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/*
* 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) {
if (a <= 0) {
return b;
}
if (b <= 0) {
return a;
}
return std::min(a, b);
}
} // namespace
namespace webrtc {
RtpBitrateConfigurator::RtpBitrateConfigurator(
const BitrateConstraints& bitrate_config)
: bitrate_config_(bitrate_config), base_bitrate_config_(bitrate_config) {
RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
RTC_DCHECK_GE(bitrate_config.start_bitrate_bps,
bitrate_config.min_bitrate_bps);
if (bitrate_config.max_bitrate_bps != -1) {
RTC_DCHECK_GE(bitrate_config.max_bitrate_bps,
bitrate_config.start_bitrate_bps);
}
}
RtpBitrateConfigurator::~RtpBitrateConfigurator() = default;
BitrateConstraints RtpBitrateConfigurator::GetConfig() const {
return bitrate_config_;
}
absl::optional<BitrateConstraints>
RtpBitrateConfigurator::UpdateWithSdpParameters(
const BitrateConstraints& bitrate_config) {
RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
RTC_DCHECK_NE(bitrate_config.start_bitrate_bps, 0);
if (bitrate_config.max_bitrate_bps != -1) {
RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
}
absl::optional<int> new_start;
// Only update the "start" bitrate if it's set, and different from the old
// value. In practice, this value comes from the x-google-start-bitrate codec
// parameter in SDP, and setting the same remote description twice shouldn't
// restart bandwidth estimation.
if (bitrate_config.start_bitrate_bps != -1 &&
bitrate_config.start_bitrate_bps !=
base_bitrate_config_.start_bitrate_bps) {
new_start.emplace(bitrate_config.start_bitrate_bps);
}
base_bitrate_config_ = bitrate_config;
return UpdateConstraints(new_start);
}
absl::optional<BitrateConstraints>
RtpBitrateConfigurator::UpdateWithClientPreferences(
const BitrateSettings& bitrate_mask) {
bitrate_config_mask_ = bitrate_mask;
return UpdateConstraints(bitrate_mask.start_bitrate_bps);
}
// Relay cap can change only max bitrate.
absl::optional<BitrateConstraints> RtpBitrateConfigurator::UpdateWithRelayCap(
DataRate cap) {
if (cap.IsFinite()) {
RTC_DCHECK(!cap.IsZero());
}
max_bitrate_over_relay_ = cap;
return UpdateConstraints(absl::nullopt);
}
absl::optional<BitrateConstraints> RtpBitrateConfigurator::UpdateConstraints(
const absl::optional<int>& new_start) {
BitrateConstraints updated;
updated.min_bitrate_bps =
std::max(bitrate_config_mask_.min_bitrate_bps.value_or(0),
base_bitrate_config_.min_bitrate_bps);
updated.max_bitrate_bps =
MinPositive(bitrate_config_mask_.max_bitrate_bps.value_or(-1),
base_bitrate_config_.max_bitrate_bps);
updated.max_bitrate_bps =
MinPositive(updated.max_bitrate_bps, max_bitrate_over_relay_.bps_or(-1));
// If the combined min ends up greater than the combined max, the max takes
// priority.
if (updated.max_bitrate_bps != -1 &&
updated.min_bitrate_bps > updated.max_bitrate_bps) {
updated.min_bitrate_bps = updated.max_bitrate_bps;
}
// If there is nothing to update (min/max unchanged, no new bandwidth
// estimation start value), return early.
if (updated.min_bitrate_bps == bitrate_config_.min_bitrate_bps &&
updated.max_bitrate_bps == bitrate_config_.max_bitrate_bps &&
!new_start) {
return absl::nullopt;
}
if (new_start) {
// Clamp start by min and max.
updated.start_bitrate_bps = MinPositive(
std::max(*new_start, updated.min_bitrate_bps), updated.max_bitrate_bps);
} else {
updated.start_bitrate_bps = -1;
}
BitrateConstraints config_to_return = updated;
if (!new_start) {
updated.start_bitrate_bps = bitrate_config_.start_bitrate_bps;
}
bitrate_config_ = updated;
return config_to_return;
}
} // namespace webrtc
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