docs fix

ahmadalwa · Jul 18, 2020 · f56dd36 · f56dd36
1 parent 8c89162
commit f56dd36
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Showing 9 changed files with 13 additions and 65 deletions.
diff --git a/.readthedocs.yml b/.readthedocs.yml
@@ -7,7 +7,7 @@ version: 2
 
 # Build documentation in the docs/ directory with Sphinx
 sphinx:
-  configuration: docs/source/conf.py
+  configuration: docs/conf.py
 
 
 # Optionally set the version of Python and requirements required to build your docs

diff --git a/docs/source/Examples.rst → docs/Examples.rst b/docs/source/Examples.rst → docs/Examples.rst
diff --git a/docs/Makefile b/docs/Makefile
diff --git a/docs/source/RTP.rst → docs/RTP.rst b/docs/source/RTP.rst → docs/RTP.rst
@@ -109,7 +109,7 @@ RTP.\ **RTPPacketManager**\ ()
     This rebuilds the ByteIO if packets are sent out of order.  Setting the argument *reset* to true will wipe all data in the ByteIO and insert in the data in the argument *data* at the position in the argument *offset*.
 
   **write**\ (offset, data)
-    Writes the data in the argument *data* to the ByteIO at the position in the argument *offset*.  RTP data comes with a timestamp that is passed as the offset in this case.  This makes it so a whole left by delayed packets can be filled later.  If a packet with a timestamp sooner than any other timestamp received, it will rebuild the ByteIO with the new data.  If this new position is over 100,000 bytes before the earliest byte, the ByteIO is completely wiped and starts over.  This is to prevent Overflow errors.
+    Writes the data in the argument *data* to the ByteIO at the position in the argument *offset*.  RTP data comes with a timestamp that is passed as the offset in this case.  This makes it so a hole left by delayed packets can be filled later.  If a packet with a timestamp sooner than any other timestamp received, it will rebuild the ByteIO with the new data.  If this new position is over 100,000 bytes before the earliest byte, the ByteIO is completely wiped and starts over.  This is to prevent Overflow errors.
 
 .. _RTPMessage:
 
@@ -163,7 +163,7 @@ RTP.\ **RTPMessage**\ (data, assoc)
     This method returns a string representation of the RTP packet excluding the payload.
 
   **parse**\ (data)
-    This method is called by the initialization of the class.  It determins the RTP version, whether the packet has padding, has a header extension, and other information about the backet.
+    This method is called by the initialization of the class.  It determines the RTP version, whether the packet has padding, has a header extension, and other information about the backet.
 
 .. _RTPClient:
 
@@ -178,11 +178,11 @@ The RTPClient is used to send and receive RTP packets and encode/decode the audi
 
     The *inIP* argument is used to receive incoming RTP message.
 
-    The *inPort* argument is the port RTPClient will bind to to receive incoming RTP messages.
+    The *inPort* argument is the port RTPClient will bind to, to receive incoming RTP packets.
 
-    The *outIP* argument is used to transmit RTP messages.
+    The *outIP* argument is used to transmit RTP packets.
 
-    The *outPort* argument is used to transmit RTP messages.
+    The *outPort* argument is used to transmit RTP packets.
 
     The *sendrecv* argument describes how the RTPClient should act.  Please reference `TransmitType<enums>` for more details.
 
@@ -204,7 +204,7 @@ The RTPClient is used to send and receive RTP packets and encode/decode the audi
     This method is called by RTPClient.start() and is responsible for transmitting RTP packets. **This should not be called by the** :term:`user`.
 
   **parsePacket**\ (packet)
-    This method is called by the recv() thread.  It convertes the argument *packet* into a :ref:`RTPMessage`, then sends it to the proper parse function depending on the :ref:`PayloadType<payload-type>`.
+    This method is called by the recv() thread.  It converts the argument *packet* into a :ref:`RTPMessage`, then sends it to the proper parse function depending on the :ref:`PayloadType<payload-type>`.
 
   **encodePacket**\ (payload)
     This method is called by the trans() thread.  It encoded the argument *payload* into the prefered codec.  Currently, PCMU is the hardcoded prefered codec.  The trans() thread will use the payload to create the RTP packet before transmitting.

diff --git a/docs/source/SIP.rst → docs/SIP.rst b/docs/source/SIP.rst → docs/SIP.rst
diff --git a/docs/source/VoIP.rst → docs/VoIP.rst b/docs/source/VoIP.rst → docs/VoIP.rst
diff --git a/docs/source/conf.py → docs/conf.py b/docs/source/conf.py → docs/conf.py
@@ -1,3 +1,5 @@
+import sphinx_rtd_theme
+
 # Configuration file for the Sphinx documentation builder.
 #
 # This file only contains a selection of the most common options. For a full
@@ -24,14 +26,15 @@
 # The full version, including alpha/beta/rc tags
 release = '0.5.0 Beta'
 
-master_doc = 'index'
+master_doc = 'index.html'
 
 # -- General configuration ---------------------------------------------------
 
 # Add any Sphinx extension module names here, as strings. They can be
 # extensions coming with Sphinx (named 'sphinx.ext.*') or your custom
 # ones.
-extensions = [
+extensions = [ 
+  "sphinx_rtd_theme"
 ]
 
 # Add any paths that contain templates here, relative to this directory.
@@ -48,7 +51,7 @@
 # The theme to use for HTML and HTML Help pages.  See the documentation for
 # a list of builtin themes.
 #
-html_theme = 'alabaster'
+html_theme = 'sphinx_rtd_theme'
 
 #pygments_style = 'sphinx'
 

diff --git a/docs/source/index.rst → docs/index.rst b/docs/source/index.rst → docs/index.rst
diff --git a/docs/make.bat b/docs/make.bat