blob: 346ed5db1a28c892891f9c39a4562b1362e6fac3 [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 "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
namespace webrtc {
// An AudioNetworkAdaptor optimizes the audio experience by suggesting a
// suitable runtime configuration (bit rate, frame length, FEC, etc.) to the
// encoder based on network metrics.
class AudioNetworkAdaptor {
virtual ~AudioNetworkAdaptor() = default;
virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0;
virtual void SetUplinkPacketLossFraction(
float uplink_packet_loss_fraction) = 0;
virtual void SetRtt(int rtt_ms) = 0;
virtual void SetTargetAudioBitrate(int target_audio_bitrate_bps) = 0;
virtual void SetOverhead(size_t overhead_bytes_per_packet) = 0;
virtual AudioEncoderRuntimeConfig GetEncoderRuntimeConfig() = 0;
virtual void StartDebugDump(FILE* file_handle) = 0;
virtual void StopDebugDump() = 0;
virtual ANAStats GetStats() const = 0;
} // namespace webrtc