feat: add ssrc definition (op 12)

src/types/events/voice_gateway/mod.rs

@@ -12,6 +12,7 @@ pub use select_protocol::*;
 pub use session_description::*;
 pub use speaking::*;
 pub use voice_backend_version::*;
+pub use ssrc_definition::*;
 
 mod client_connect;
 mod client_disconnect;
@@ -23,6 +24,7 @@ mod select_protocol;
 mod session_description;
 mod speaking;
 mod voice_backend_version;
+mod ssrc_definition;
 
 #[derive(Debug, Default, Serialize, Clone)]
 /// The payload used for sending events to the webrtc gateway.
@@ -102,8 +104,7 @@ pub const VOICE_HEARTBEAT_ACK: u8 = 6;
 pub const VOICE_RESUME: u8 = 7;
 pub const VOICE_HELLO: u8 = 8;
 pub const VOICE_RESUMED: u8 = 9;
-/// See <https://discord-userdoccers.vercel.app/topics/opcodes-and-status-codes#voice-opcodes>
-pub const VOICE_VIDEO: u8 = 12;
+pub const VOICE_SSRC_DEFINITION: u8 = 12;
 pub const VOICE_CLIENT_DISCONNECT: u8 = 13;
 pub const VOICE_SESSION_UPDATE: u8 = 14;
 

src/types/events/voice_gateway/ssrc_definition.rs

@@ -0,0 +1,39 @@
+use crate::types::{Snowflake, WebSocketEvent};
+use serde::{Serialize, Deserialize};
+
+/// Defines an event which provides ssrcs for voice and video for a user id.
+///
+/// This event is sent via opcode 12.
+///
+/// Examples of the event:
+///
+/// When receiving:
+/// ```
+/// {"op":12,"d":{"video_ssrc":0,"user_id":"463640391196082177","streams":[{"ssrc":26595,"rtx_ssrc":26596,"rid":"100","quality":100,"max_resolution":{"width":1280,"type":"fixed","height":720},"max_framerate":30,"active":false}],"audio_ssrc":26597}}{"op":12,"d":{"video_ssrc":0,"user_id":"463640391196082177","streams":[{"ssrc":26595,"rtx_ssrc":26596,"rid":"100","quality":100,"max_resolution":{"width":1280,"type":"fixed","height":720},"max_framerate":30,"active":false}],"audio_ssrc":26597}}
+/// ```
+///
+/// When sending:
+/// ```
+/// {"op":12,"d":{"audio_ssrc":2307250864,"video_ssrc":0,"rtx_ssrc":0,"streams":[{"type":"video","rid":"100","ssrc":26595,"active":false,"quality":100,"rtx_ssrc":26596,"max_bitrate":2500000,"max_framerate":30,"max_resolution":{"type":"fixed","width":1280,"height":720}}]}}
+/// ```
+#[derive(Debug, Deserialize, Serialize, Default, Clone, PartialEq, Eq)]
+pub struct SsrcDefinition {
+    /// The ssrc used for video communications.
+    ///
+    /// Is always sent and received, though is 0 if describing only the audio ssrc.
+    #[serde(default)]
+    pub video_ssrc: usize,
+    /// The ssrc used for audio communications.
+    ///
+    /// Is always sent and received, though is 0 if describing only the video ssrc.
+    #[serde(default)]
+    pub audio_ssrc: usize,
+    /// The user id these ssrcs apply to.
+    ///
+    /// Is never sent by the user and is filled in by the server
+    #[serde(skip_serializing)]
+    pub user_id: Option<Snowflake>,
+    // TODO: Add video streams
+}
+
+impl WebSocketEvent for SsrcDefinition {}

src/voice/gateway.rs

@@ -17,11 +17,12 @@ use tokio_tungstenite::{connect_async_tls_with_config, Connector, WebSocketStrea
 use crate::errors::VoiceGatewayError;
 use crate::gateway::{heartbeat::HEARTBEAT_ACK_TIMEOUT, GatewayEvent};
 use crate::types::{
-    self, SelectProtocol, Speaking, VoiceGatewayReceivePayload, VoiceGatewaySendPayload,
-    VoiceIdentify, WebSocketEvent, VOICE_BACKEND_VERSION, VOICE_CLIENT_CONNECT_FLAGS,
-    VOICE_CLIENT_CONNECT_PLATFORM, VOICE_CLIENT_DISCONNECT, VOICE_HEARTBEAT, VOICE_HEARTBEAT_ACK,
-    VOICE_HELLO, VOICE_IDENTIFY, VOICE_MEDIA_SINK_WANTS, VOICE_READY, VOICE_RESUME,
-    VOICE_SELECT_PROTOCOL, VOICE_SESSION_DESCRIPTION, VOICE_SESSION_UPDATE, VOICE_SPEAKING,
+    self, SelectProtocol, Speaking, SsrcDefinition, VoiceGatewayReceivePayload,
+    VoiceGatewaySendPayload, VoiceIdentify, WebSocketEvent, VOICE_BACKEND_VERSION,
+    VOICE_CLIENT_CONNECT_FLAGS, VOICE_CLIENT_CONNECT_PLATFORM, VOICE_CLIENT_DISCONNECT,
+    VOICE_HEARTBEAT, VOICE_HEARTBEAT_ACK, VOICE_HELLO, VOICE_IDENTIFY, VOICE_MEDIA_SINK_WANTS,
+    VOICE_READY, VOICE_RESUME, VOICE_SELECT_PROTOCOL, VOICE_SESSION_DESCRIPTION,
+    VOICE_SESSION_UPDATE, VOICE_SPEAKING, VOICE_SSRC_DEFINITION,
 };
 
 use self::voice_events::VoiceEvents;
@@ -159,6 +160,15 @@ impl VoiceGatewayHandle {
         self.send_json(VOICE_SPEAKING, to_send_value).await;
     }
 
+    /// Sends an ssrc definition event
+    pub async fn send_ssrc_definition(&self, to_send: SsrcDefinition) {
+        let to_send_value = serde_json::to_value(&to_send).unwrap();
+
+        trace!("VGW: Sending SsrcDefinition");
+
+        self.send_json(VOICE_SSRC_DEFINITION, to_send_value).await;
+    }
+
     /// Sends a voice backend version request to the gateway
     pub async fn send_voice_backend_version_request(&self) {
         let data_empty_object = json!("{}");
@@ -408,6 +418,18 @@ impl VoiceGateway {
                     warn!("Failed to parse VOICE_SPEAKING ({})", result.err().unwrap());
                 }
             }
+            VOICE_SSRC_DEFINITION => {
+                trace!("VGW: Received Ssrc Definition");
+
+                let event = &mut self.events.lock().await.ssrc_definition;
+                let result = VoiceGateway::handle_event(gateway_payload.data.get(), event).await;
+                if result.is_err() {
+                    warn!(
+                        "Failed to parse VOICE_SSRC_DEFINITION ({})",
+                        result.err().unwrap()
+                    );
+                }
+            }
             VOICE_CLIENT_DISCONNECT => {
                 trace!("VGW: Received Client Disconnect");
 
@@ -663,6 +685,7 @@ pub mod voice_events {
         pub session_description: GatewayEvent<SessionDescription>,
         pub session_update: GatewayEvent<SessionUpdate>,
         pub speaking: GatewayEvent<Speaking>,
+        pub ssrc_definition: GatewayEvent<SsrcDefinition>,
         pub client_disconnect: GatewayEvent<VoiceClientDisconnection>,
         pub client_connect_flags: GatewayEvent<VoiceClientConnectFlags>,
         pub client_connect_platform: GatewayEvent<VoiceClientConnectPlatform>,