b9b146c9fe
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
70 lines
2.9 KiB
C++
70 lines
2.9 KiB
C++
/*
|
|
* Copyright (c) 2018 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_RTP_BITRATE_CONFIGURATOR_H_
|
|
#define CALL_RTP_BITRATE_CONFIGURATOR_H_
|
|
|
|
#include "api/transport/bitrate_settings.h"
|
|
#include "call/bitrate_constraints.h"
|
|
#include "rtc_base/constructormagic.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// RtpBitrateConfigurator calculates the bitrate configuration based on received
|
|
// remote configuration combined with local overrides.
|
|
class RtpBitrateConfigurator {
|
|
public:
|
|
explicit RtpBitrateConfigurator(const BitrateConstraints& bitrate_config);
|
|
~RtpBitrateConfigurator();
|
|
BitrateConstraints GetConfig() const;
|
|
|
|
// The greater min and smaller max set by this and SetClientBitratePreferences
|
|
// will be used. The latest non-negative start value from either call will be
|
|
// used. Specifying a start bitrate (>0) will reset the current bitrate
|
|
// estimate. This is due to how the 'x-google-start-bitrate' flag is currently
|
|
// implemented. Passing -1 leaves the start bitrate unchanged. Behavior is not
|
|
// guaranteed for other negative values or 0.
|
|
// The optional return value is set with new configuration if it was updated.
|
|
absl::optional<BitrateConstraints> UpdateWithSdpParameters(
|
|
const BitrateConstraints& bitrate_config_);
|
|
|
|
// The greater min and smaller max set by this and SetSdpBitrateParameters
|
|
// will be used. The latest non-negative start value form either call will be
|
|
// used. Specifying a start bitrate will reset the current bitrate estimate.
|
|
// Assumes 0 <= min <= start <= max holds for set parameters.
|
|
// Update the bitrate configuration
|
|
// The optional return value is set with new configuration if it was updated.
|
|
absl::optional<BitrateConstraints> UpdateWithClientPreferences(
|
|
const BitrateSettings& bitrate_mask);
|
|
|
|
private:
|
|
// Applies update to the BitrateConstraints cached in |config_|, resetting
|
|
// with |new_start| if set.
|
|
absl::optional<BitrateConstraints> UpdateConstraints(
|
|
const absl::optional<int>& new_start);
|
|
|
|
// Bitrate config used until valid bitrate estimates are calculated. Also
|
|
// used to cap total bitrate used. This comes from the remote connection.
|
|
BitrateConstraints bitrate_config_;
|
|
|
|
// The config mask set by SetClientBitratePreferences.
|
|
// 0 <= min <= start <= max
|
|
BitrateSettings bitrate_config_mask_;
|
|
|
|
// The config set by SetSdpBitrateParameters.
|
|
// min >= 0, start != 0, max == -1 || max > 0
|
|
BitrateConstraints base_bitrate_config_;
|
|
|
|
RTC_DISALLOW_COPY_AND_ASSIGN(RtpBitrateConfigurator);
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // CALL_RTP_BITRATE_CONFIGURATOR_H_
|