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_RECEIVE_STREAM_H_ 12 #define WEBRTC_VIDEO_RECEIVE_STREAM_H_ 13 14 #include <map> 15 #include <string> 16 #include <vector> 17 18 #include "webrtc/common_types.h" 19 #include "webrtc/config.h" 20 #include "webrtc/frame_callback.h" 21 #include "webrtc/stream.h" 22 #include "webrtc/transport.h" 23 #include "webrtc/video_renderer.h" 24 25 namespace webrtc { 26 27 class VideoDecoder; 28 29 class VideoReceiveStream : public ReceiveStream { 30 public: 31 // TODO(mflodman) Move all these settings to VideoDecoder and move the 32 // declaration to common_types.h. 33 struct Decoder { 34 std::string ToString() const; 35 36 // The actual decoder instance. 37 VideoDecoder* decoder = nullptr; 38 39 // Received RTP packets with this payload type will be sent to this decoder 40 // instance. 41 int payload_type = 0; 42 43 // Name of the decoded payload (such as VP8). Maps back to the depacketizer 44 // used to unpack incoming packets. 45 std::string payload_name; 46 }; 47 48 struct Stats { 49 int network_frame_rate = 0; 50 int decode_frame_rate = 0; 51 int render_frame_rate = 0; 52 53 // Decoder stats. 54 std::string decoder_implementation_name = "unknown"; 55 FrameCounts frame_counts; 56 int decode_ms = 0; 57 int max_decode_ms = 0; 58 int current_delay_ms = 0; 59 int target_delay_ms = 0; 60 int jitter_buffer_ms = 0; 61 int min_playout_delay_ms = 0; 62 int render_delay_ms = 10; 63 64 int current_payload_type = -1; 65 66 int total_bitrate_bps = 0; 67 int discarded_packets = 0; 68 69 uint32_t ssrc = 0; 70 std::string c_name; 71 StreamDataCounters rtp_stats; 72 RtcpPacketTypeCounter rtcp_packet_type_counts; 73 RtcpStatistics rtcp_stats; 74 }; 75 76 struct Config { 77 Config() = delete; ConfigConfig78 explicit Config(Transport* rtcp_send_transport) 79 : rtcp_send_transport(rtcp_send_transport) {} 80 81 std::string ToString() const; 82 83 // Decoders for every payload that we can receive. 84 std::vector<Decoder> decoders; 85 86 // Receive-stream specific RTP settings. 87 struct Rtp { 88 std::string ToString() const; 89 90 // Synchronization source (stream identifier) to be received. 91 uint32_t remote_ssrc = 0; 92 // Sender SSRC used for sending RTCP (such as receiver reports). 93 uint32_t local_ssrc = 0; 94 95 // See RtcpMode for description. 96 RtcpMode rtcp_mode = RtcpMode::kCompound; 97 98 // Extended RTCP settings. 99 struct RtcpXr { 100 // True if RTCP Receiver Reference Time Report Block extension 101 // (RFC 3611) should be enabled. 102 bool receiver_reference_time_report = false; 103 } rtcp_xr; 104 105 // See draft-alvestrand-rmcat-remb for information. 106 bool remb = false; 107 108 // See draft-holmer-rmcat-transport-wide-cc-extensions for details. 109 bool transport_cc = false; 110 111 // See NackConfig for description. 112 NackConfig nack; 113 114 // See FecConfig for description. 115 FecConfig fec; 116 117 // RTX settings for incoming video payloads that may be received. RTX is 118 // disabled if there's no config present. 119 struct Rtx { 120 // SSRCs to use for the RTX streams. 121 uint32_t ssrc = 0; 122 123 // Payload type to use for the RTX stream. 124 int payload_type = 0; 125 }; 126 127 // Map from video RTP payload type -> RTX config. 128 typedef std::map<int, Rtx> RtxMap; 129 RtxMap rtx; 130 131 // If set to true, the RTX payload type mapping supplied in |rtx| will be 132 // used when restoring RTX packets. Without it, RTX packets will always be 133 // restored to the last non-RTX packet payload type received. 134 bool use_rtx_payload_mapping_on_restore = false; 135 136 // RTP header extensions used for the received stream. 137 std::vector<RtpExtension> extensions; 138 } rtp; 139 140 // Transport for outgoing packets (RTCP). 141 Transport* rtcp_send_transport = nullptr; 142 143 // VideoRenderer will be called for each decoded frame. 'nullptr' disables 144 // rendering of this stream. 145 VideoRenderer* renderer = nullptr; 146 147 // Expected delay needed by the renderer, i.e. the frame will be delivered 148 // this many milliseconds, if possible, earlier than the ideal render time. 149 // Only valid if 'renderer' is set. 150 int render_delay_ms = 10; 151 152 // Identifier for an A/V synchronization group. Empty string to disable. 153 // TODO(pbos): Synchronize streams in a sync group, not just video streams 154 // to one of the audio streams. 155 std::string sync_group; 156 157 // Called for each incoming video frame, i.e. in encoded state. E.g. used 158 // when 159 // saving the stream to a file. 'nullptr' disables the callback. 160 EncodedFrameObserver* pre_decode_callback = nullptr; 161 162 // Called for each decoded frame. E.g. used when adding effects to the 163 // decoded 164 // stream. 'nullptr' disables the callback. 165 I420FrameCallback* pre_render_callback = nullptr; 166 167 // Target delay in milliseconds. A positive value indicates this stream is 168 // used for streaming instead of a real-time call. 169 int target_delay_ms = 0; 170 }; 171 172 // TODO(pbos): Add info on currently-received codec to Stats. 173 virtual Stats GetStats() const = 0; 174 }; 175 176 } // namespace webrtc 177 178 #endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_ 179