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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/pc/rtp_transport_internal.h | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/rtp_transport_internal.h')
-rw-r--r-- | third_party/libwebrtc/pc/rtp_transport_internal.h | 105 |
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_ |