blob: aeeb28e21871b974564645c5b8cbffe4cf9afe23 [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 <memory>
#include "api/rtc_event_log/rtc_event.h"
#include "api/units/timestamp.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
namespace webrtc {
struct AudioEncoderRuntimeConfig;
class RtcEventAudioNetworkAdaptation final : public RtcEvent {
static constexpr Type kType = Type::AudioNetworkAdaptation;
explicit RtcEventAudioNetworkAdaptation(
std::unique_ptr<AudioEncoderRuntimeConfig> config);
~RtcEventAudioNetworkAdaptation() override;
Type GetType() const override { return kType; }
bool IsConfigEvent() const override { return false; }
std::unique_ptr<RtcEventAudioNetworkAdaptation> Copy() const;
const AudioEncoderRuntimeConfig& config() const { return *config_; }
RtcEventAudioNetworkAdaptation(const RtcEventAudioNetworkAdaptation& other);
const std::unique_ptr<const AudioEncoderRuntimeConfig> config_;
struct LoggedAudioNetworkAdaptationEvent {
LoggedAudioNetworkAdaptationEvent() = default;
LoggedAudioNetworkAdaptationEvent(Timestamp timestamp,
const AudioEncoderRuntimeConfig& config)
: timestamp(timestamp), config(config) {}
int64_t log_time_us() const { return; }
int64_t log_time_ms() const { return; }
Timestamp timestamp = Timestamp::MinusInfinity();
AudioEncoderRuntimeConfig config;
} // namespace webrtc