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..b779f6cc1e02f46404fdf6e0ff3c4e0b3f8a52e4 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"
@@ -172,7 +173,9 @@ class RTPSender : public RTPSenderInterface {
 
   size_t RtpHeaderExtensionTotalLength() const;
 
-  uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer, bool marker_bit) const;
+  uint16_t BuildRTPHeaderExtension(uint32_t ssrc,
+                                   uint8_t* data_buffer,
+                                   bool marker_bit) const;

[sprang_webrtc, 2016/05/24 14:46:08]

RtpSender should already know which ssrc is used?

[Irfan, 2016/05/25 09:32:53]

Removed

 
   uint8_t BuildTransmissionTimeOffsetExtension(uint8_t *data_buffer) const;
   uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const;
@@ -180,6 +183,9 @@ 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
@@ -220,6 +226,7 @@ class RTPSender : public RTPSenderInterface {
   int SetSelectiveRetransmissions(uint8_t settings);
   void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers,
                       int64_t avg_rtt);
+  void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks);
 
   void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
 
@@ -381,6 +388,12 @@ 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 +472,11 @@ 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