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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_receiver.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_receiver.h')
-rw-r--r-- | third_party/libwebrtc/pc/rtp_receiver.h | 101 |
1 files changed, 101 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/rtp_receiver.h b/third_party/libwebrtc/pc/rtp_receiver.h new file mode 100644 index 0000000000..16ab011f14 --- /dev/null +++ b/third_party/libwebrtc/pc/rtp_receiver.h @@ -0,0 +1,101 @@ +/* + * Copyright 2015 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. + */ + +// This file contains classes that implement RtpReceiverInterface. +// An RtpReceiver associates a MediaStreamTrackInterface with an underlying +// transport (provided by cricket::VoiceChannel/cricket::VideoChannel) + +#ifndef PC_RTP_RECEIVER_H_ +#define PC_RTP_RECEIVER_H_ + +#include <stdint.h> + +#include <string> +#include <vector> + +#include "absl/types/optional.h" +#include "api/crypto/frame_decryptor_interface.h" +#include "api/dtls_transport_interface.h" +#include "api/media_stream_interface.h" +#include "api/media_types.h" +#include "api/rtp_parameters.h" +#include "api/rtp_receiver_interface.h" +#include "api/scoped_refptr.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" +#include "api/video/video_source_interface.h" +#include "media/base/media_channel.h" +#include "media/base/video_broadcaster.h" +#include "pc/video_track_source.h" +#include "rtc_base/thread.h" + +namespace webrtc { + +// Internal class used by PeerConnection. +class RtpReceiverInternal : public RtpReceiverInterface { + public: + // Call on the signaling thread, to let the receiver know that the the + // embedded source object should enter a stopped/ended state and the track's + // state set to `kEnded`, a final state that cannot be reversed. + virtual void Stop() = 0; + + // Sets the underlying MediaEngine channel associated with this RtpSender. + // A VoiceMediaChannel should be used for audio RtpSenders and + // a VideoMediaChannel should be used for video RtpSenders. + // NOTE: + // * SetMediaChannel(nullptr) must be called before the media channel is + // destroyed. + // * This method must be invoked on the worker thread. + virtual void SetMediaChannel( + cricket::MediaReceiveChannelInterface* media_channel) = 0; + + // Configures the RtpReceiver with the underlying media channel, with the + // given SSRC as the stream identifier. + virtual void SetupMediaChannel(uint32_t ssrc) = 0; + + // Configures the RtpReceiver with the underlying media channel to receive an + // unsignaled receive stream. + virtual void SetupUnsignaledMediaChannel() = 0; + + virtual void set_transport( + rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) = 0; + // This SSRC is used as an identifier for the receiver between the API layer + // and the WebRtcVideoEngine, WebRtcVoiceEngine layer. + virtual absl::optional<uint32_t> ssrc() const = 0; + + // Call this to notify the RtpReceiver when the first packet has been received + // on the corresponding channel. + virtual void NotifyFirstPacketReceived() = 0; + + // Set the associated remote media streams for this receiver. The remote track + // will be removed from any streams that are no longer present and added to + // any new streams. + virtual void set_stream_ids(std::vector<std::string> stream_ids) = 0; + // TODO(https://crbug.com/webrtc/9480): Remove SetStreams() in favor of + // set_stream_ids() as soon as downstream projects are no longer dependent on + // stream objects. + virtual void SetStreams( + const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) = 0; + + // Returns an ID that changes if the attached track changes, but + // otherwise remains constant. Used to generate IDs for stats. + // The special value zero means that no track is attached. + virtual int AttachmentId() const = 0; + + protected: + static int GenerateUniqueId(); + + static std::vector<rtc::scoped_refptr<MediaStreamInterface>> + CreateStreamsFromIds(std::vector<std::string> stream_ids); +}; + +} // namespace webrtc + +#endif // PC_RTP_RECEIVER_H_ |