From 74670b5019f3b264415393a0ea5bd4c7aab8d1be Mon Sep 17 00:00:00 2001
From: Wojciech Kozyra <wojciech.kozyra@swmansion.com>
Date: Fri, 26 Jan 2024 12:32:08 +0100
Subject: [PATCH] [docs] Fix inconsistent header sizing in routes page (#362)

---
 docs/pages/api/routes.md | 68 +++++++++++++++++++++++++++++++++++++---
 1 file changed, 64 insertions(+), 4 deletions(-)

diff --git a/docs/pages/api/routes.md b/docs/pages/api/routes.md
index d0eb1c356..7ffc22418 100644
--- a/docs/pages/api/routes.md
+++ b/docs/pages/api/routes.md
@@ -2,8 +2,6 @@
 description: API routes to configure the compositor.
 ---
 
-import Docs from "@site/pages/api/generated/renderer-RegisterInputRequest.md"
-
 # Routes
 
 ### Start
@@ -38,7 +36,69 @@ type OutputScene = {
 
 ***
 
-<Docs />
+### RegisterInputRequest
+
+```typescript
+type RegisterInputStream = {
+  type: "register";
+  entity_type: "input_stream";
+  input_id: string;
+  port: Port;
+  video?: Video;
+  audio?: Audio;
+}
+```
+
+Parameters of registered RTP input stream. Before using input in video composition or output mixing, input has to be firstly registered using `register_input` request.
+
+At least one of `video` and `audio` has to be defined.
+
+- `input_id` - An identifier for the input stream.
+- `port` - UDP port or port range on which the compositor should listen for the stream.
+- `video` - Parameters of a video source included in the RTP stream.
+- `audio` - Parameters of an audio source included in the RTP stream.
+
+#### Port
+
+```typescript
+type Port = string | u16
+```
+
+#### Video
+
+```typescript
+type Video = {
+  codec?: "h264";
+  rtp_payload_type?: u8;
+}
+```
+
+- `codec` - (**default=`"h264"`**) Video codec.
+  - `"h264"` - H264 video.
+- `rtp_payload_type` - (**default=`96`**) Value of payload type field in received RTP packets.
+  Packets with different payload type won't be treated as video and included in composing. Values should be in [0, 64] or [96, 255]. Values in range [65, 95] can't be used. For more information, see [RFC](https://datatracker.ietf.org/doc/html/rfc5761#section-4) Packets with different payload type won't be treated as video and included in composing.
+
+#### Audio
+
+```typescript
+type Audio = {
+  codec?: "opus";
+  sample_rate: u32;
+  channels: "mono" | "stereo";
+  rtp_payload_type?: u8;
+  forward_error_correction?: bool;
+}
+```
+
+- `codec` - (**default=`"opus"`**) Audio codec.
+  - `"opus"` - Opus audio.
+- `sample_rate` - Sample rate. If the specified sample rate doesn't match real sample rate, audio won't be mixed properly.
+- `channels` - Audio channels.
+  - `"mono"` - Mono audio (single channel).
+  - `"stereo"` - Stereo audio (two channels).
+- `rtp_payload_type` - (**default=`97`**) Value of payload type field in received RTP packets.
+  Packets with different payload type won't be treated as audio and included in mixing. Values should be in range [0, 64] or [96, 255]. Values in range [65, 95] can't be used. For more information, check out [RFC](https://datatracker.ietf.org/doc/html/rfc5761#section-4).
+- `forward_error_correction` - (**default=`"false"`**) Specifies whether the stream uses forward error correction. It's specific for Opus codec. For more information, check out [RFC](https://datatracker.ietf.org/doc/html/rfc6716#section-2.1.7).
 
 ***
 
@@ -107,4 +167,4 @@ type Unregister =
   | { type: "unregister", entity_type: "shader", shader_id: string }
   | { type: "unregister", entity_type: "image", image_id: string }
   | { type: "unregister", entity_type: "web_renderer", instance_id: string }
-```
\ No newline at end of file
+```