Add writing and parsing of the abs-capture-time RTP header extension.

This change adds the writing and parsing of the `abs-capture-time` RTP header extension defined at:

  http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time

We are still missing the code to:

- Negotiate the header extension.
- Collect capture time for audio and video and have the info sent with the header extension.
- Receive the header extension and use its info.

Bug: webrtc:10739
Change-Id: I75af492e994367f45a5bdc110af199900327b126
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144221
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28468}

modules/rtp_rtcp/source/rtp_packet.cc

@@ -183,6 +183,7 @@ void RtpPacket::CopyAndZeroMutableExtensions(
         break;
       }
       case RTPExtensionType::kRtpExtensionAudioLevel:
+      case RTPExtensionType::kRtpExtensionAbsoluteCaptureTime:
       case RTPExtensionType::kRtpExtensionColorSpace:
       case RTPExtensionType::kRtpExtensionFrameMarking:
       case RTPExtensionType::kRtpExtensionGenericFrameDescriptor00: