Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery

The PlayoutDelayOracle was responsible for making sure the PlayoutDelay
header extension was successfully propagated to the receiving side. Once
it was determined that the receiver had received a frame with the new
delay tag, it's no longer necessary to propagate.

The issue with this implementation is that it is based on max
extended sequence number reported via RTCP, which makes it often slow
to react, could theoretically fail to produce desired outcome (max
received > X does not guarantee X was fully received and decoded), and
added a lot of code complexity.

The guarantee of delivery can in fact be accomplished more reliably and
with less code by making sure to tag each frame until an undiscardable
frame is sent.

This allows containing the logic fully within RTPSenderVideo.

Bug: webrtc:11340
Change-Id: I2d1d2b6b67f4f07b8b33336f8fcfcde724243eef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168221
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30473}

modules/rtp_rtcp/include/rtp_rtcp_defines.h

@@ -392,19 +392,6 @@ struct RtpReceiveStats {
   RtpPacketCounter packet_counter;
 };
 
-class RtcpAckObserver {
- public:
-  // This method is called on received report blocks matching the sender ssrc.
-  // TODO(nisse): Use of "extended" sequence number is a bit brittle, since the
-  // observer for this callback typically has its own sequence number unwrapper,
-  // and there's no guarantee that they are in sync. Change to pass raw sequence
-  // number, possibly augmented with timestamp (if available) to aid
-  // disambiguation.
-  virtual void OnReceivedAck(int64_t extended_highest_sequence_number) = 0;
-
-  virtual ~RtcpAckObserver() = default;
-};
-
 // Callback, used to notify an observer whenever new rates have been estimated.
 class BitrateStatisticsObserver {
  public: