1 files changed, 144 insertions, 0 deletions
diff --git a/webrtc/video/vie_remb.cc b/webrtc/video/vie_remb.cc
new file mode 100644
index 0000000000..95c2f1e130
--- /dev/null
+++ b/webrtc/video/vie_remb.cc
@@ -0,0 +1,144 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/video/vie_remb.h"
+
+#include <assert.h>
+
+#include <algorithm>
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/trace.h"
+
+namespace webrtc {
+
+const int kRembSendIntervalMs = 200;
+
+// % threshold for if we should send a new REMB asap.
+const unsigned int kSendThresholdPercent = 97;
+
+VieRemb::VieRemb(Clock* clock)
+    : clock_(clock),
+      list_crit_(CriticalSectionWrapper::CreateCriticalSection()),
+      last_remb_time_(clock_->TimeInMilliseconds()),
+      last_send_bitrate_(0),
+      bitrate_(0) {}
+
+VieRemb::~VieRemb() {}
+
+void VieRemb::AddReceiveChannel(RtpRtcp* rtp_rtcp) {
+  assert(rtp_rtcp);
+
+  CriticalSectionScoped cs(list_crit_.get());
+  if (std::find(receive_modules_.begin(), receive_modules_.end(), rtp_rtcp) !=
+      receive_modules_.end())
+    return;
+
+  // The module probably doesn't have a remote SSRC yet, so don't add it to the
+  // map.
+  receive_modules_.push_back(rtp_rtcp);
+}
+
+void VieRemb::RemoveReceiveChannel(RtpRtcp* rtp_rtcp) {
+  assert(rtp_rtcp);
+
+  CriticalSectionScoped cs(list_crit_.get());
+  for (RtpModules::iterator it = receive_modules_.begin();
+       it != receive_modules_.end(); ++it) {
+    if ((*it) == rtp_rtcp) {
+      receive_modules_.erase(it);
+      break;
+    }
+  }
+}
+
+void VieRemb::AddRembSender(RtpRtcp* rtp_rtcp) {
+  assert(rtp_rtcp);
+
+  CriticalSectionScoped cs(list_crit_.get());
+
+  // Verify this module hasn't been added earlier.
+  if (std::find(rtcp_sender_.begin(), rtcp_sender_.end(), rtp_rtcp) !=
+      rtcp_sender_.end())
+    return;
+  rtcp_sender_.push_back(rtp_rtcp);
+}
+
+void VieRemb::RemoveRembSender(RtpRtcp* rtp_rtcp) {
+  assert(rtp_rtcp);
+
+  CriticalSectionScoped cs(list_crit_.get());
+  for (RtpModules::iterator it = rtcp_sender_.begin();
+       it != rtcp_sender_.end(); ++it) {
+    if ((*it) == rtp_rtcp) {
+      rtcp_sender_.erase(it);
+      return;
+    }
+  }
+}
+
+bool VieRemb::InUse() const {
+  CriticalSectionScoped cs(list_crit_.get());
+  if (receive_modules_.empty() && rtcp_sender_.empty())
+    return false;
+  else
+    return true;
+}
+
+void VieRemb::OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
+                                      unsigned int bitrate) {
+  list_crit_->Enter();
+  // If we already have an estimate, check if the new total estimate is below
+  // kSendThresholdPercent of the previous estimate.
+  if (last_send_bitrate_ > 0) {
+    unsigned int new_remb_bitrate = last_send_bitrate_ - bitrate_ + bitrate;
+
+    if (new_remb_bitrate < kSendThresholdPercent * last_send_bitrate_ / 100) {
+      // The new bitrate estimate is less than kSendThresholdPercent % of the
+      // last report. Send a REMB asap.
+      last_remb_time_ = clock_->TimeInMilliseconds() - kRembSendIntervalMs;
+    }
+  }
+  bitrate_ = bitrate;
+
+  // Calculate total receive bitrate estimate.
+  int64_t now = clock_->TimeInMilliseconds();
+
+  if (now - last_remb_time_ < kRembSendIntervalMs) {
+    list_crit_->Leave();
+    return;
+  }
+  last_remb_time_ = now;
+
+  if (ssrcs.empty() || receive_modules_.empty()) {
+    list_crit_->Leave();
+    return;
+  }
+
+  // Send a REMB packet.
+  RtpRtcp* sender = NULL;
+  if (!rtcp_sender_.empty()) {
+    sender = rtcp_sender_.front();
+  } else {
+    sender = receive_modules_.front();
+  }
+  last_send_bitrate_ = bitrate_;
+
+  list_crit_->Leave();
+
+  if (sender) {
+    sender->SetREMBData(bitrate_, ssrcs);
+  }
+}
+
+}  // namespace webrtc