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 WEBRTC_VIDEO_SEND_STREAM_H_ 12 #define WEBRTC_VIDEO_SEND_STREAM_H_ 13 14 #include <map> 15 #include <string> 16 17 #include "webrtc/common_types.h" 18 #include "webrtc/config.h" 19 #include "webrtc/frame_callback.h" 20 #include "webrtc/stream.h" 21 #include "webrtc/transport.h" 22 #include "webrtc/video_renderer.h" 23 24 namespace webrtc { 25 26 class LoadObserver; 27 class VideoEncoder; 28 29 class EncodingTimeObserver { 30 public: ~EncodingTimeObserver()31 virtual ~EncodingTimeObserver() {} 32 33 virtual void OnReportEncodedTime(int64_t ntp_time_ms, int encode_time_ms) = 0; 34 }; 35 36 // Class to deliver captured frame to the video send stream. 37 class VideoCaptureInput { 38 public: 39 // These methods do not lock internally and must be called sequentially. 40 // If your application switches input sources synchronization must be done 41 // externally to make sure that any old frames are not delivered concurrently. 42 virtual void IncomingCapturedFrame(const VideoFrame& video_frame) = 0; 43 44 protected: ~VideoCaptureInput()45 virtual ~VideoCaptureInput() {} 46 }; 47 48 class VideoSendStream : public SendStream { 49 public: 50 struct StreamStats { 51 FrameCounts frame_counts; 52 int width = 0; 53 int height = 0; 54 // TODO(holmer): Move bitrate_bps out to the webrtc::Call layer. 55 int total_bitrate_bps = 0; 56 int retransmit_bitrate_bps = 0; 57 int avg_delay_ms = 0; 58 int max_delay_ms = 0; 59 StreamDataCounters rtp_stats; 60 RtcpPacketTypeCounter rtcp_packet_type_counts; 61 RtcpStatistics rtcp_stats; 62 }; 63 64 struct Stats { 65 std::string encoder_implementation_name = "unknown"; 66 int input_frame_rate = 0; 67 int encode_frame_rate = 0; 68 int avg_encode_time_ms = 0; 69 int encode_usage_percent = 0; 70 int target_media_bitrate_bps = 0; 71 int media_bitrate_bps = 0; 72 bool suspended = false; 73 bool bw_limited_resolution = false; 74 std::map<uint32_t, StreamStats> substreams; 75 }; 76 77 struct Config { 78 Config() = delete; ConfigConfig79 explicit Config(Transport* send_transport) 80 : send_transport(send_transport) {} 81 82 std::string ToString() const; 83 84 struct EncoderSettings { 85 std::string ToString() const; 86 87 std::string payload_name; 88 int payload_type = -1; 89 90 // TODO(sophiechang): Delete this field when no one is using internal 91 // sources anymore. 92 bool internal_source = false; 93 94 // Uninitialized VideoEncoder instance to be used for encoding. Will be 95 // initialized from inside the VideoSendStream. 96 VideoEncoder* encoder = nullptr; 97 } encoder_settings; 98 99 static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4. 100 struct Rtp { 101 std::string ToString() const; 102 103 std::vector<uint32_t> ssrcs; 104 105 // See RtcpMode for description. 106 RtcpMode rtcp_mode = RtcpMode::kCompound; 107 108 // Max RTP packet size delivered to send transport from VideoEngine. 109 size_t max_packet_size = kDefaultMaxPacketSize; 110 111 // RTP header extensions to use for this send stream. 112 std::vector<RtpExtension> extensions; 113 114 // See NackConfig for description. 115 NackConfig nack; 116 117 // See FecConfig for description. 118 FecConfig fec; 119 120 // Settings for RTP retransmission payload format, see RFC 4588 for 121 // details. 122 struct Rtx { 123 std::string ToString() const; 124 // SSRCs to use for the RTX streams. 125 std::vector<uint32_t> ssrcs; 126 127 // Payload type to use for the RTX stream. 128 int payload_type = -1; 129 } rtx; 130 131 // RTCP CNAME, see RFC 3550. 132 std::string c_name; 133 } rtp; 134 135 // Transport for outgoing packets. 136 Transport* send_transport = nullptr; 137 138 // Callback for overuse and normal usage based on the jitter of incoming 139 // captured frames. 'nullptr' disables the callback. 140 LoadObserver* overuse_callback = nullptr; 141 142 // Called for each I420 frame before encoding the frame. Can be used for 143 // effects, snapshots etc. 'nullptr' disables the callback. 144 I420FrameCallback* pre_encode_callback = nullptr; 145 146 // Called for each encoded frame, e.g. used for file storage. 'nullptr' 147 // disables the callback. 148 EncodedFrameObserver* post_encode_callback = nullptr; 149 150 // Renderer for local preview. The local renderer will be called even if 151 // sending hasn't started. 'nullptr' disables local rendering. 152 VideoRenderer* local_renderer = nullptr; 153 154 // Expected delay needed by the renderer, i.e. the frame will be delivered 155 // this many milliseconds, if possible, earlier than expected render time. 156 // Only valid if |local_renderer| is set. 157 int render_delay_ms = 0; 158 159 // Target delay in milliseconds. A positive value indicates this stream is 160 // used for streaming instead of a real-time call. 161 int target_delay_ms = 0; 162 163 // True if the stream should be suspended when the available bitrate fall 164 // below the minimum configured bitrate. If this variable is false, the 165 // stream may send at a rate higher than the estimated available bitrate. 166 bool suspend_below_min_bitrate = false; 167 168 // Called for each encoded frame. Passes the total time spent on encoding. 169 // TODO(ivica): Consolidate with post_encode_callback: 170 // https://code.google.com/p/webrtc/issues/detail?id=5042 171 EncodingTimeObserver* encoding_time_observer = nullptr; 172 }; 173 174 // Gets interface used to insert captured frames. Valid as long as the 175 // VideoSendStream is valid. 176 virtual VideoCaptureInput* Input() = 0; 177 178 // Set which streams to send. Must have at least as many SSRCs as configured 179 // in the config. Encoder settings are passed on to the encoder instance along 180 // with the VideoStream settings. 181 virtual bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0; 182 183 virtual Stats GetStats() = 0; 184 }; 185 186 } // namespace webrtc 187 188 #endif // WEBRTC_VIDEO_SEND_STREAM_H_ 189