From 36d22d82aa202bb199967e9512281e9a53db42c9 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 21:33:14 +0200 Subject: Adding upstream version 115.7.0esr. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/call/call.h | 147 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 147 insertions(+) create mode 100644 third_party/libwebrtc/call/call.h (limited to 'third_party/libwebrtc/call/call.h') diff --git a/third_party/libwebrtc/call/call.h b/third_party/libwebrtc/call/call.h new file mode 100644 index 0000000000..42daa95a6c --- /dev/null +++ b/third_party/libwebrtc/call/call.h @@ -0,0 +1,147 @@ +/* + * 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 CALL_CALL_H_ +#define CALL_CALL_H_ + +#include +#include +#include +#include + +#include "absl/strings/string_view.h" +#include "api/adaptation/resource.h" +#include "api/media_types.h" +#include "api/task_queue/task_queue_base.h" +#include "call/audio_receive_stream.h" +#include "call/audio_send_stream.h" +#include "call/call_basic_stats.h" +#include "call/call_config.h" +#include "call/flexfec_receive_stream.h" +#include "call/packet_receiver.h" +#include "call/rtp_transport_controller_send_interface.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" +#include "rtc_base/copy_on_write_buffer.h" +#include "rtc_base/network/sent_packet.h" +#include "rtc_base/network_route.h" + +namespace webrtc { + +// A Call represents a two-way connection carrying zero or more outgoing +// and incoming media streams, transported over one or more RTP transports. + +// A Call instance can contain several send and/or receive streams. All streams +// are assumed to have the same remote endpoint and will share bitrate estimates +// etc. + +// When using the PeerConnection API, there is an one to one relationship +// between the PeerConnection and the Call. + +class Call { + public: + using Config = CallConfig; + using Stats = CallBasicStats; + + static Call* Create(const Call::Config& config); + static Call* Create(const Call::Config& config, + Clock* clock, + std::unique_ptr + transportControllerSend); + + virtual AudioSendStream* CreateAudioSendStream( + const AudioSendStream::Config& config) = 0; + + virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; + + virtual AudioReceiveStreamInterface* CreateAudioReceiveStream( + const AudioReceiveStreamInterface::Config& config) = 0; + virtual void DestroyAudioReceiveStream( + AudioReceiveStreamInterface* receive_stream) = 0; + + virtual VideoSendStream* CreateVideoSendStream( + VideoSendStream::Config config, + VideoEncoderConfig encoder_config) = 0; + virtual VideoSendStream* CreateVideoSendStream( + VideoSendStream::Config config, + VideoEncoderConfig encoder_config, + std::unique_ptr fec_controller); + virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0; + + virtual VideoReceiveStreamInterface* CreateVideoReceiveStream( + VideoReceiveStreamInterface::Config configuration) = 0; + virtual void DestroyVideoReceiveStream( + VideoReceiveStreamInterface* receive_stream) = 0; + + // In order for a created VideoReceiveStreamInterface to be aware that it is + // protected by a FlexfecReceiveStream, the latter should be created before + // the former. + virtual FlexfecReceiveStream* CreateFlexfecReceiveStream( + const FlexfecReceiveStream::Config config) = 0; + virtual void DestroyFlexfecReceiveStream( + FlexfecReceiveStream* receive_stream) = 0; + + // When a resource is overused, the Call will try to reduce the load on the + // sysem, for example by reducing the resolution or frame rate of encoded + // streams. + virtual void AddAdaptationResource(rtc::scoped_refptr resource) = 0; + + // All received RTP and RTCP packets for the call should be inserted to this + // PacketReceiver. The PacketReceiver pointer is valid as long as the + // Call instance exists. + virtual PacketReceiver* Receiver() = 0; + + // This is used to access the transport controller send instance owned by + // Call. The send transport controller is currently owned by Call for legacy + // reasons. (for instance variants of call tests are built on this assumtion) + // TODO(srte): Move ownership of transport controller send out of Call and + // remove this method interface. + virtual RtpTransportControllerSendInterface* GetTransportControllerSend() = 0; + + // Returns the call statistics, such as estimated send and receive bandwidth, + // pacing delay, etc. + virtual Stats GetStats() const = 0; + + // TODO(skvlad): When the unbundled case with multiple streams for the same + // media type going over different networks is supported, track the state + // for each stream separately. Right now it's global per media type. + virtual void SignalChannelNetworkState(MediaType media, + NetworkState state) = 0; + + virtual void OnAudioTransportOverheadChanged( + int transport_overhead_per_packet) = 0; + + // Called when a receive stream's local ssrc has changed and association with + // send streams needs to be updated. + virtual void OnLocalSsrcUpdated(AudioReceiveStreamInterface& stream, + uint32_t local_ssrc) = 0; + virtual void OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream, + uint32_t local_ssrc) = 0; + virtual void OnLocalSsrcUpdated(FlexfecReceiveStream& stream, + uint32_t local_ssrc) = 0; + + virtual void OnUpdateSyncGroup(AudioReceiveStreamInterface& stream, + absl::string_view sync_group) = 0; + + virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; + + virtual void SetClientBitratePreferences( + const BitrateSettings& preferences) = 0; + + virtual const FieldTrialsView& trials() const = 0; + + virtual TaskQueueBase* network_thread() const = 0; + virtual TaskQueueBase* worker_thread() const = 0; + + virtual ~Call() {} +}; + +} // namespace webrtc + +#endif // CALL_CALL_H_ -- cgit v1.2.3