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 128 matching lines @@
   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;
 
-  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;
   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 20 matching lines @@
 
   bool TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms,
                         bool retransmission);
   size_t TimeToSendPadding(size_t bytes);
 
   // NACK.
   int SelectiveRetransmissions() const;
   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);
 
   bool StorePackets() const;
 
   int32_t ReSendPacket(uint16_t packet_id, int64_t min_resend_time = 0);
 
   bool ProcessNACKBitRate(uint32_t now);
 
   // RTX.
@@ skipping 141 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_