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/*
* 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.
*/
#ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_AUDIO_SEND_STREAM_CONFIG_H_
#define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_AUDIO_SEND_STREAM_CONFIG_H_
#include <memory>
#include "api/rtc_event_log/rtc_event.h"
#include "logging/rtc_event_log/rtc_stream_config.h"
namespace webrtc {
class RtcEventAudioSendStreamConfig final : public RtcEvent {
public:
static constexpr Type kType = Type::AudioSendStreamConfig;
explicit RtcEventAudioSendStreamConfig(
std::unique_ptr<rtclog::StreamConfig> config);
~RtcEventAudioSendStreamConfig() override;
Type GetType() const override { return kType; }
bool IsConfigEvent() const override { return true; }
std::unique_ptr<RtcEventAudioSendStreamConfig> Copy() const;
const rtclog::StreamConfig& config() const { return *config_; }
private:
RtcEventAudioSendStreamConfig(const RtcEventAudioSendStreamConfig& other);
const std::unique_ptr<const rtclog::StreamConfig> config_;
};
struct LoggedAudioSendConfig {
LoggedAudioSendConfig() = default;
LoggedAudioSendConfig(int64_t timestamp_us, const rtclog::StreamConfig config)
: timestamp_us(timestamp_us), config(config) {}
int64_t log_time_us() const { return timestamp_us; }
int64_t log_time_ms() const { return timestamp_us / 1000; }
int64_t timestamp_us;
rtclog::StreamConfig config;
};
} // namespace webrtc
#endif // LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_AUDIO_SEND_STREAM_CONFIG_H_
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