182 lines
6.9 KiB
C++
182 lines
6.9 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef CALL_CALL_H_
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#define CALL_CALL_H_
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#include <algorithm>
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/adaptation/resource.h"
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#include "api/media_types.h"
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#include "api/task_queue/task_queue_base.h"
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#include "call/audio_receive_stream.h"
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#include "call/audio_send_stream.h"
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#include "call/call_config.h"
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#include "call/flexfec_receive_stream.h"
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#include "call/packet_receiver.h"
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#include "call/rtp_transport_controller_send_interface.h"
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#include "call/video_receive_stream.h"
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#include "call/video_send_stream.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/network/sent_packet.h"
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#include "rtc_base/network_route.h"
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#include "rtc_base/ref_count.h"
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namespace webrtc {
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// A restricted way to share the module process thread across multiple instances
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// of Call that are constructed on the same worker thread (which is what the
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// peer connection factory guarantees).
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// SharedModuleThread supports a callback that is issued when only one reference
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// remains, which is used to indicate to the original owner that the thread may
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// be discarded.
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class SharedModuleThread : public rtc::RefCountInterface {
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protected:
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SharedModuleThread(std::unique_ptr<ProcessThread> process_thread,
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std::function<void()> on_one_ref_remaining);
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friend class rtc::scoped_refptr<SharedModuleThread>;
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~SharedModuleThread() override;
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public:
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// Allows injection of an externally created process thread.
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static rtc::scoped_refptr<SharedModuleThread> Create(
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std::unique_ptr<ProcessThread> process_thread,
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std::function<void()> on_one_ref_remaining);
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void EnsureStarted();
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ProcessThread* process_thread();
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private:
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void AddRef() const override;
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rtc::RefCountReleaseStatus Release() const override;
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class Impl;
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mutable std::unique_ptr<Impl> impl_;
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};
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// A Call instance can contain several send and/or receive streams. All streams
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// are assumed to have the same remote endpoint and will share bitrate estimates
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// etc.
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class Call {
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public:
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using Config = CallConfig;
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struct Stats {
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std::string ToString(int64_t time_ms) const;
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int send_bandwidth_bps = 0; // Estimated available send bandwidth.
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int max_padding_bitrate_bps = 0; // Cumulative configured max padding.
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int recv_bandwidth_bps = 0; // Estimated available receive bandwidth.
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int64_t pacer_delay_ms = 0;
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int64_t rtt_ms = -1;
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};
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static Call* Create(const Call::Config& config);
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static Call* Create(const Call::Config& config,
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Clock* clock,
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rtc::scoped_refptr<SharedModuleThread> call_thread,
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std::unique_ptr<ProcessThread> pacer_thread);
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static Call* Create(const Call::Config& config,
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Clock* clock,
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rtc::scoped_refptr<SharedModuleThread> call_thread,
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std::unique_ptr<RtpTransportControllerSendInterface>
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transportControllerSend);
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virtual AudioSendStream* CreateAudioSendStream(
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const AudioSendStream::Config& config) = 0;
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virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
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virtual AudioReceiveStream* CreateAudioReceiveStream(
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const AudioReceiveStream::Config& config) = 0;
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virtual void DestroyAudioReceiveStream(
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AudioReceiveStream* receive_stream) = 0;
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virtual VideoSendStream* CreateVideoSendStream(
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VideoSendStream::Config config,
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VideoEncoderConfig encoder_config) = 0;
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virtual VideoSendStream* CreateVideoSendStream(
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VideoSendStream::Config config,
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VideoEncoderConfig encoder_config,
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std::unique_ptr<FecController> fec_controller);
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virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
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virtual VideoReceiveStream* CreateVideoReceiveStream(
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VideoReceiveStream::Config configuration) = 0;
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virtual void DestroyVideoReceiveStream(
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VideoReceiveStream* receive_stream) = 0;
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// In order for a created VideoReceiveStream to be aware that it is
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// protected by a FlexfecReceiveStream, the latter should be created before
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// the former.
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virtual FlexfecReceiveStream* CreateFlexfecReceiveStream(
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const FlexfecReceiveStream::Config& config) = 0;
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virtual void DestroyFlexfecReceiveStream(
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FlexfecReceiveStream* receive_stream) = 0;
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// When a resource is overused, the Call will try to reduce the load on the
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// sysem, for example by reducing the resolution or frame rate of encoded
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// streams.
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virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) = 0;
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// All received RTP and RTCP packets for the call should be inserted to this
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// PacketReceiver. The PacketReceiver pointer is valid as long as the
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// Call instance exists.
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virtual PacketReceiver* Receiver() = 0;
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// This is used to access the transport controller send instance owned by
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// Call. The send transport controller is currently owned by Call for legacy
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// reasons. (for instance variants of call tests are built on this assumtion)
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// TODO(srte): Move ownership of transport controller send out of Call and
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// remove this method interface.
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virtual RtpTransportControllerSendInterface* GetTransportControllerSend() = 0;
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// Returns the call statistics, such as estimated send and receive bandwidth,
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// pacing delay, etc.
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virtual Stats GetStats() const = 0;
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// TODO(skvlad): When the unbundled case with multiple streams for the same
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// media type going over different networks is supported, track the state
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// for each stream separately. Right now it's global per media type.
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virtual void SignalChannelNetworkState(MediaType media,
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NetworkState state) = 0;
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virtual void OnAudioTransportOverheadChanged(
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int transport_overhead_per_packet) = 0;
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// Called when a receive stream's local ssrc has changed and association with
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// send streams needs to be updated.
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virtual void OnLocalSsrcUpdated(AudioReceiveStream& stream,
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uint32_t local_ssrc) = 0;
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virtual void OnUpdateSyncGroup(AudioReceiveStream& stream,
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const std::string& sync_group) = 0;
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virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
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virtual void SetClientBitratePreferences(
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const BitrateSettings& preferences) = 0;
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virtual const WebRtcKeyValueConfig& trials() const = 0;
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virtual TaskQueueBase* network_thread() const = 0;
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virtual TaskQueueBase* worker_thread() const = 0;
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virtual ~Call() {}
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};
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} // namespace webrtc
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#endif // CALL_CALL_H_
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