Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 91 insertions, 0 deletions

diff --git a/third_party/libwebrtc/call/call_config.h b/third_party/libwebrtc/call/call_config.h
new file mode 100644
index 0000000000..918c077435
--- /dev/null
+++ b/third_party/libwebrtc/call/call_config.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2018 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 CALL_CALL_CONFIG_H_
+#define CALL_CALL_CONFIG_H_
+
+#include "api/fec_controller.h"
+#include "api/field_trials_view.h"
+#include "api/metronome/metronome.h"
+#include "api/neteq/neteq_factory.h"
+#include "api/network_state_predictor.h"
+#include "api/rtc_error.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/transport/bitrate_settings.h"
+#include "api/transport/network_control.h"
+#include "call/audio_state.h"
+#include "call/rtp_transport_config.h"
+#include "call/rtp_transport_controller_send_factory_interface.h"
+
+namespace webrtc {
+
+class AudioProcessing;
+class RtcEventLog;
+
+struct CallConfig {
+  // If `network_task_queue` is set to nullptr, Call will assume that network
+  // related callbacks will be made on the same TQ as the Call instance was
+  // constructed on.
+  explicit CallConfig(RtcEventLog* event_log,
+                      TaskQueueBase* network_task_queue = nullptr);
+  CallConfig(const CallConfig&);
+  RtpTransportConfig ExtractTransportConfig() const;
+  ~CallConfig();
+
+  // Bitrate config used until valid bitrate estimates are calculated. Also
+  // used to cap total bitrate used. This comes from the remote connection.
+  BitrateConstraints bitrate_config;
+
+  // AudioState which is possibly shared between multiple calls.
+  rtc::scoped_refptr<AudioState> audio_state;
+
+  // Audio Processing Module to be used in this call.
+  AudioProcessing* audio_processing = nullptr;
+
+  // RtcEventLog to use for this call. Required.
+  // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
+  RtcEventLog* const event_log = nullptr;
+
+  // FecController to use for this call.
+  FecControllerFactoryInterface* fec_controller_factory = nullptr;
+
+  // Task Queue Factory to be used in this call. Required.
+  TaskQueueFactory* task_queue_factory = nullptr;
+
+  // NetworkStatePredictor to use for this call.
+  NetworkStatePredictorFactoryInterface* network_state_predictor_factory =
+      nullptr;
+
+  // Network controller factory to use for this call.
+  NetworkControllerFactoryInterface* network_controller_factory = nullptr;
+
+  // NetEq factory to use for this call.
+  NetEqFactory* neteq_factory = nullptr;
+
+  // Key-value mapping of internal configurations to apply,
+  // e.g. field trials.
+  const FieldTrialsView* trials = nullptr;
+
+  TaskQueueBase* const network_task_queue_ = nullptr;
+  // RtpTransportControllerSend to use for this call.
+  RtpTransportControllerSendFactoryInterface*
+      rtp_transport_controller_send_factory = nullptr;
+
+  Metronome* metronome = nullptr;
+
+  // The burst interval of the pacer, see TaskQueuePacedSender constructor.
+  absl::optional<TimeDelta> pacer_burst_interval;
+
+  // Enables send packet batching from the egress RTP sender.
+  bool enable_send_packet_batching = false;
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_CALL_CONFIG_H_