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 // TODO(pbos): Move Config from common.h to here. 12 13 #ifndef WEBRTC_CONFIG_H_ 14 #define WEBRTC_CONFIG_H_ 15 16 #include <string> 17 #include <vector> 18 19 #include "webrtc/common_types.h" 20 #include "webrtc/typedefs.h" 21 22 namespace webrtc { 23 24 struct RtpStatistics { RtpStatisticsRtpStatistics25 RtpStatistics() 26 : ssrc(0), 27 fraction_loss(0), 28 cumulative_loss(0), 29 extended_max_sequence_number(0) {} 30 uint32_t ssrc; 31 int fraction_loss; 32 int cumulative_loss; 33 int extended_max_sequence_number; 34 }; 35 36 struct StreamStats { StreamStatsStreamStats37 StreamStats() 38 : key_frames(0), 39 delta_frames(0), 40 bitrate_bps(0), 41 avg_delay_ms(0), 42 max_delay_ms(0) {} 43 uint32_t key_frames; 44 uint32_t delta_frames; 45 int32_t bitrate_bps; 46 int avg_delay_ms; 47 int max_delay_ms; 48 StreamDataCounters rtp_stats; 49 RtcpStatistics rtcp_stats; 50 }; 51 52 // Settings for NACK, see RFC 4585 for details. 53 struct NackConfig { NackConfigNackConfig54 NackConfig() : rtp_history_ms(0) {} 55 // Send side: the time RTP packets are stored for retransmissions. 56 // Receive side: the time the receiver is prepared to wait for 57 // retransmissions. 58 // Set to '0' to disable. 59 int rtp_history_ms; 60 }; 61 62 // Settings for forward error correction, see RFC 5109 for details. Set the 63 // payload types to '-1' to disable. 64 struct FecConfig { FecConfigFecConfig65 FecConfig() : ulpfec_payload_type(-1), red_payload_type(-1) {} 66 std::string ToString() const; 67 // Payload type used for ULPFEC packets. 68 int ulpfec_payload_type; 69 70 // Payload type used for RED packets. 71 int red_payload_type; 72 }; 73 74 // RTP header extension to use for the video stream, see RFC 5285. 75 struct RtpExtension { RtpExtensionRtpExtension76 RtpExtension(const std::string& name, int id) : name(name), id(id) {} 77 std::string ToString() const; 78 static bool IsSupported(const std::string& name); 79 80 static const char* kTOffset; 81 static const char* kAbsSendTime; 82 std::string name; 83 int id; 84 }; 85 86 struct VideoStream { VideoStreamVideoStream87 VideoStream() 88 : width(0), 89 height(0), 90 max_framerate(-1), 91 min_bitrate_bps(-1), 92 target_bitrate_bps(-1), 93 max_bitrate_bps(-1), 94 max_qp(-1) {} 95 std::string ToString() const; 96 97 size_t width; 98 size_t height; 99 int max_framerate; 100 101 int min_bitrate_bps; 102 int target_bitrate_bps; 103 int max_bitrate_bps; 104 105 int max_qp; 106 107 // Bitrate thresholds for enabling additional temporal layers. 108 std::vector<int> temporal_layers; 109 }; 110 111 struct VideoEncoderConfig { 112 enum ContentType { 113 kRealtimeVideo, 114 kScreenshare, 115 }; 116 VideoEncoderConfigVideoEncoderConfig117 VideoEncoderConfig() 118 : content_type(kRealtimeVideo), encoder_specific_settings(NULL) {} 119 120 std::vector<VideoStream> streams; 121 ContentType content_type; 122 void* encoder_specific_settings; 123 }; 124 125 } // namespace webrtc 126 127 #endif // WEBRTC_CONFIG_H_ 128