Adding upstream version 110.0.1.upstream/110.0.1upstream

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

1 files changed, 105 insertions, 0 deletions

diff --git a/third_party/libwebrtc/pc/rtp_transport_internal.h b/third_party/libwebrtc/pc/rtp_transport_internal.h
new file mode 100644
index 0000000000..9e816113f1
--- /dev/null
+++ b/third_party/libwebrtc/pc/rtp_transport_internal.h
@@ -0,0 +1,105 @@
+/*
+ *  Copyright 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.
+ */
+
+#ifndef PC_RTP_TRANSPORT_INTERNAL_H_
+#define PC_RTP_TRANSPORT_INTERNAL_H_
+
+#include <string>
+
+#include "call/rtp_demuxer.h"
+#include "p2p/base/ice_transport_internal.h"
+#include "pc/session_description.h"
+#include "rtc_base/network_route.h"
+#include "rtc_base/ssl_stream_adapter.h"
+#include "rtc_base/third_party/sigslot/sigslot.h"
+
+namespace rtc {
+class CopyOnWriteBuffer;
+struct PacketOptions;
+}  // namespace rtc
+
+namespace webrtc {
+
+// This class is an internal interface; it is not accessible to API consumers
+// but is accessible to internal classes in order to send and receive RTP and
+// RTCP packets belonging to a single RTP session. Additional convenience and
+// configuration methods are also provided.
+class RtpTransportInternal : public sigslot::has_slots<> {
+ public:
+  virtual ~RtpTransportInternal() = default;
+
+  virtual void SetRtcpMuxEnabled(bool enable) = 0;
+
+  virtual const std::string& transport_name() const = 0;
+
+  // Sets socket options on the underlying RTP or RTCP transports.
+  virtual int SetRtpOption(rtc::Socket::Option opt, int value) = 0;
+  virtual int SetRtcpOption(rtc::Socket::Option opt, int value) = 0;
+
+  virtual bool rtcp_mux_enabled() const = 0;
+
+  virtual bool IsReadyToSend() const = 0;
+
+  // Called whenever a transport's ready-to-send state changes. The argument
+  // is true if all used transports are ready to send. This is more specific
+  // than just "writable"; it means the last send didn't return ENOTCONN.
+  sigslot::signal1<bool> SignalReadyToSend;
+
+  // Called whenever an RTCP packet is received. There is no equivalent signal
+  // for RTP packets because they would be forwarded to the BaseChannel through
+  // the RtpDemuxer callback.
+  sigslot::signal2<rtc::CopyOnWriteBuffer*, int64_t> SignalRtcpPacketReceived;
+
+  // Called whenever the network route of the P2P layer transport changes.
+  // The argument is an optional network route.
+  sigslot::signal1<absl::optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
+
+  // Called whenever a transport's writable state might change. The argument is
+  // true if the transport is writable, otherwise it is false.
+  sigslot::signal1<bool> SignalWritableState;
+
+  sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
+
+  virtual bool IsWritable(bool rtcp) const = 0;
+
+  // TODO(zhihuang): Pass the `packet` by copy so that the original data
+  // wouldn't be modified.
+  virtual bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
+                             const rtc::PacketOptions& options,
+                             int flags) = 0;
+
+  virtual bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
+                              const rtc::PacketOptions& options,
+                              int flags) = 0;
+
+  // This method updates the RTP header extension map so that the RTP transport
+  // can parse the received packets and identify the MID. This is called by the
+  // BaseChannel when setting the content description.
+  //
+  // TODO(zhihuang): Merging and replacing following methods handling header
+  // extensions with SetParameters:
+  //   UpdateRtpHeaderExtensionMap,
+  //   UpdateSendEncryptedHeaderExtensionIds,
+  //   UpdateRecvEncryptedHeaderExtensionIds,
+  //   CacheRtpAbsSendTimeHeaderExtension,
+  virtual void UpdateRtpHeaderExtensionMap(
+      const cricket::RtpHeaderExtensions& header_extensions) = 0;
+
+  virtual bool IsSrtpActive() const = 0;
+
+  virtual bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
+                                      RtpPacketSinkInterface* sink) = 0;
+
+  virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // PC_RTP_TRANSPORT_INTERNAL_H_