webrtc_sys/
rtp_parameters.rs

1// Copyright 2025 LiveKit, Inc.
2//
3// Licensed under the Apache License, Version 2.0 (the "License");
4// you may not use this file except in compliance with the License.
5// You may obtain a copy of the License at
6//
7//     http://www.apache.org/licenses/LICENSE-2.0
8//
9// Unless required by applicable law or agreed to in writing, software
10// distributed under the License is distributed on an "AS IS" BASIS,
11// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
12// See the License for the specific language governing permissions and
13// limitations under the License.
14
15pub const DEFAULT_BITRATE_PRIORITY: f64 = 1.0;
16
17#[cxx::bridge(namespace = "livekit_ffi")]
18pub mod ffi {
19
20    // Used to replace std::map
21    #[derive(Debug)]
22    pub struct StringKeyValue {
23        pub key: String,
24        pub value: String,
25    }
26
27    #[derive(Debug)]
28    #[repr(i32)]
29    pub enum FecMechanism {
30        Red,
31        RedAndUlpfec,
32        FlexFec,
33    }
34
35    #[derive(Debug)]
36    #[repr(i32)]
37    pub enum RtcpFeedbackType {
38        Ccm,
39        Lntf,
40        Nack,
41        Remb,
42        TransportCC,
43    }
44
45    #[derive(Debug)]
46    #[repr(i32)]
47    pub enum RtcpFeedbackMessageType {
48        GenericNack,
49        Pli,
50        Fir,
51    }
52
53    #[derive(Debug)]
54    #[repr(i32)]
55    pub enum DegradationPreference {
56        /// Maintain both framerate and resolution. Frames may be dropped before encoding
57        /// if necessary to avoid overusing network and encoder resources.
58        /// Note: This was previously called `Disabled` in older WebRTC versions.
59        MaintainFramerateAndResolution,
60        MaintainFramerate,
61        MaintainResolution,
62        Balanced,
63    }
64
65    #[derive(Debug)]
66    pub struct RtcpFeedback {
67        pub feedback_type: RtcpFeedbackType,
68        pub has_message_type: bool,
69        pub message_type: RtcpFeedbackMessageType,
70    }
71
72    #[derive(Debug)]
73    pub struct RtpCodecCapability {
74        pub mime_type: String, // filled with mime_type fnc
75        pub name: String,
76        pub kind: MediaType,
77        pub has_clock_rate: bool,
78        pub clock_rate: i32,
79        pub has_preferred_payload_type: bool,
80        pub preferred_payload_type: i32,
81        pub has_num_channels: bool,
82        pub num_channels: i32,
83        pub rtcp_feedback: Vec<RtcpFeedback>,
84        pub parameters: Vec<StringKeyValue>,
85    }
86
87    #[derive(Debug)]
88    pub struct RtpHeaderExtensionCapability {
89        pub uri: String,
90        pub has_preferred_id: bool,
91        pub preferred_id: i32,
92        pub preferred_encrypt: bool,
93        pub direction: RtpTransceiverDirection,
94    }
95
96    #[derive(Debug)]
97    #[repr(i32)]
98    pub enum RtpExtensionFilter {
99        DiscardEncryptedExtension,
100        PreferEncryptedExtension,
101        RequireEncryptedExtension,
102    }
103
104    #[derive(Debug)]
105    pub struct RtpExtension {
106        // TODO(theomonnom): export available URI inside api/rtp_parameters.h
107        pub uri: String,
108        pub id: i32,
109        pub encrypt: bool,
110    }
111
112    #[derive(Debug)]
113    pub struct RtpFecParameters {
114        pub has_ssrc: bool,
115        pub ssrc: u32,
116        pub mechanism: FecMechanism,
117    }
118
119    #[derive(Debug)]
120    pub struct RtpRtxParameters {
121        pub has_ssrc: bool,
122        pub ssrc: u32,
123    }
124
125    #[derive(Debug)]
126    pub struct RtpEncodingParameters {
127        pub has_ssrc: bool,
128        pub ssrc: u32,
129        pub bitrate_priority: f64,
130        pub network_priority: Priority, // Todo link type
131        pub has_max_bitrate_bps: bool,
132        pub max_bitrate_bps: i32,
133        pub has_min_bitrate_bps: bool,
134        pub min_bitrate_bps: i32,
135        pub has_max_framerate: bool,
136        pub max_framerate: f64,
137        pub has_num_temporal_layers: bool,
138        pub num_temporal_layers: i32,
139        pub has_scale_resolution_down_by: bool,
140        pub scale_resolution_down_by: f64,
141        pub has_scalability_mode: bool,
142        pub scalability_mode: String,
143        pub active: bool,
144        pub rid: String,
145        pub adaptive_ptime: bool,
146    }
147
148    #[derive(Debug)]
149    pub struct RtpCodecParameters {
150        pub mime_type: String, // filled with mime_type fnc
151        pub name: String,
152        pub kind: MediaType,
153        pub payload_type: i32,
154        pub has_clock_rate: bool,
155        pub clock_rate: i32,
156        pub has_num_channels: bool,
157        pub num_channels: i32,
158        pub has_max_ptime: bool,
159        pub max_ptime: i32,
160        pub has_ptime: bool,
161        pub ptime: i32,
162        pub rtcp_feedback: Vec<RtcpFeedback>,
163        pub parameters: Vec<StringKeyValue>,
164    }
165
166    #[derive(Debug)]
167    pub struct RtpCapabilities {
168        pub codecs: Vec<RtpCodecCapability>,
169        pub header_extensions: Vec<RtpHeaderExtensionCapability>,
170        pub fec: Vec<FecMechanism>,
171    }
172
173    #[derive(Debug)]
174    pub struct RtcpParameters {
175        pub has_ssrc: bool,
176        pub ssrc: u32,
177        pub cname: String,
178        pub reduced_size: bool,
179        pub mux: bool,
180    }
181
182    #[derive(Debug)]
183    pub struct RtpParameters {
184        pub transaction_id: String,
185        pub mid: String,
186        pub codecs: Vec<RtpCodecParameters>,
187        pub header_extensions: Vec<RtpExtension>,
188        pub encodings: Vec<RtpEncodingParameters>,
189        pub rtcp: RtcpParameters,
190        pub has_degradation_preference: bool,
191        pub degradation_preference: DegradationPreference,
192    }
193
194    extern "C++" {
195        include!("livekit/webrtc.h");
196
197        type Priority = crate::webrtc::ffi::Priority;
198        type MediaType = crate::webrtc::ffi::MediaType;
199        type RtpTransceiverDirection = crate::webrtc::ffi::RtpTransceiverDirection;
200    }
201}
webrtc_sys/rtp_parameters.rs

webrtc_sys/
rtp_parameters.rs
