Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 130 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc b/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc
new file mode 100644
index 0000000000..e042678897
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc
@@ -0,0 +1,130 @@
+/*
+ *  Copyright (c) 2017 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 "modules/congestion_controller/include/receive_side_congestion_controller.h"
+
+#include "api/media_types.h"
+#include "api/units/data_rate.h"
+#include "modules/pacing/packet_router.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
+#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+static const uint32_t kTimeOffsetSwitchThreshold = 30;
+}  // namespace
+
+void ReceiveSideCongestionController::OnRttUpdate(int64_t avg_rtt_ms,
+                                                  int64_t max_rtt_ms) {
+  MutexLock lock(&mutex_);
+  rbe_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
+}
+
+void ReceiveSideCongestionController::RemoveStream(uint32_t ssrc) {
+  MutexLock lock(&mutex_);
+  rbe_->RemoveStream(ssrc);
+}
+
+DataRate ReceiveSideCongestionController::LatestReceiveSideEstimate() const {
+  MutexLock lock(&mutex_);
+  return rbe_->LatestEstimate();
+}
+
+void ReceiveSideCongestionController::PickEstimator(
+    bool has_absolute_send_time) {
+  if (has_absolute_send_time) {
+    // If we see AST in header, switch RBE strategy immediately.
+    if (!using_absolute_send_time_) {
+      RTC_LOG(LS_INFO)
+          << "WrappingBitrateEstimator: Switching to absolute send time RBE.";
+      using_absolute_send_time_ = true;
+      rbe_ = std::make_unique<RemoteBitrateEstimatorAbsSendTime>(
+          &remb_throttler_, &clock_);
+    }
+    packets_since_absolute_send_time_ = 0;
+  } else {
+    // When we don't see AST, wait for a few packets before going back to TOF.
+    if (using_absolute_send_time_) {
+      ++packets_since_absolute_send_time_;
+      if (packets_since_absolute_send_time_ >= kTimeOffsetSwitchThreshold) {
+        RTC_LOG(LS_INFO)
+            << "WrappingBitrateEstimator: Switching to transmission "
+               "time offset RBE.";
+        using_absolute_send_time_ = false;
+        rbe_ = std::make_unique<RemoteBitrateEstimatorSingleStream>(
+            &remb_throttler_, &clock_);
+      }
+    }
+  }
+}
+
+ReceiveSideCongestionController::ReceiveSideCongestionController(
+    Clock* clock,
+    RemoteEstimatorProxy::TransportFeedbackSender feedback_sender,
+    RembThrottler::RembSender remb_sender,
+    NetworkStateEstimator* network_state_estimator)
+    : clock_(*clock),
+      remb_throttler_(std::move(remb_sender), clock),
+      remote_estimator_proxy_(std::move(feedback_sender),
+                              network_state_estimator),
+      rbe_(new RemoteBitrateEstimatorSingleStream(&remb_throttler_, clock)),
+      using_absolute_send_time_(false),
+      packets_since_absolute_send_time_(0) {}
+
+void ReceiveSideCongestionController::OnReceivedPacket(
+    const RtpPacketReceived& packet,
+    MediaType media_type) {
+  bool has_transport_sequence_number =
+      packet.HasExtension<TransportSequenceNumber>() ||
+      packet.HasExtension<TransportSequenceNumberV2>();
+  if (media_type == MediaType::AUDIO && !has_transport_sequence_number) {
+    // For audio, we only support send side BWE.
+    return;
+  }
+
+  if (has_transport_sequence_number) {
+    // Send-side BWE.
+    remote_estimator_proxy_.IncomingPacket(packet);
+  } else {
+    // Receive-side BWE.
+    MutexLock lock(&mutex_);
+    PickEstimator(packet.HasExtension<AbsoluteSendTime>());
+    rbe_->IncomingPacket(packet);
+  }
+}
+
+void ReceiveSideCongestionController::OnBitrateChanged(int bitrate_bps) {
+  remote_estimator_proxy_.OnBitrateChanged(bitrate_bps);
+}
+
+TimeDelta ReceiveSideCongestionController::MaybeProcess() {
+  Timestamp now = clock_.CurrentTime();
+  mutex_.Lock();
+  TimeDelta time_until_rbe = rbe_->Process();
+  mutex_.Unlock();
+  TimeDelta time_until_rep = remote_estimator_proxy_.Process(now);
+  TimeDelta time_until = std::min(time_until_rbe, time_until_rep);
+  return std::max(time_until, TimeDelta::Zero());
+}
+
+void ReceiveSideCongestionController::SetMaxDesiredReceiveBitrate(
+    DataRate bitrate) {
+  remb_throttler_.SetMaxDesiredReceiveBitrate(bitrate);
+}
+
+void ReceiveSideCongestionController::SetTransportOverhead(
+    DataSize overhead_per_packet) {
+  remote_estimator_proxy_.SetTransportOverhead(overhead_per_packet);
+}
+
+}  // namespace webrtc