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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/rtp_receiver_interface.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/rtp_receiver_interface.h')
-rw-r--r-- | third_party/libwebrtc/api/rtp_receiver_interface.h | 123 |
1 files changed, 123 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/rtp_receiver_interface.h b/third_party/libwebrtc/api/rtp_receiver_interface.h new file mode 100644 index 0000000000..e4ec9b5986 --- /dev/null +++ b/third_party/libwebrtc/api/rtp_receiver_interface.h @@ -0,0 +1,123 @@ +/* + * 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 interfaces for RtpReceivers +// http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface + +#ifndef API_RTP_RECEIVER_INTERFACE_H_ +#define API_RTP_RECEIVER_INTERFACE_H_ + +#include <string> +#include <vector> + +#include "api/crypto/frame_decryptor_interface.h" +#include "api/dtls_transport_interface.h" +#include "api/frame_transformer_interface.h" +#include "api/media_stream_interface.h" +#include "api/media_types.h" +#include "api/rtp_parameters.h" +#include "api/scoped_refptr.h" +#include "api/transport/rtp/rtp_source.h" +#include "rtc_base/ref_count.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +class RtpReceiverObserverInterface { + public: + // Note: Currently if there are multiple RtpReceivers of the same media type, + // they will all call OnFirstPacketReceived at once. + // + // In the future, it's likely that an RtpReceiver will only call + // OnFirstPacketReceived when a packet is received specifically for its + // SSRC/mid. + virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0; + + protected: + virtual ~RtpReceiverObserverInterface() {} +}; + +class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface { + public: + virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0; + + // The dtlsTransport attribute exposes the DTLS transport on which the + // media is received. It may be null. + // https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-transport + // TODO(https://bugs.webrtc.org/907849) remove default implementation + virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const; + + // The list of streams that `track` is associated with. This is the same as + // the [[AssociatedRemoteMediaStreams]] internal slot in the spec. + // https://w3c.github.io/webrtc-pc/#dfn-associatedremotemediastreams + // TODO(hbos): Make pure virtual as soon as Chromium's mock implements this. + // TODO(https://crbug.com/webrtc/9480): Remove streams() in favor of + // stream_ids() as soon as downstream projects are no longer dependent on + // stream objects. + virtual std::vector<std::string> stream_ids() const; + virtual std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams() const; + + // Audio or video receiver? + virtual cricket::MediaType media_type() const = 0; + + // Not to be confused with "mid", this is a field we can temporarily use + // to uniquely identify a receiver until we implement Unified Plan SDP. + virtual std::string id() const = 0; + + // The WebRTC specification only defines RTCRtpParameters in terms of senders, + // but this API also applies them to receivers, similar to ORTC: + // http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*. + virtual RtpParameters GetParameters() const = 0; + // TODO(dinosaurav): Delete SetParameters entirely after rolling to Chromium. + // Currently, doesn't support changing any parameters. + virtual bool SetParameters(const RtpParameters& parameters) { return false; } + + // Does not take ownership of observer. + // Must call SetObserver(nullptr) before the observer is destroyed. + virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0; + + // Sets the jitter buffer minimum delay until media playout. Actual observed + // delay may differ depending on the congestion control. `delay_seconds` is a + // positive value including 0.0 measured in seconds. `nullopt` means default + // value must be used. + virtual void SetJitterBufferMinimumDelay( + absl::optional<double> delay_seconds) = 0; + + // TODO(zhihuang): Remove the default implementation once the subclasses + // implement this. Currently, the only relevant subclass is the + // content::FakeRtpReceiver in Chromium. + virtual std::vector<RtpSource> GetSources() const; + + // Sets a user defined frame decryptor that will decrypt the entire frame + // before it is sent across the network. This will decrypt the entire frame + // using the user provided decryption mechanism regardless of whether SRTP is + // enabled or not. + // TODO(bugs.webrtc.org/12772): Remove. + virtual void SetFrameDecryptor( + rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor); + + // Returns a pointer to the frame decryptor set previously by the + // user. This can be used to update the state of the object. + // TODO(bugs.webrtc.org/12772): Remove. + virtual rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor() const; + + // Sets a frame transformer between the depacketizer and the decoder to enable + // client code to transform received frames according to their own processing + // logic. + virtual void SetDepacketizerToDecoderFrameTransformer( + rtc::scoped_refptr<FrameTransformerInterface> frame_transformer); + + protected: + ~RtpReceiverInterface() override = default; +}; + +} // namespace webrtc + +#endif // API_RTP_RECEIVER_INTERFACE_H_ |