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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/api/call/transport.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/api/call/transport.h')
-rw-r--r-- | third_party/libwebrtc/api/call/transport.h | 54 |
1 files changed, 54 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/call/transport.h b/third_party/libwebrtc/api/call/transport.h new file mode 100644 index 0000000000..8bff28825d --- /dev/null +++ b/third_party/libwebrtc/api/call/transport.h @@ -0,0 +1,54 @@ +/* + * 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 API_CALL_TRANSPORT_H_ +#define API_CALL_TRANSPORT_H_ + +#include <stddef.h> +#include <stdint.h> + +#include "api/ref_counted_base.h" +#include "api/scoped_refptr.h" + +namespace webrtc { + +// TODO(holmer): Look into unifying this with the PacketOptions in +// asyncpacketsocket.h. +struct PacketOptions { + PacketOptions(); + PacketOptions(const PacketOptions&); + ~PacketOptions(); + + // A 16 bits positive id. Negative ids are invalid and should be interpreted + // as packet_id not being set. + int packet_id = -1; + // Additional data bound to the RTP packet for use in application code, + // outside of WebRTC. + rtc::scoped_refptr<rtc::RefCountedBase> additional_data; + // Whether this is a retransmission of an earlier packet. + bool is_retransmit = false; + bool included_in_feedback = false; + bool included_in_allocation = false; +}; + +class Transport { + public: + virtual bool SendRtp(const uint8_t* packet, + size_t length, + const PacketOptions& options) = 0; + virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0; + + protected: + virtual ~Transport() {} +}; + +} // namespace webrtc + +#endif // API_CALL_TRANSPORT_H_ |