blob: 148dec2aa86377dd2a81602c44322b32a06645ed [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 "webrtc/api/optional.h"
namespace webrtc {
struct AudioEncoderRuntimeConfig {
AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other);
AudioEncoderRuntimeConfig& operator=(const AudioEncoderRuntimeConfig& other);
rtc::Optional<int> bitrate_bps;
rtc::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.
rtc::Optional<float> uplink_packet_loss_fraction;
rtc::Optional<bool> enable_fec;
rtc::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.
rtc::Optional<size_t> num_channels;
} // namespace webrtc