mirror of
https://github.com/DrKLO/Telegram.git
synced 2024-12-23 06:50:36 +01:00
76 lines
3.1 KiB
C++
76 lines
3.1 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 "absl/types/optional.h"
|
|
#include "api/transport/bitrate_settings.h"
|
|
#include "api/units/data_rate.h"
|
|
#include "rtc_base/constructor_magic.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);
|
|
|
|
// Apply a cap for relayed calls.
|
|
absl::optional<BitrateConstraints> UpdateWithRelayCap(DataRate cap);
|
|
|
|
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_;
|
|
|
|
// Bandwidth cap applied for relayed calls.
|
|
DataRate max_bitrate_over_relay_ = DataRate::PlusInfinity();
|
|
|
|
RTC_DISALLOW_COPY_AND_ASSIGN(RtpBitrateConfigurator);
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // CALL_RTP_BITRATE_CONFIGURATOR_H_
|