blob: 94e8ed961e7e683bf0d89be57b10e2be347cefac [file] [log] [blame]
* Copyright (c) 2017 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 <stddef.h>
#include "absl/types/optional.h"
namespace webrtc {
struct AudioEncoderRuntimeConfig {
AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other);
AudioEncoderRuntimeConfig& operator=(const AudioEncoderRuntimeConfig& other);
bool operator==(const AudioEncoderRuntimeConfig& other) const;
absl::optional<int> bitrate_bps;
absl::optional<int> frame_length_ms;
// Note: This is what we tell the encoder. It doesn't have to reflect
// the actual NetworkMetrics; it's subject to our decision.
absl::optional<float> uplink_packet_loss_fraction;
absl::optional<bool> enable_fec;
absl::optional<bool> enable_dtx;
// Some encoders can encode fewer channels than the actual input to make
// better use of the bandwidth. |num_channels| sets the number of channels
// to encode.
absl::optional<size_t> num_channels;
// This is true if the last frame length change was an increase, and otherwise
// false.
// The value of this boolean is used to apply a different offset to the
// per-packet overhead that is reported by the BWE. The exact offset value
// is most important right after a frame length change, because the frame
// length change affects the overhead. In the steady state, the exact value is
// not important because the BWE will compensate.
bool last_fl_change_increase = false;
} // namespace webrtc