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      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 {
     25   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 {
     37   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 {
     54   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 {
     65   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 {
     76   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 {
     87   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 
    117   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