Rename index.md to README.md to make it automatically show up

Also add a heading to each file.

No-Try: True
Bug: webrtc:11335
Change-Id: I5e935741662558e72e417fa80a48c5ecda66c5f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169854
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30702}

docs/native-code/rtp-hdrext/abs-capture-time/README.md

@@ -1,3 +1,5 @@
+# Absolute Capture Time
+
 The Absolute Capture Time extension is used to stamp RTP packets with a NTP
 timestamp showing when the first audio or video frame in a packet was originally
 captured. The intent of this extension is to provide a way to accomplish

docs/native-code/rtp-hdrext/abs-send-time/README.md

@@ -1,3 +1,5 @@
+# Absolute Send Time
+
 The Absolute Send Time extension is used to stamp RTP packets with a timestamp
 showing the departure time from the system that put this packet on the wire
 (or as close to this as we can manage). Contact <solenberg@google.com> for

docs/native-code/rtp-hdrext/color-space/README.md

@@ -1,3 +1,5 @@
+# Color Space
+
 The color space extension is used to communicate color space information and
 optionally also metadata that is needed in order to properly render a high
 dynamic range (HDR) video stream. Contact <kron@google.com> for more info.

docs/native-code/rtp-hdrext/inband-cn/README.md

@@ -1,3 +1,5 @@
+# Inband Comfort Noise
+
 **Name:** "Inband Comfort Noise" ; "RTP Header Extension to signal inband comfort noise"
 
 **Formal name:** <http://www.webrtc.org/experiments/rtp-hdrext/inband-cn>

docs/native-code/rtp-hdrext/index.md

@@ -1,10 +1,12 @@
+# Experimental RTP header extensions
+
 The following subpages define experiemental RTP header extensions:
 
-  * [abs-send-time](abs-send-time/index.md)
+  * [abs-send-time](abs-send-time/README.md)
-  * [abs-capture-time](abs-capture-time/index.md)
+  * [abs-capture-time](abs-capture-time/README.md)
-  * [color-space](color-space/index.md)
+  * [color-space](color-space/README.md)
-  * [playout-delay](playout-delay/index.md)
+  * [playout-delay](playout-delay/README.md)
-  * [transport-wide-cc-02](transport-wide-cc-02/index.md)
+  * [transport-wide-cc-02](transport-wide-cc-02/README.md)
-  * [video-content-type](video-content-type/index.md)
+  * [video-content-type](video-content-type/README.md)
-  * [video-timing](video-timing/index.md)
+  * [video-timing](video-timing/README.md)
-  * [inband-cn](inband-cn/index.md)
+  * [inband-cn](inband-cn/README.md)

docs/native-code/rtp-hdrext/playout-delay/README.md

@@ -1,3 +1,5 @@
+# Playout Delay
+
 **Name:** "Playout Delay" ; "RTP Header Extension to control Playout Delay"
 
 **Formal name:** <http://www.webrtc.org/experiments/rtp-hdrext/playout-delay>

docs/native-code/rtp-hdrext/transport-wide-cc-02/README.md

@@ -1,3 +1,5 @@
+# Transport-Wide Congestion Control
+
 This RTP header extension is an extended version of the extension defined in
 <https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01>
 

docs/native-code/rtp-hdrext/video-content-type/README.md

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+# Video Content Type
+
 The Video Content Type extension is used to communicate a video content type
 from sender to receiver of rtp video stream. Contact <ilnik@google.com> for
 more info.

docs/native-code/rtp-hdrext/video-timing/README.md

@@ -1,3 +1,5 @@
+# Video Timing
+
 The Video Timing extension is used to communicate a timing information on
 per-frame basis to receiver of rtp video stream. Contact <ilnik@google.com> for
 more info. It may be generalized to audio frames as well in the future.