| /* | 
 |  *  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_CONFIG_H_ | 
 | #define WEBRTC_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; | 
 | }; | 
 |  | 
 | struct StreamStats { | 
 |   StreamStats() | 
 |       : key_frames(0), | 
 |         delta_frames(0), | 
 |         bitrate_bps(0), | 
 |         avg_delay_ms(0), | 
 |         max_delay_ms(0) {} | 
 |   uint32_t key_frames; | 
 |   uint32_t delta_frames; | 
 |   int32_t bitrate_bps; | 
 |   int avg_delay_ms; | 
 |   int max_delay_ms; | 
 |   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) {} | 
 |   std::string ToString() const; | 
 |   // 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 { | 
 |   RtpExtension(const std::string& name, int id) : name(name), id(id) {} | 
 |   std::string ToString() const; | 
 |   static bool IsSupported(const std::string& name); | 
 |  | 
 |   static const char* kTOffset; | 
 |   static const char* kAbsSendTime; | 
 |   std::string name; | 
 |   int id; | 
 | }; | 
 |  | 
 | struct VideoStream { | 
 |   VideoStream() | 
 |       : width(0), | 
 |         height(0), | 
 |         max_framerate(-1), | 
 |         min_bitrate_bps(-1), | 
 |         target_bitrate_bps(-1), | 
 |         max_bitrate_bps(-1), | 
 |         max_qp(-1) {} | 
 |   std::string ToString() const; | 
 |  | 
 |   size_t width; | 
 |   size_t height; | 
 |   int max_framerate; | 
 |  | 
 |   int min_bitrate_bps; | 
 |   int target_bitrate_bps; | 
 |   int max_bitrate_bps; | 
 |  | 
 |   int max_qp; | 
 |  | 
 |   // Bitrate thresholds for enabling additional temporal layers. | 
 |   std::vector<int> temporal_layers; | 
 | }; | 
 |  | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // WEBRTC_CONFIG_H_ |