| /* |
| * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| // TODO(pbos): Move Config from common.h to here. |
| |
| #ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CONFIG_H_ |
| #define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CONFIG_H_ |
| |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/common_types.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| struct RtpStatistics { |
| RtpStatistics() |
| : ssrc(0), |
| fraction_loss(0), |
| cumulative_loss(0), |
| extended_max_sequence_number(0) {} |
| uint32_t ssrc; |
| int fraction_loss; |
| int cumulative_loss; |
| int extended_max_sequence_number; |
| std::string c_name; |
| }; |
| |
| struct StreamStats { |
| StreamStats() : key_frames(0), delta_frames(0), bitrate_bps(0) {} |
| uint32_t key_frames; |
| uint32_t delta_frames; |
| int32_t bitrate_bps; |
| StreamDataCounters rtp_stats; |
| RtcpStatistics rtcp_stats; |
| }; |
| |
| // Settings for NACK, see RFC 4585 for details. |
| struct NackConfig { |
| NackConfig() : rtp_history_ms(0) {} |
| // Send side: the time RTP packets are stored for retransmissions. |
| // Receive side: the time the receiver is prepared to wait for |
| // retransmissions. |
| // Set to '0' to disable. |
| int rtp_history_ms; |
| }; |
| |
| // Settings for forward error correction, see RFC 5109 for details. Set the |
| // payload types to '-1' to disable. |
| struct FecConfig { |
| FecConfig() : ulpfec_payload_type(-1), red_payload_type(-1) {} |
| // Payload type used for ULPFEC packets. |
| int ulpfec_payload_type; |
| |
| // Payload type used for RED packets. |
| int red_payload_type; |
| }; |
| |
| // RTP header extension to use for the video stream, see RFC 5285. |
| struct RtpExtension { |
| static const char* kTOffset; |
| static const char* kAbsSendTime; |
| RtpExtension(const char* name, int id) : name(name), id(id) {} |
| // TODO(mflodman) Add API to query supported extensions. |
| std::string name; |
| int id; |
| }; |
| } // namespace webrtc |
| |
| #endif // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CONFIG_H_ |