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}
57 lines
1.9 KiB
C++
57 lines
1.9 KiB
C++
/*
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* Copyright (c) 2018 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_CONFIG_H_
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#define CALL_CALL_CONFIG_H_
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#include "api/bitrate_constraints.h"
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#include "api/fec_controller.h"
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#include "api/rtcerror.h"
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#include "api/transport/network_control.h"
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#include "call/audio_state.h"
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#include "rtc_base/platform_file.h"
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namespace webrtc {
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class AudioProcessing;
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class RtcEventLog;
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struct CallConfig {
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explicit CallConfig(RtcEventLog* event_log);
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~CallConfig();
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RTC_DEPRECATED static constexpr int kDefaultStartBitrateBps = 300000;
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// Bitrate config used until valid bitrate estimates are calculated. Also
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// used to cap total bitrate used. This comes from the remote connection.
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BitrateConstraints bitrate_config;
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// AudioState which is possibly shared between multiple calls.
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// TODO(solenberg): Change this to a shared_ptr once we can use C++11.
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rtc::scoped_refptr<AudioState> audio_state;
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// Audio Processing Module to be used in this call.
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// TODO(solenberg): Change this to a shared_ptr once we can use C++11.
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AudioProcessing* audio_processing = nullptr;
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// RtcEventLog to use for this call. Required.
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// Use webrtc::RtcEventLog::CreateNull() for a null implementation.
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RtcEventLog* event_log = nullptr;
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// FecController to use for this call.
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FecControllerFactoryInterface* fec_controller_factory = nullptr;
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// Network controller factory to use for this call.
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NetworkControllerFactoryInterface* network_controller_factory = nullptr;
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};
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} // namespace webrtc
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#endif // CALL_CALL_CONFIG_H_
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