Add sender controlled playout delay limits

webrtc/modules/rtp_rtcp/source/playout_delay_oracle.cc

+/*
+ *  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.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+PlayoutDelayOracle::PlayoutDelayOracle() {}
+PlayoutDelayOracle::~PlayoutDelayOracle() {}
+
+bool PlayoutDelayOracle::ShouldIncludePlayoutDelayExtension(int ssrc) const {
+  bool ret =
+      send_playout_delay_ssrc_.find(ssrc) != send_playout_delay_ssrc_.end() &&
+      send_playout_delay_ssrc_.at(ssrc);

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

Store iterator and dereference if != end(), rather than doing the lookup twice.

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

I removed the map

+  return ret;
+}
+
+int PlayoutDelayOracle::MinPlayoutDelayMs(int ssrc) const {
+  return ssrc_to_min_playout_delay_.at(ssrc);
+}
+
+int PlayoutDelayOracle::MaxPlayoutDelayMs(int ssrc) const {
+  return ssrc_to_max_playout_delay_.at(ssrc);
+}
+
+void PlayoutDelayOracle::Update(int ssrc,
+                                int min_playout_delay_ms,
+                                int max_playout_delay_ms,
+                                int seq_num) {
+  RTC_DCHECK_LE(min_playout_delay_ms, kPlayoutDelayMaxMs);
+  RTC_DCHECK_LE(max_playout_delay_ms, kPlayoutDelayMaxMs);
+  if (min_playout_delay_ms >= 0 &&
+      min_playout_delay_ms != ssrc_to_min_playout_delay_[ssrc]) {
+    send_playout_delay_ssrc_[ssrc] = true;
+    ssrc_to_min_playout_delay_[ssrc] = min_playout_delay_ms;
+    ssrc_to_high_seq_num_[ssrc] = seq_num;
+  }
+
+  if (max_playout_delay_ms >= 0 &&
+      max_playout_delay_ms != ssrc_to_max_playout_delay_[ssrc]) {
+    send_playout_delay_ssrc_[ssrc] = true;
+    ssrc_to_max_playout_delay_[ssrc] = max_playout_delay_ms;
+    ssrc_to_high_seq_num_[ssrc] = seq_num;
+  }
+}
+
+// If an ACK is received on the packet containing the playout delay extension,
+// we stop sending the extension on future packets.
+void PlayoutDelayOracle::OnReceivedRtcpReceiverReport(
+    const ReportBlockList& report_blocks) {
+  for (const RTCPReportBlock& report_block : report_blocks) {
+    auto found = ssrc_to_high_seq_num_.find(report_block.sourceSSRC);
+    if (found != ssrc_to_high_seq_num_.end()) {
+      if (send_playout_delay_ssrc_[report_block.sourceSSRC] &&
+          report_block.extendedHighSeqNum >
+              ssrc_to_high_seq_num_[report_block.sourceSSRC]) {
+        send_playout_delay_ssrc_[report_block.sourceSSRC] = false;
+      }
+    }
+  }
+}
+
+}  // namespace webrtc