Add TrySendPacket() method to RTP modules.

This method will be called when PacedSender is using the new code path
that directly owns the packets to be sent.

It can be seen as combining a few features of the old code path:
* It checks if this is the correct RTP module and then sends, without
  the need for PacketRouter to poll multiple methods for SSRC etc first.
* It partly corresponds to TimeToSendPacket(), but RTX encapsulation
  now happens pre-pacer and FEC does not need to have a packet history,
  so most of that method is not used.
* It implements most of PrepareAndSendPacket(), such as updating header
  extensions, reporting stats and of course forwards to Transport. It
  now also handles the history a bit differently, since media packets
  will only be stored for potential retransmission post-pacer.

Bug: webrtc:10633
Change-Id: Ie97952eeef6e56e462e115d67f7c7929f36c1817
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142165
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28298}

modules/rtp_rtcp/include/rtp_rtcp.h

@@ -268,6 +268,12 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
       bool retransmission,
       const PacedPacketInfo& pacing_info) = 0;
 
+  // Try to send the provided packet. Returns true iff packet matches any of
+  // the SSRCs for this module (media/rtx/fec etc) and was forwarded to the
+  // transport.
+  virtual bool TrySendPacket(RtpPacketToSend* packet,
+                             const PacedPacketInfo& pacing_info) = 0;
+
   virtual size_t TimeToSendPadding(size_t bytes,
                                    const PacedPacketInfo& pacing_info) = 0;