Add sender controlled playout delay limits

webrtc/modules/rtp_rtcp/source/rtp_sender.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
 
 #include <list>
 #include <map>
 #include <memory>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/random.h"
 #include "webrtc/base/thread_annotations.h"
 #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"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
 #include "webrtc/transport.h"
 
 namespace webrtc {
 
 class RTPSenderAudio;
@@ skipping 17 matching lines @@
   virtual uint32_t Timestamp() const = 0;
 
   virtual int32_t BuildRTPheader(uint8_t* data_buffer,
                                  int8_t payload_type,
                                  bool marker_bit,
                                  uint32_t capture_timestamp,
                                  int64_t capture_time_ms,
                                  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.
   virtual uint16_t AllocateSequenceNumber(uint16_t packets_to_send) = 0;
   virtual uint16_t SequenceNumber() const = 0;
   virtual size_t MaxPayloadLength() const = 0;
   virtual size_t MaxDataPayloadLength() const = 0;
   virtual uint16_t ActualSendBitrateKbit() const = 0;
 
   virtual int32_t SendToNetwork(uint8_t* data_buffer,
@@ skipping 88 matching lines @@
   // RTP header extension
   int32_t SetTransmissionTimeOffset(int32_t transmission_time_offset);
   int32_t SetAbsoluteSendTime(uint32_t absolute_send_time);
   void SetVideoRotation(VideoRotation rotation);
   int32_t SetTransportSequenceNumber(uint16_t sequence_number);
 
   int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
   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;
 
   uint8_t BuildTransmissionTimeOffsetExtension(uint8_t *data_buffer) const;
   uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const;
   uint8_t BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const;
   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
   // returned. If extension cannot be found in the rtp header, or if it is
   // malformed, kError is returned. Otherwise *extension_offset is set to the
   // offset of the extension from the beginning of the rtp packet and kOk is
   // returned.
   enum class ExtensionStatus {
     kNotRegistered,
     kOk,
@@ skipping 29 matching lines @@
                       int64_t avg_rtt);
 
   void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
 
   bool StorePackets() const;
 
   int32_t ReSendPacket(uint16_t packet_id, int64_t min_resend_time = 0);
 
   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;
 
   uint32_t RtxSsrc() const;
   void SetRtxSsrc(uint32_t ssrc);
 
   void SetRtxPayloadType(int payload_type, int associated_payload_type);
 
   // Functions wrapping RTPSenderInterface.
   int32_t BuildRTPheader(uint8_t* data_buffer,
                          int8_t payload_type,
                          bool marker_bit,
                          uint32_t capture_timestamp,
                          int64_t capture_time_ms,
                          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;
 
   // Current timestamp.
   uint32_t Timestamp() const override;
   uint32_t SSRC() const override;
 
   int32_t SendToNetwork(uint8_t* data_buffer,
                         size_t payload_length,
                         size_t rtp_header_length,
@@ skipping 113 matching lines @@
   void UpdateAbsoluteSendTime(uint8_t* rtp_packet,
                               size_t rtp_packet_length,
                               const RTPHeader& rtp_header,
                               int64_t now_ms) const;
 
   bool UpdateTransportSequenceNumber(uint16_t sequence_number,
                                      uint8_t* rtp_packet,
                                      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,
                       size_t packet_length,
                       const RTPHeader& header,
                       bool is_rtx,
                       bool is_retransmit);
   bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const;
 
   class BitrateAggregator {
@@ skipping 58 matching lines @@
   uint32_t absolute_send_time_;
   VideoRotation rotation_;
   CVOMode cvo_mode_;
   uint16_t transport_sequence_number_;
 
   // NACK
   uint32_t nack_byte_count_times_[NACK_BYTECOUNT_SIZE];
   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
   rtc::CriticalSection statistics_crit_;
   SendDelayMap send_delays_ GUARDED_BY(statistics_crit_);
   FrameCounts frame_counts_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
   FrameCountObserver* const frame_count_observer_;
@@ skipping 28 matching lines @@
   // that the target bitrate is still valid.
   rtc::CriticalSection target_bitrate_critsect_;
   uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_