Adding upstream version 115.7.0esr.upstream/115.7.0esr

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

1 files changed, 147 insertions, 0 deletions

diff --git a/third_party/libwebrtc/call/call.h b/third_party/libwebrtc/call/call.h
new file mode 100644
index 0000000000..42daa95a6c
--- /dev/null
+++ b/third_party/libwebrtc/call/call.h
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2013 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_H_
+#define CALL_CALL_H_
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/adaptation/resource.h"
+#include "api/media_types.h"
+#include "api/task_queue/task_queue_base.h"
+#include "call/audio_receive_stream.h"
+#include "call/audio_send_stream.h"
+#include "call/call_basic_stats.h"
+#include "call/call_config.h"
+#include "call/flexfec_receive_stream.h"
+#include "call/packet_receiver.h"
+#include "call/rtp_transport_controller_send_interface.h"
+#include "call/video_receive_stream.h"
+#include "call/video_send_stream.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/network/sent_packet.h"
+#include "rtc_base/network_route.h"
+
+namespace webrtc {
+
+// A Call represents a two-way connection carrying zero or more outgoing
+// and incoming media streams, transported over one or more RTP transports.
+
+// A Call instance can contain several send and/or receive streams. All streams
+// are assumed to have the same remote endpoint and will share bitrate estimates
+// etc.
+
+// When using the PeerConnection API, there is an one to one relationship
+// between the PeerConnection and the Call.
+
+class Call {
+ public:
+  using Config = CallConfig;
+  using Stats = CallBasicStats;
+
+  static Call* Create(const Call::Config& config);
+  static Call* Create(const Call::Config& config,
+                      Clock* clock,
+                      std::unique_ptr<RtpTransportControllerSendInterface>
+                          transportControllerSend);
+
+  virtual AudioSendStream* CreateAudioSendStream(
+      const AudioSendStream::Config& config) = 0;
+
+  virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
+
+  virtual AudioReceiveStreamInterface* CreateAudioReceiveStream(
+      const AudioReceiveStreamInterface::Config& config) = 0;
+  virtual void DestroyAudioReceiveStream(
+      AudioReceiveStreamInterface* receive_stream) = 0;
+
+  virtual VideoSendStream* CreateVideoSendStream(
+      VideoSendStream::Config config,
+      VideoEncoderConfig encoder_config) = 0;
+  virtual VideoSendStream* CreateVideoSendStream(
+      VideoSendStream::Config config,
+      VideoEncoderConfig encoder_config,
+      std::unique_ptr<FecController> fec_controller);
+  virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
+
+  virtual VideoReceiveStreamInterface* CreateVideoReceiveStream(
+      VideoReceiveStreamInterface::Config configuration) = 0;
+  virtual void DestroyVideoReceiveStream(
+      VideoReceiveStreamInterface* receive_stream) = 0;
+
+  // In order for a created VideoReceiveStreamInterface to be aware that it is
+  // protected by a FlexfecReceiveStream, the latter should be created before
+  // the former.
+  virtual FlexfecReceiveStream* CreateFlexfecReceiveStream(
+      const FlexfecReceiveStream::Config config) = 0;
+  virtual void DestroyFlexfecReceiveStream(
+      FlexfecReceiveStream* receive_stream) = 0;
+
+  // When a resource is overused, the Call will try to reduce the load on the
+  // sysem, for example by reducing the resolution or frame rate of encoded
+  // streams.
+  virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) = 0;
+
+  // All received RTP and RTCP packets for the call should be inserted to this
+  // PacketReceiver. The PacketReceiver pointer is valid as long as the
+  // Call instance exists.
+  virtual PacketReceiver* Receiver() = 0;
+
+  // This is used to access the transport controller send instance owned by
+  // Call. The send transport controller is currently owned by Call for legacy
+  // reasons. (for instance  variants of call tests are built on this assumtion)
+  // TODO(srte): Move ownership of transport controller send out of Call and
+  // remove this method interface.
+  virtual RtpTransportControllerSendInterface* GetTransportControllerSend() = 0;
+
+  // Returns the call statistics, such as estimated send and receive bandwidth,
+  // pacing delay, etc.
+  virtual Stats GetStats() const = 0;
+
+  // TODO(skvlad): When the unbundled case with multiple streams for the same
+  // media type going over different networks is supported, track the state
+  // for each stream separately. Right now it's global per media type.
+  virtual void SignalChannelNetworkState(MediaType media,
+                                         NetworkState state) = 0;
+
+  virtual void OnAudioTransportOverheadChanged(
+      int transport_overhead_per_packet) = 0;
+
+  // Called when a receive stream's local ssrc has changed and association with
+  // send streams needs to be updated.
+  virtual void OnLocalSsrcUpdated(AudioReceiveStreamInterface& stream,
+                                  uint32_t local_ssrc) = 0;
+  virtual void OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream,
+                                  uint32_t local_ssrc) = 0;
+  virtual void OnLocalSsrcUpdated(FlexfecReceiveStream& stream,
+                                  uint32_t local_ssrc) = 0;
+
+  virtual void OnUpdateSyncGroup(AudioReceiveStreamInterface& stream,
+                                 absl::string_view sync_group) = 0;
+
+  virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
+
+  virtual void SetClientBitratePreferences(
+      const BitrateSettings& preferences) = 0;
+
+  virtual const FieldTrialsView& trials() const = 0;
+
+  virtual TaskQueueBase* network_thread() const = 0;
+  virtual TaskQueueBase* worker_thread() const = 0;
+
+  virtual ~Call() {}
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_CALL_H_