blob: 38c2c2122a701c6944fd95c14e817b28ae68be9d [file] [log] [blame]
* Copyright (c) 2016 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"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
namespace webrtc {
class Controller {
struct NetworkMetrics {
rtc::Optional<int> uplink_bandwidth_bps;
rtc::Optional<float> uplink_packet_loss_fraction;
rtc::Optional<float> uplink_recoverable_packet_loss_fraction;
rtc::Optional<int> target_audio_bitrate_bps;
rtc::Optional<int> rtt_ms;
rtc::Optional<size_t> overhead_bytes_per_packet;
virtual ~Controller() = default;
// Informs network metrics update to this controller. Any non-empty field
// indicates an update on the corresponding network metric.
virtual void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) = 0;
virtual void MakeDecision(AudioEncoderRuntimeConfig* config) = 0;
} // namespace webrtc