Some docs typos

docs/advanced-configuration.md

@@ -24,9 +24,8 @@ webrtc = Stream(
 
 !!! warning
 
-    WebRTC may not enforce your constaints. For example, it may rescale your video
-    (while keeping the same resolution) in order to maintain the desired (or reach a better) frame rate. If you
-    really want to enforce height, width and resolution constraints, use the `rtp_params` parameter as set `"degradationPreference": "maintain-resolution"`. 
+    WebRTC may not enforce your constraints. For example, it may rescale your video
+    (while keeping the same resolution) in order to maintain the desired frame rate (or reach a better one). If you really want to enforce height, width and resolution constraints, use the `rtp_params` parameter as set `"degradationPreference": "maintain-resolution"`. 
 
     ```python
     image = Stream(

docs/deployment.md

@@ -66,7 +66,7 @@ Stream(
 )
 ```
 
-!!! tip "Automatic Login"
+!!! tip "Automatic login"
 
     You can log in automatically with the `get_twilio_turn_credentials` helper
 

docs/faq.md

@@ -6,7 +6,7 @@ Make sure you are using a TURN server. See [deployment](../deployment).
 
 By default, the microphone is [configured](https://github.com/freddyaboulton/gradio-webrtc/blob/903f1f70bd586f638ad3b5a3940c7a8ec70ad1f5/backend/gradio_webrtc/webrtc.py#L575) to do echo cancellation.
 This is what's causing the recorded audio to sound muffled when the streamed audio starts playing.
-You can disable this via the `track_constraints` (see [advanced configuration](../advanced-configuration])) with the following code:
+You can disable this via the `track_constraints` (see [Advanced Configuration](../advanced-configuration)) with the following code:
 
 ```python
 stream = Stream(

docs/index.md

@@ -214,7 +214,7 @@ Learn more about the [Stream](userguide/streams) in the user guide.
 
 :simple-webstorm:{ .lg } Websocket Support - Use the `.mount(app)` method to mount the stream on a FastAPI app and get a websocket endpoint for your own frontend! 
 
-:telephone:{ .lg } Automatic Telephone Support - Us the `fastphone()` method of the stream to launch the application and get a free temporary phone number!
+:telephone:{ .lg } Automatic Telephone Support - Use the `fastphone()` method of the stream to launch the application and get a free temporary phone number!
 
 :robot:{ .lg } Completely customizable backend - A `Stream` is just a FastAPI app so you can easily extend it to fit your production application. See the [Talk To Claude](https://huggingface.co/spaces/fastrtc/talk-to-claude) demo for an example on how to serve a custom JS frontend.
 

docs/userguide/api.md

@@ -64,9 +64,9 @@ The `ReplyOnPause` handler can also send the following `log` messages.
 
 ### Additional Inputs
 
-When the `send_input` message is received, update the inputs of your handler however you like by by using the `set_input` method of the `Stream` object.
+When the `send_input` message is received, update the inputs of your handler however you like by using the `set_input` method of the `Stream` object.
 
-A common pattern in to use a `POST` request to send the updated data. The first argument to the `set_input` method is the `webrtc_id` of the handler.
+A common pattern is to use a `POST` request to send the updated data. The first argument to the `set_input` method is the `webrtc_id` of the handler.
 
 ```python
 from pydantic import BaseModel, Field
@@ -87,7 +87,7 @@ The updated data will be passed to the handler on the **next** call.
 
 The `fetch_output` message is sent to the client whenever an instance of [`AdditionalOutputs`](../streams/#additional-outputs) is available. You can access the latest output data by calling the `fetch_latest_output` method of the `Stream` object. 
 
-However, rather than fetching each output manually, a common pattern is to fetch the entire stream of output data by calling `
+However, rather than fetching each output manually, a common pattern is to fetch the entire stream of output data by calling the `output_stream` method.
 
 Here is an example:
 ```python

docs/userguide/streams.md

@@ -24,19 +24,19 @@ stream = Stream(
     additional_outputs_handler=None # (6)
 )
 
-# Optional: Add routes
-@stream.get("/") # (7)
+stream.ui.launch()
+
+# OR: Mount to a larger FastAPI app and add custom routes
+
+app = FastAPI()
+stream.mount(app)
+
+@app.get("/") # (7)
 async def _():
     return HTMLResponse(content=open("index.html").read())
 
 # launch the stream
-# uvicorn app:stream --host 0.0.0.0 --port 8000
-
-# Will print out:
-# INFO:	  Visit /webrtc/docs to test the stream and access WebRTC docs. # (8)
-# INFO:	  Visit /ui to access a sample UI for the stream.
-# INFO:	  Visit /websocket/docs for websocket docs.
-# INFO:	  Visit /telephone/docs for docs on connecting with a telephone
+# uvicorn app:app --host 0.0.0.0 --port 8000
 ```
 
 1. See [Handlers](#handlers) for more information.
@@ -71,7 +71,7 @@ The `handler` argument is the main component of the `Stream` object. A handler s
 
 | Modality | send-receive | send | receive |
 |----------|--------------|------|----------|
-| video | Function that takes a video frame and returns modified frame | Function that takes a video frame and returns modified frame | Function that takes a video frame and returns modified frame |
+| video | Function that takes a video frame and returns a new video frame | Function that takes a video frame and returns a new frame | Function that takes a video frame and returns a new frame |
 | audio | `StreamHandler` or `AsyncStreamHandler` subclass | `StreamHandler` or `AsyncStreamHandler` subclass | Generator yielding audio frames |
 | audio-video | `AudioVideoStreamHandler` or `AsyncAudioVideoStreamHandler` subclass | Not Supported Yet | Not Supported Yet |