int retransmit_bitrate_bps = 0; 79 int avg_delay_ms = 0; 80 int max_delay_ms = 0; 81 uint64_t total_packet_send_delay_ms = 0; 82 StreamDataCounters rtp_stats; 83 RtcpPacketTypeCounter rtcp_packet_type_counts; 84 RtcpStatistics rtcp_stats; 85 // A snapshot of the most recent Report Block with additional data of 86 // interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats. 87 absl::optional<ReportBlockData> report_block_data; 88 double encode_frame_rate = 0.0; 89 int frames_encoded = 0; 90 absl::optional<uint64_t> qp_sum; 91 uint64_t total_encode_time_ms = 0; 92 uint64_t total_encoded_bytes_target = 0; 93 uint32_t huge_frames_sent = 0; 94 }; 95 96 struct Stats { 97 Stats(); 98 ~Stats(); 99 std::string ToString(int64_t time_ms) const; 100 std::string encoder_implementation_name = "unknown"; 101 int input_frame_rate = 0; 102 int encode_frame_rate = 0; 103 int avg_encode_time_ms = 0; 104 int encode_usage_percent = 0; 105 uint32_t frames_encoded = 0; 106 // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodetime 107 uint64_t total_encode_time_ms = 0; 108 // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodedbytestarget 109 uint64_t total_encoded_bytes_target = 0; 110 uint32_t frames_dropped_by_capturer = 0; 111 uint32_t frames_dropped_by_encoder_queue = 0; 112 uint32_t frames_dropped_by_rate_limiter = 0; 113 uint32_t frames_dropped_by_congestion_window = 0; 114 uint32_t frames_dropped_by_encoder = 0; 115 // Bitrate the encoder is currently configured to use due to bandwidth 116 // limitations. 117 int target_media_bitrate_bps = 0; 118 // Bitrate the encoder is actually producing. 119 int media_bitrate_bps = 0; 120 bool suspended = false; 121 bool bw_limited_resolution = false; 122 bool cpu_limited_resolution = false; 123 bool bw_limited_framerate = false; 124 bool cpu_limited_framerate = false; 125 // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationreason 126 QualityLimitationReason quality_limitation_reason = 127 QualityLimitationReason::kNone; 128 // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationdurations 129 std::map<QualityLimitationReason, int64_t> quality_limitation_durations_ms; 130 // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges 131 uint32_t quality_limitation_resolution_changes = 0; 132 // Total number of times resolution as been requested to be changed due to 133 // CPU/quality adaptation. 134 int number_of_cpu_adapt_changes = 0; 135 int number_of_quality_adapt_changes = 0; 136 bool has_entered_low_resolution = false; 137 std::map<uint32_t, StreamStats> substreams; 138 webrtc::VideoContentType content_type = 139 webrtc::VideoContentType::UNSPECIFIED; 140 uint32_t frames_sent = 0; 141 uint32_t huge_frames_sent = 0; 142 }; 143 144 struct Config { 145 public: 146 Config() = delete; 147 Config(Config&&); 148 explicit Config(Transport* send_transport); 149 150 Config& operator=(Config&&); 151 Config& operator=(const Config&) = delete; 152 153 ~Config(); 154 155 // Mostly used by tests. Avoid creating copies if you can. CopyConfig156 Config Copy() const { return Config(*this); } 157 158 std::string ToString() const; 159 160 RtpConfig rtp; 161 162 VideoStreamEncoderSettings encoder_settings; 163 164 // Time interval between RTCP report for video 165 int rtcp_report_interval_ms = 1000; 166 167 // Transport for outgoing packets. 168 Transport* send_transport = nullptr; 169 170 // Expected delay needed by the renderer, i.e. the frame will be delivered 171 // this many milliseconds, if possible, earlier than expected render time. 172 // Only valid if |local_renderer| is set. 173 int render_delay_ms = 0; 174 175 // Target delay in milliseconds. A positive value indicates this stream is 176 // used for streaming instead of a real-time call. 177 int target_delay_ms = 0; 178 179 // True if the stream should be suspended when the available bitrate fall 180 // below the minimum configured bitrate. If this variable is false, the 181 // stream may send at a rate higher than the estimated available bitrate. 182 bool suspend_below_min_bitrate = false; 183 184 // Enables periodic bandwidth probing in application-limited region. 185 bool periodic_alr_bandwidth_probing = false; 186 187 // An optional custom frame encryptor that allows the entire frame to be 188 // encrypted in whatever way the caller chooses. This is not required by 189 // default. 190 rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor; 191 192 // Per PeerConnection cryptography options. 193 CryptoOptions crypto_options; 194 195 rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer; 196 197 private: 198 // Access to the copy constructor is private to force use of the Copy() 199 // method for those exceptional cases where we do use it. 200 Config(const Config&); 201 }; 202 203 // Updates the sending state for all simulcast layers that the video send 204 // stream owns. This can mean updating the activity one or for multiple 205 // layers. The ordering of active layers is the order in which the 206 // rtp modules are stored in the VideoSendStream. 207 // Note: This starts stream activity if it is inactive and one of the layers 208 // is active. This stops stream activity if it is active and all layers are 209 // inactive. 210 virtual void UpdateActiveSimulcastLayers( 211 const std::vector<bool> active_layers) = 0; 212 213 // Starts stream activity. 214 // When a stream is active, it can receive, process and deliver packets. 215 virtual void Start() = 0; 216 // Stops stream activity. 217 // When a stream is stopped, it can't receive, process or deliver packets. 218 virtual void Stop() = 0; 219 220 // If the resource is overusing, the VideoSendStream will try to reduce 221 // resolution or frame rate until no resource is overusing. 222 // TODO(https://crbug.com/webrtc/11565): When the ResourceAdaptationProcessor 223 // is moved to Call this method could be deleted altogether in favor of 224 // Call-level APIs only. 225 virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) = 0; 226 virtual std::vector<rtc::scoped_refptr<Resource>> 227 GetAdaptationResources() = 0; 228 229 virtual void SetSource( 230 rtc::VideoSourceInterface<webrtc::VideoFrame>* source, 231 const DegradationPreference& degradation_preference) = 0; 232 233 // Set which streams to send. Must have at least as many SSRCs as configured 234 // in the config. Encoder settings are passed on to the encoder instance along 235 // with the VideoStream settings. 236 virtual void ReconfigureVideoEncoder(VideoEncoderConfig config) = 0; 237 238 virtual Stats GetStats() = 0; 239 240 protected: ~VideoSendStream()241 virtual ~VideoSendStream() {} 242 }; 243 244 } // namespace webrtc 245 246 #endif // CALL_VIDEO_SEND_STREAM_H_ 247