webrtc/modules/audio_coding/audio_network_adaptor/controller.h
Sebastian Jansson cd2a92f8e0 Removes RPLR based FEC controller.
This is not used and adds a lot of maintenance overhead to
the code since it requires that the transport feedback adapter
communicates directly with audio send stream.

This also means that the packet loss tracker used as input for
this can be removed and a lot of wiring up code overall.

Bug: webrtc:9883
Change-Id: I25689fb622ed89cbb378c27212a159485f5f53be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156502
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29667}
2019-10-31 13:56:44 +00:00

42 lines
1.4 KiB
C++

/*
* 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.
*/
#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
#include "absl/types/optional.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
namespace webrtc {
class Controller {
public:
struct NetworkMetrics {
NetworkMetrics();
~NetworkMetrics();
absl::optional<int> uplink_bandwidth_bps;
absl::optional<float> uplink_packet_loss_fraction;
absl::optional<int> target_audio_bitrate_bps;
absl::optional<int> rtt_ms;
absl::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
#endif // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_