chromium/third_party/webrtc/call/rtp_bitrate_configurator.cc

/*
 *  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.
 */

#include "call/rtp_bitrate_configurator.h"

#include <algorithm>

#include "rtc_base/checks.h"

namespace {

// Returns its smallest positive argument. If neither argument is positive,
// returns an arbitrary nonpositive value.
int MinPositive(int a, int b) { … }

}  // namespace

namespace webrtc {
RtpBitrateConfigurator::RtpBitrateConfigurator(
    const BitrateConstraints& bitrate_config)
    : … { … }

RtpBitrateConfigurator::~RtpBitrateConfigurator() = default;

BitrateConstraints RtpBitrateConfigurator::GetConfig() const { … }

absl::optional<BitrateConstraints>
RtpBitrateConfigurator::UpdateWithSdpParameters(
    const BitrateConstraints& bitrate_config) { … }

absl::optional<BitrateConstraints>
RtpBitrateConfigurator::UpdateWithClientPreferences(
    const BitrateSettings& bitrate_mask) { … }

// Relay cap can change only max bitrate.
absl::optional<BitrateConstraints> RtpBitrateConfigurator::UpdateWithRelayCap(
    DataRate cap) { … }

absl::optional<BitrateConstraints> RtpBitrateConfigurator::UpdateConstraints(
    const absl::optional<int>& new_start) { … }

}  // namespace webrtc