Fix spelling mistakes and typos

Author: JoeTurki

README.md

@@ -5,7 +5,7 @@
 The following are a collection of example applications built by Pion users.  These applications show real world usage of Pion,
 and should serve as a good starting point for your next project. For more minimal examples check out [examples](https://github.com/pion/webrtc/tree/master/examples) in the Pion WebRTC repository
 
-If you have a request please make an issue, we also love contributions more examples are always welcome.
+If you have a request please make an issue, we also love contributions more examples, are always welcome.
 
 Have any questions? Join [the Slack channel](https://pion.ly/slack) to follow development and speak with the maintainers.
 
@@ -22,7 +22,7 @@ Have any questions? Join [the Slack channel](https://pion.ly/slack) to follow de
 * [Snapshot](snapshot) Example snapshot shows how you can convert incoming video frames to jpeg and serve them via HTTP.
 * [SIP to WebRTC](sip-to-webrtc) Example sip-to-webrtc shows how to bridge WebRTC and SIP traffic.
 * [GoCV to WebRTC](gocv-to-webrtc): Example gocv-to-webrtc captures webcam and performs motion detection in Go, it then sends results to view in the browser.
-* [Ebiten Gmae](ebiten-game): Example ebiten-game a cross-platform WebRTC game demo using [Ebitengine](https://ebitengine.org/).
+* [Ebiten Game](ebiten-game): Example ebiten-game a cross-platform WebRTC game demo using [Ebitengine](https://ebitengine.org/).
 
 ### Usage
 We've made it easy to run the browser based examples on your local machine.

ebiten-game/README.md

@@ -4,7 +4,7 @@ This is a pretty nifty demo on how to use [ebitengine](https://ebitengine.org/)
 
 You can have a client running on the browser and one running on a desktop and they can talk to each other, provided they are connected to the same signaling server
 
-Requires the signaling server to be running. To do go, just go inside the folder /signaling-server and do ``go run .``
+Requires the signaling server to be running. To do so, just go inside the folder /signaling-server and do ``go run .``
 
 you can then run the game by going in /game and doing either
 

gstreamer-send-offer/README.md

@@ -9,7 +9,7 @@ This example requires you have GStreamer installed, these are the supported plat
 #### Windows MinGW64/MSYS2
 `pacman -S mingw-w64-x86_64-gstreamer mingw-w64-x86_64-gst-libav mingw-w64-x86_64-gst-plugins-good mingw-w64-x86_64-gst-plugins-bad mingw-w64-x86_64-gst-plugins-ugly`
 #### macOS
-` brew install gst-plugins-good pkg-config && export PKG_CONFIG_PATH="/usr/local/opt/libffi/lib/pkgconfig`
+` brew install gst-plugins-good pkg-config && export PKG_CONFIG_PATH="/usr/local/opt/libffi/lib/pkgconfig"`
 
 ### Run gstreamer-send-offer and make an offer to gstreamer-receive via stdin
 ```
@@ -29,7 +29,7 @@ Congrats, you have used Pion WebRTC! Now start building something cool
 ## Customizing your video or audio
 `gstreamer-send-offer` also accepts the command line arguments `-video-src` and `-audio-src` allowing you to provide custom inputs.
 
-When prototyping with GStreamer it is highly recommended that you enable debug output, this is done by setting the `GST_DEBUG` enviroment variable.
+When prototyping with GStreamer it is highly recommended that you enable debug output, this is done by setting the `GST_DEBUG` environment variable.
 You can read about that [here](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gst-running.html) a good default value is `GST_DEBUG=*:3`
 
 You can also prototype a GStreamer pipeline by using `gst-launch-1.0` to see how things look before trying them with `gstreamer-send` for the examples below you

gstreamer-send/README.md

@@ -39,7 +39,7 @@ Congrats, you have used Pion WebRTC! Now start building something cool
 ## Customizing your video or audio
 `gstreamer-send` also accepts the command line arguments `-video-src` and `-audio-src` allowing you to provide custom inputs.
 
-When prototyping with GStreamer it is highly recommended that you enable debug output, this is done by setting the `GST_DEBUG` enviroment variable.
+When prototyping with GStreamer it is highly recommended that you enable debug output, this is done by setting the `GST_DEBUG` environment variable.
 You can read about that [here](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gst-running.html) a good default value is `GST_DEBUG=*:3`
 
 You can also prototype a GStreamer pipeline by using `gst-launch-1.0` to see how things look before trying them with `gstreamer-send` for the examples below you

sfu-ws/flutter/README.md

@@ -31,7 +31,7 @@ Add the following to the bottom of the `./ios/Podfile`:
 
 ## Android
 
-Add the following entry to your Android Manifest file, located in `./android/app/src/main/AndroidManifest.xml:
+Add the following entry to your Android Manifest file, located in `./android/app/src/main/AndroidManifest.xml`:
 
 ```xml
 <uses-feature android:name="android.hardware.camera" />
@@ -47,7 +47,7 @@ Add the following entry to your Android Manifest file, located in `./android/app
 </application>
 ```
 
-Edit`android/app/build.gradle`, modify minSdkVersion to 18
+Edit `android/app/build.gradle`, modify minSdkVersion to 18
 
 ```gradle
     defaultConfig {

sip-to-webrtc/README.md

@@ -1,5 +1,5 @@
 # sip-to-webrtc
-SIP is an example of accepting inbounding SIP traffic (Invites) and bridging it with WebRTC.
+SIP is an example of accepting inbound SIP traffic (Invites) and bridging it with WebRTC.
 This is the most common way to connect phone calls with your WebRTC application.
 This is possible because of the excellent [emiago/sipgo](https://github.com/emiago/sipgo) library.
 

snapshot/README.md

@@ -3,7 +3,7 @@ snapshot shows how you can convert incoming video frames to jpeg and serve them
 
 ## Instructions
 
-### Download snapshow
+### Download snapshot
 This example requires you to clone the repo since it is serving static HTML.
 
 ```