Add sender controlled playout delay limits

webrtc/modules/rtp_rtcp/source/rtp_sender.h

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index f501d27a723c62f745b3120d76e7e338734e1da0..c31e7747366e556ccdd05ce5e7e39f7b3d0ea8e2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -24,6 +24,7 @@
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
+#include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
@@ -61,7 +62,12 @@ class RTPSenderInterface {
                                  bool timestamp_provided = true,
                                  bool inc_sequence_number = true) = 0;
 
-  virtual size_t RTPHeaderLength() const = 0;
+  // This returns the expected header length taking into consideration
+  // the optional RTP header extensions that may not be currently enabled.
+  virtual size_t RtpHeaderCurrentLength() const = 0;
+  // This returns the maximum RTP header length if all the header extensions
+  // were on the packet.
+  virtual size_t RtpHeaderMaxLength() const = 0;
   // Returns the next sequence number to use for a packet and allocates
   // 'packets_to_send' number of sequence numbers. It's important all allocated
   // sequence numbers are used in sequence to avoid perceived packet loss.
@@ -170,7 +176,8 @@ class RTPSender : public RTPSenderInterface {
   bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) override;
   int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
 
-  size_t RtpHeaderExtensionTotalLength() const;
+  size_t RtpHeaderExtensionCurrentLength() const;
+  size_t RtpHeaderExtensionMaxLength() const;
 
   uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer, bool marker_bit) const;
 
@@ -180,6 +187,9 @@ class RTPSender : public RTPSenderInterface {
   uint8_t BuildVideoRotationExtension(uint8_t* data_buffer) const;
   uint8_t BuildTransportSequenceNumberExtension(uint8_t* data_buffer,
                                                 uint16_t sequence_number) const;
+  uint8_t BuildPlayoutDelayExtension(uint8_t* data_buffer,
+                                     uint16_t min_playout_delay_ms,
+                                     uint16_t max_playout_delay_ms) const;
 
   // Verifies that the specified extension is registered, and that it is
   // present in rtp packet. If extension is not registered kNotRegistered is
@@ -229,6 +239,9 @@ class RTPSender : public RTPSenderInterface {
 
   bool ProcessNACKBitRate(uint32_t now);
 
+  // Feedback to decide when to stop sending playout delay.
+  void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks);
+
   // RTX.
   void SetRtxStatus(int mode);
   int RtxStatus() const;
@@ -247,7 +260,8 @@ class RTPSender : public RTPSenderInterface {
                          const bool timestamp_provided = true,
                          const bool inc_sequence_number = true) override;
 
-  size_t RTPHeaderLength() const override;
+  size_t RtpHeaderCurrentLength() const override;
+  size_t RtpHeaderMaxLength() const override;
   uint16_t AllocateSequenceNumber(uint16_t packets_to_send) override;
   size_t MaxPayloadLength() const override;
 
@@ -381,6 +395,12 @@ class RTPSender : public RTPSenderInterface {
                                      size_t rtp_packet_length,
                                      const RTPHeader& rtp_header) const;
 
+  void UpdatePlayoutDelayLimits(uint8_t* rtp_packet,
+                                size_t rtp_packet_length,
+                                const RTPHeader& rtp_header,
+                                uint16_t min_playout_delay,
+                                uint16_t max_playout_delay) const;
+
   bool AllocateTransportSequenceNumber(int* packet_id) const;
 
   void UpdateRtpStats(const uint8_t* buffer,
@@ -459,6 +479,11 @@ class RTPSender : public RTPSenderInterface {
   size_t nack_byte_count_[NACK_BYTECOUNT_SIZE];
   Bitrate nack_bitrate_;
 
+  // Tracks the current request for playout delay limits from application
+  // and decides whether the current RTP frame should include the playout
+  // delay extension on header.
+  PlayoutDelayOracle playout_delay_oracle_;
+
   RTPPacketHistory packet_history_;
 
   // Statistics