blob: d3c60683b44008fe35ddf2107afdf348c709b04c [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 "logging/rtc_event_log/rtc_stream_config.h"
namespace webrtc {
class RtcEventAudioSendStreamConfig final : public RtcEvent {
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_; }
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