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 070c282d98a13bec5940e8a96d81d7d4225e0e3a..ffbcb817e709cb569da1ec09c0e5d43adfd14735 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"
@@ -42,14 +43,6 @@ class RTPSenderInterface {
   RTPSenderInterface() {}
   virtual ~RTPSenderInterface() {}
 
-  enum CVOMode {
-    kCVONone,
-    kCVOInactive,  // CVO rtp header extension is registered but haven't
-                   // received any frame with rotation pending.
-    kCVOActivated,  // CVO rtp header extension will be present in the rtp
-                    // packets.
-  };
-
   virtual uint32_t SSRC() const = 0;
   virtual uint32_t Timestamp() const = 0;
 
@@ -61,7 +54,9 @@ 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 active.
+  virtual size_t RtpHeaderLength() 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.
@@ -83,7 +78,7 @@ class RTPSenderInterface {
                                    const RTPHeader& rtp_header,
                                    VideoRotation rotation) const = 0;
   virtual bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) = 0;
-  virtual CVOMode ActivateCVORtpHeaderExtension() = 0;
+  virtual bool ActivateCVORtpHeaderExtension() = 0;
 };
 
 class RTPSender : public RTPSenderInterface {
@@ -170,7 +165,7 @@ class RTPSender : public RTPSenderInterface {
   bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) override;
   int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
 
-  size_t RtpHeaderExtensionTotalLength() const;
+  size_t RtpHeaderExtensionLength() const;
 
   uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer, bool marker_bit) const;
 
@@ -180,6 +175,9 @@ 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
@@ -231,6 +229,9 @@ class RTPSender : public RTPSenderInterface {
 
   bool ProcessNACKBitRate(uint32_t now);
 
+  // Feedback to decide when to stop sending playout delay.
+  void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks);
+
   // RTX.
   void SetRtxStatus(int mode);
   int RtxStatus() const;
@@ -249,7 +250,7 @@ class RTPSender : public RTPSenderInterface {
                          const bool timestamp_provided = true,
                          const bool inc_sequence_number = true) override;
 
-  size_t RTPHeaderLength() const override;
+  size_t RtpHeaderLength() const override;
   uint16_t AllocateSequenceNumber(uint16_t packets_to_send) override;
   size_t MaxPayloadLength() const override;
 
@@ -320,7 +321,7 @@ class RTPSender : public RTPSenderInterface {
   RtpState GetRtpState() const;
   void SetRtxRtpState(const RtpState& rtp_state);
   RtpState GetRtxRtpState() const;
-  CVOMode ActivateCVORtpHeaderExtension() override;
+  bool ActivateCVORtpHeaderExtension() override;
 
  protected:
   int32_t CheckPayloadType(int8_t payload_type, RtpVideoCodecTypes* video_type);
@@ -390,6 +391,12 @@ 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,
@@ -460,7 +467,7 @@ class RTPSender : public RTPSenderInterface {
   int32_t transmission_time_offset_;
   uint32_t absolute_send_time_;
   VideoRotation rotation_;
-  CVOMode cvo_mode_;
+  bool video_rotation_active_;
   uint16_t transport_sequence_number_;
 
   // NACK
@@ -468,6 +475,12 @@ 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_;
+  bool playout_delay_active_ GUARDED_BY(send_critsect_);
+
   RTPPacketHistory packet_history_;
 
   // Statistics