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}
70 lines
2.9 KiB
C++
70 lines
2.9 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_RTP_BITRATE_CONFIGURATOR_H_
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#define CALL_RTP_BITRATE_CONFIGURATOR_H_
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#include "api/bitrate_constraints.h"
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#include "api/transport/bitrate_settings.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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// RtpBitrateConfigurator calculates the bitrate configuration based on received
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// remote configuration combined with local overrides.
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class RtpBitrateConfigurator {
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public:
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explicit RtpBitrateConfigurator(const BitrateConstraints& bitrate_config);
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~RtpBitrateConfigurator();
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BitrateConstraints GetConfig() const;
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// The greater min and smaller max set by this and SetClientBitratePreferences
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// will be used. The latest non-negative start value from either call will be
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// used. Specifying a start bitrate (>0) will reset the current bitrate
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// estimate. This is due to how the 'x-google-start-bitrate' flag is currently
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// implemented. Passing -1 leaves the start bitrate unchanged. Behavior is not
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// guaranteed for other negative values or 0.
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// The optional return value is set with new configuration if it was updated.
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absl::optional<BitrateConstraints> UpdateWithSdpParameters(
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const BitrateConstraints& bitrate_config_);
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// The greater min and smaller max set by this and SetSdpBitrateParameters
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// will be used. The latest non-negative start value form either call will be
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// used. Specifying a start bitrate will reset the current bitrate estimate.
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// Assumes 0 <= min <= start <= max holds for set parameters.
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// Update the bitrate configuration
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// The optional return value is set with new configuration if it was updated.
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absl::optional<BitrateConstraints> UpdateWithClientPreferences(
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const BitrateSettings& bitrate_mask);
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private:
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// Applies update to the BitrateConstraints cached in |config_|, resetting
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// with |new_start| if set.
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absl::optional<BitrateConstraints> UpdateConstraints(
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const absl::optional<int>& new_start);
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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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// The config mask set by SetClientBitratePreferences.
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// 0 <= min <= start <= max
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BitrateSettings bitrate_config_mask_;
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// The config set by SetSdpBitrateParameters.
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// min >= 0, start != 0, max == -1 || max > 0
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BitrateConstraints base_bitrate_config_;
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RTC_DISALLOW_COPY_AND_ASSIGN(RtpBitrateConfigurator);
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};
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} // namespace webrtc
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#endif // CALL_RTP_BITRATE_CONFIGURATOR_H_
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