Add sender controlled playout delay limits

webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h

Index: webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h
diff --git a/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h b/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h
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index 0000000000000000000000000000000000000000..5261415fba62d69b8c67d96882d5627687273302
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+/*
+ *  Copyright (c) 2016 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_PLAYOUT_DELAY_ORACLE_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+// This class tracks the application requests to limit minimum and maximum
+// playout delay and makes a decision on whether the current RTP frame
+// should include the playout out delay extension header.
+//
+//  Playout delay can be defined in terms of capture and render time as follows:
+//
+// Render time = Capture time in receiver time + playout delay
+//
+// The application specifies a minimum and maximum limit for the playout delay
+// which are both communicated to the receiver and the receiver can adapt
+// the playout delay within this range based on observed network jitter.
+class PlayoutDelayOracle {
+ public:
+  PlayoutDelayOracle();
+  ~PlayoutDelayOracle();
+
+  // Returns true if the current frame should include the playout delay
+  // extension
+  bool send_playout_delay() const {
+    rtc::CritScope lock(&crit_sect_);
+    return send_playout_delay_;
+  }
+
+  // Returns current minimum playout delay in milliseconds.
+  int min_playout_delay_ms() const {
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return min_playout_delay_ms_;
+  }
+
+  // Returns current maximum playout delay in milliseconds.
+  int max_playout_delay_ms() const {
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return max_playout_delay_ms_;
+  }
+
+  // Updates the application requested playout delay, current ssrc
+  // and the current sequence number.
+  void UpdateRequest(uint32_t ssrc,
+                     PlayoutDelay playout_delay,
+                     uint16_t seq_num);
+
+  void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks);
+
+ private:
+  // The playout delay information is updated from the encoder thread or
+  // a thread controlled by application in case of external encoder.
+  // The sequence number feedback is updated from the worker thread.
+  // Guards access to data across the two threads.
+  rtc::CriticalSection crit_sect_;
+  // The current highest sequence number on which playout delay has been sent.
+  int64_t high_sequence_number_ GUARDED_BY(crit_sect_);
+  // Indicates whether the playout delay should go on the next frame.
+  bool send_playout_delay_ GUARDED_BY(crit_sect_);
+  // Sender ssrc.
+  uint32_t ssrc_ GUARDED_BY(crit_sect_);
+
+  // Data in this section is accessed on the sending/encoder thread alone.
+  rtc::ThreadChecker thread_checker_;
+  // Sequence number unwrapper.
+  SequenceNumberUnwrapper unwrapper_ ACCESS_ON(thread_checker_);
+  // Min playout delay value on the next frame if |send_playout_delay_| is set.
+  int min_playout_delay_ms_ ACCESS_ON(thread_checker_);
+  // Max playout delay value on the next frame if |send_playout_delay_| is set.
+  int max_playout_delay_ms_ ACCESS_ON(thread_checker_);
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(PlayoutDelayOracle);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_