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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/call/rtp_bitrate_configurator.cc | |
parent | Initial commit. (diff) | |
download | firefox-esr-upstream.tar.xz firefox-esr-upstream.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/call/rtp_bitrate_configurator.cc')
-rw-r--r-- | third_party/libwebrtc/call/rtp_bitrate_configurator.cc | 135 |
1 files changed, 135 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/rtp_bitrate_configurator.cc b/third_party/libwebrtc/call/rtp_bitrate_configurator.cc new file mode 100644 index 0000000000..264dcdcb81 --- /dev/null +++ b/third_party/libwebrtc/call/rtp_bitrate_configurator.cc @@ -0,0 +1,135 @@ +/* + * 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 |