Fix some typos

lib/ex_webrtc/peer_connection.ex

@@ -165,7 +165,7 @@ defmodule ExWebRTC.PeerConnection do
   should be used for sending.
 
   Once the first RTP packet is sent (via `send_rtp/4`), `set_sender_codec/3`
-  can only be called with a codec with the clock rate.
+  can only be called with a codec with the same clock rate.
 
   Although very unlikely, keep in mind that after renegotiation,
   the selected codec may no longer be supported by the remote side and you might
@@ -177,6 +177,10 @@ defmodule ExWebRTC.PeerConnection do
   {:ok, pc} = PeerConnection.start_link()
   {:ok, rtp_sender} = PeerConnection.add_track(MediaStreamTrack.new(:video))
 
+  # exchange SDP with the remote side
+  # {:ok, offer} = PeerConnection.create_offer(pc)
+  # ...
+
   tr =
     pc
     |> PeerConnection.get_transceivers()

lib/ex_webrtc/rtp_sender.ex

@@ -263,7 +263,7 @@ defmodule ExWebRTC.RTPSender do
   defp rtx?(codec), do: String.ends_with?(codec.mime_type, "rtx")
   defp supported?(sender, codec), do: codec in sender.codecs
 
-  # As long as report recorder is not initialized i.e. we have not send any RTP packet
+  # As long as report recorder is not initialized i.e. we have not sent any RTP packet
   # allow for codec changes. Once we start sending RTP packet, require the same clock rate.
   defp same_clock_rate?(%{report_recorder: %{clock_rate: nil}}, _codec), do: true
   defp same_clock_rate?(sender, codec), do: sender.report_recorder.clock_rate == codec.clock_rate

test/ex_webrtc/rtp_sender_test.exs

@@ -49,7 +49,7 @@ defmodule ExWebRTC.RTPSenderTest do
     %{sender: sender}
   end
 
-  test "new/6" do
+  test "new/7" do
     track = MediaStreamTrack.new(:video)
 
     codecs = [@vp8, @av1, @av1_45khz, @rtx]