b6b29e0718
The purpose is to make the fixture reusable in downstream projects. The CL adds the following things to API: - api/test/video_quality_test_fixture.h - api/test/create_video_quality_test_fixture.h The following things are moved to API: - call/bitrate_constraints.h (api/bitrate_constraints.h) - call/simulated_network.h (api/test/simulated_network.h) - call/media_type.h (api/mediatypes.h) These are required by the params struct passed to the fixture. I didn't attempt to split the params struct into an internal-only and public version in this CL, and as a result we need to pull in the above things. They are quite harmless though, so I think it's worth it in order to avoid splitting up the test config struct. This CL doesn't solve all the problems we need to implement downstream tests; we probably need to upstream tracing variants of FakeNetworkPipe for instance, but that will come later. This puts in place the basic structure for now. Bug: None Change-Id: I35e26ed126fad27bc7b2a465400291084f6ac911 Reviewed-on: https://webrtc-review.googlesource.com/69601 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23714}
118 lines
4.7 KiB
C++
118 lines
4.7 KiB
C++
/*
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* Copyright (c) 2017 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_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
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#define CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <string>
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#include "absl/types/optional.h"
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#include "api/bitrate_constraints.h"
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#include "api/transport/bitrate_settings.h"
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namespace rtc {
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struct SentPacket;
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struct NetworkRoute;
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class TaskQueue;
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} // namespace rtc
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namespace webrtc {
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class CallStatsObserver;
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class TargetTransferRateObserver;
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class Module;
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class PacedSender;
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class PacketFeedbackObserver;
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class PacketRouter;
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class RateLimiter;
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class RtcpBandwidthObserver;
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class RtpPacketSender;
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struct RtpKeepAliveConfig;
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class TransportFeedbackObserver;
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// An RtpTransportController should own everything related to the RTP
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// transport to/from a remote endpoint. We should have separate
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// interfaces for send and receive side, even if they are implemented
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// by the same class. This is an ongoing refactoring project. At some
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// point, this class should be promoted to a public api under
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// webrtc/api/rtp/.
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//
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// For a start, this object is just a collection of the objects needed
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// by the VideoSendStream constructor. The plan is to move ownership
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// of all RTP-related objects here, and add methods to create per-ssrc
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// objects which would then be passed to VideoSendStream. Eventually,
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// direct accessors like packet_router() should be removed.
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//
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// This should also have a reference to the underlying
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// webrtc::Transport(s). Currently, webrtc::Transport is implemented by
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// WebRtcVideoChannel and WebRtcVoiceMediaChannel, and owned by
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// WebrtcSession. Video and audio always uses different transport
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// objects, even in the common case where they are bundled over the
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// same underlying transport.
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//
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// Extracting the logic of the webrtc::Transport from BaseChannel and
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// subclasses into a separate class seems to be a prerequesite for
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// moving the transport here.
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class RtpTransportControllerSendInterface {
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public:
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virtual ~RtpTransportControllerSendInterface() {}
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virtual rtc::TaskQueue* GetWorkerQueue() = 0;
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virtual PacketRouter* packet_router() = 0;
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virtual TransportFeedbackObserver* transport_feedback_observer() = 0;
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virtual RtpPacketSender* packet_sender() = 0;
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virtual const RtpKeepAliveConfig& keepalive_config() const = 0;
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// SetAllocatedSendBitrateLimits sets bitrates limits imposed by send codec
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// settings.
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// |min_send_bitrate_bps| is the total minimum send bitrate required by all
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// sending streams. This is the minimum bitrate the PacedSender will use.
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// Note that SendSideCongestionController::OnNetworkChanged can still be
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// called with a lower bitrate estimate. |max_padding_bitrate_bps| is the max
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// bitrate the send streams request for padding. This can be higher than the
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// current network estimate and tells the PacedSender how much it should max
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// pad unless there is real packets to send.
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virtual void SetAllocatedSendBitrateLimits(int min_send_bitrate_bps,
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int max_padding_bitrate_bps,
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int total_bitrate_bps) = 0;
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virtual void SetPacingFactor(float pacing_factor) = 0;
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virtual void SetQueueTimeLimit(int limit_ms) = 0;
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virtual CallStatsObserver* GetCallStatsObserver() = 0;
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virtual void RegisterPacketFeedbackObserver(
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PacketFeedbackObserver* observer) = 0;
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virtual void DeRegisterPacketFeedbackObserver(
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PacketFeedbackObserver* observer) = 0;
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virtual void RegisterTargetTransferRateObserver(
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TargetTransferRateObserver* observer) = 0;
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virtual void OnNetworkRouteChanged(
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const std::string& transport_name,
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const rtc::NetworkRoute& network_route) = 0;
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virtual void OnNetworkAvailability(bool network_available) = 0;
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virtual RtcpBandwidthObserver* GetBandwidthObserver() = 0;
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virtual int64_t GetPacerQueuingDelayMs() const = 0;
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virtual int64_t GetFirstPacketTimeMs() const = 0;
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virtual void EnablePeriodicAlrProbing(bool enable) = 0;
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virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
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virtual void SetPerPacketFeedbackAvailable(bool available) = 0;
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virtual void SetSdpBitrateParameters(
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const BitrateConstraints& constraints) = 0;
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virtual void SetClientBitratePreferences(
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const BitrateSettings& preferences) = 0;
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};
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} // namespace webrtc
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#endif // CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
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