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/transport.h" 22 #include "webrtc/video_renderer.h" 23 24 namespace webrtc { 25 26 namespace newapi { 27 // RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size 28 // RTCP mode is described by RFC 5506. 29 enum RtcpMode { kRtcpCompound, kRtcpReducedSize }; 30 } // namespace newapi 31 32 class VideoDecoder; 33 34 // TODO(mflodman) Move all these settings to VideoDecoder and move the 35 // declaration to common_types.h. 36 struct ExternalVideoDecoder { ExternalVideoDecoderExternalVideoDecoder37 ExternalVideoDecoder() 38 : decoder(NULL), payload_type(0), renderer(false), expected_delay_ms(0) {} 39 // The actual decoder. 40 VideoDecoder* decoder; 41 42 // Received RTP packets with this payload type will be sent to this decoder 43 // instance. 44 int payload_type; 45 46 // 'true' if the decoder handles rendering as well. 47 bool renderer; 48 49 // The expected delay for decoding and rendering, i.e. the frame will be 50 // delivered this many milliseconds, if possible, earlier than the ideal 51 // render time. 52 // Note: Ignored if 'renderer' is false. 53 int expected_delay_ms; 54 }; 55 56 class VideoReceiveStream { 57 public: 58 struct Stats : public StreamStats { StatsStats59 Stats() 60 : network_frame_rate(0), 61 decode_frame_rate(0), 62 render_frame_rate(0), 63 avg_delay_ms(0), 64 discarded_packets(0), 65 ssrc(0) {} 66 67 int network_frame_rate; 68 int decode_frame_rate; 69 int render_frame_rate; 70 int avg_delay_ms; 71 int discarded_packets; 72 uint32_t ssrc; 73 std::string c_name; 74 }; 75 76 struct Config { ConfigConfig77 Config() 78 : renderer(NULL), 79 render_delay_ms(0), 80 audio_channel_id(0), 81 pre_decode_callback(NULL), 82 pre_render_callback(NULL), 83 target_delay_ms(0) {} 84 // Codecs the receive stream can receive. 85 std::vector<VideoCodec> codecs; 86 87 // Receive-stream specific RTP settings. 88 struct Rtp { RtpConfig::Rtp89 Rtp() 90 : remote_ssrc(0), 91 local_ssrc(0), 92 rtcp_mode(newapi::kRtcpReducedSize), 93 remb(true) {} 94 95 // Synchronization source (stream identifier) to be received. 96 uint32_t remote_ssrc; 97 // Sender SSRC used for sending RTCP (such as receiver reports). 98 uint32_t local_ssrc; 99 100 // See RtcpMode for description. 101 newapi::RtcpMode rtcp_mode; 102 103 // Extended RTCP settings. 104 struct RtcpXr { RtcpXrConfig::Rtp::RtcpXr105 RtcpXr() : receiver_reference_time_report(false) {} 106 107 // True if RTCP Receiver Reference Time Report Block extension 108 // (RFC 3611) should be enabled. 109 bool receiver_reference_time_report; 110 } rtcp_xr; 111 112 // See draft-alvestrand-rmcat-remb for information. 113 bool remb; 114 115 // See NackConfig for description. 116 NackConfig nack; 117 118 // See FecConfig for description. 119 FecConfig fec; 120 121 // RTX settings for incoming video payloads that may be received. RTX is 122 // disabled if there's no config present. 123 struct Rtx { RtxConfig::Rtp::Rtx124 Rtx() : ssrc(0), payload_type(0) {} 125 126 // SSRCs to use for the RTX streams. 127 uint32_t ssrc; 128 129 // Payload type to use for the RTX stream. 130 int payload_type; 131 }; 132 133 // Map from video RTP payload type -> RTX config. 134 typedef std::map<int, Rtx> RtxMap; 135 RtxMap rtx; 136 137 // RTP header extensions used for the received stream. 138 std::vector<RtpExtension> extensions; 139 } rtp; 140 141 // VideoRenderer will be called for each decoded frame. 'NULL' disables 142 // rendering of this stream. 143 VideoRenderer* renderer; 144 145 // Expected delay needed by the renderer, i.e. the frame will be delivered 146 // this many milliseconds, if possible, earlier than the ideal render time. 147 // Only valid if 'renderer' is set. 148 int render_delay_ms; 149 150 // Audio channel corresponding to this video stream, used for audio/video 151 // synchronization. 'audio_channel_id' is ignored if no VoiceEngine is set 152 // when creating the VideoEngine instance. '-1' disables a/v sync. 153 int audio_channel_id; 154 155 // Called for each incoming video frame, i.e. in encoded state. E.g. used 156 // when 157 // saving the stream to a file. 'NULL' disables the callback. 158 EncodedFrameObserver* pre_decode_callback; 159 160 // Called for each decoded frame. E.g. used when adding effects to the 161 // decoded 162 // stream. 'NULL' disables the callback. 163 I420FrameCallback* pre_render_callback; 164 165 // External video decoders to be used if incoming payload type matches the 166 // registered type for an external decoder. 167 std::vector<ExternalVideoDecoder> external_decoders; 168 169 // Target delay in milliseconds. A positive value indicates this stream is 170 // used for streaming instead of a real-time call. 171 int target_delay_ms; 172 }; 173 174 virtual void Start() = 0; 175 virtual void Stop() = 0; 176 virtual Stats GetStats() const = 0; 177 178 // TODO(mflodman) Replace this with callback. 179 virtual void GetCurrentReceiveCodec(VideoCodec* receive_codec) = 0; 180 181 protected: ~VideoReceiveStream()182 virtual ~VideoReceiveStream() {} 183 }; 184 185 } // namespace webrtc 186 187 #endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_ 188