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
new file mode 100644
index 0000000000000000000000000000000000000000..a3d1d308d786985d7c26e5d60c6bb1600ae8e919
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h
@@ -0,0 +1,66 @@
+/*
+ *  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 <stdint.h>

[danilchap, 2016/05/24 13:28:24]

#include "webrtc/base/basictypes.h" instead of <stdint.h> directly

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

Done.

+
+#include <unordered_map>
+
+#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 ShouldIncludePlayoutDelayExtension(int ssrc) const;

[danilchap, 2016/05/24 13:28:24]

We use uint32_t for ssrcs (it is always 32bits)

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

Done.

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

Done.

+
+  // Returns current minimum playout delay in milliseconds.
+  int MinPlayoutDelayMs(int ssrc) const;
+
+  // Returns current maximum playout delay in milliseconds.
+  int MaxPlayoutDelayMs(int ssrc) const;
+
+  void Update(int ssrc,
+              int min_playout_delay_ms,
+              int max_playout_delay_ms,
+              int seq_num);
+
+  void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks);
+
+ private:
+  std::unordered_map<uint32_t, uint32_t> ssrc_to_high_seq_num_;

[danilchap, 2016/05/24 13:28:24]

PlayoutDelayOracle lives inside RtpSender that is responsible for single remote
point and has two ssrcs: one for normal packets, another - for rtx channel.
It feels like you do not need maps or ssrc in this class. rtcp receiver should
filter out unrelated ssrc when assembling receiver report list.

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

+1 to not have a map here if not necessary. Otherwise please have map from ssrc
to a struct of the context you need, rather than having four maps keyed the same
way.

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

Thanks for pointing this out Danil. Makes it much simpler.

+  std::unordered_map<uint32_t, bool> send_playout_delay_ssrc_;
+  std::unordered_map<uint32_t, int> ssrc_to_min_playout_delay_;
+  std::unordered_map<uint32_t, int> ssrc_to_max_playout_delay_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(PlayoutDelayOracle);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_