blob: 196c3ca2470d2db4b76bd8163735a23db67f356a [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.
*/
#ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_AUDIO_PLAYOUT_H_
#define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_AUDIO_PLAYOUT_H_
#include <stdint.h>
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/rtc_event_log/rtc_event.h"
#include "api/units/timestamp.h"
#include "logging/rtc_event_log/events/rtc_event_definition.h"
namespace webrtc {
struct LoggedAudioPlayoutEvent {
LoggedAudioPlayoutEvent() = default;
LoggedAudioPlayoutEvent(Timestamp timestamp, uint32_t ssrc)
: timestamp(timestamp), ssrc(ssrc) {}
int64_t log_time_us() const { return timestamp.us(); }
int64_t log_time_ms() const { return timestamp.ms(); }
Timestamp log_time() const { return timestamp; }
Timestamp timestamp = Timestamp::MinusInfinity();
uint32_t ssrc;
};
class RtcEventAudioPlayout final : public RtcEvent {
public:
static constexpr Type kType = Type::AudioPlayout;
explicit RtcEventAudioPlayout(uint32_t ssrc);
~RtcEventAudioPlayout() override = default;
Type GetType() const override { return kType; }
bool IsConfigEvent() const override { return false; }
std::unique_ptr<RtcEventAudioPlayout> Copy() const;
uint32_t ssrc() const { return ssrc_; }
static std::string Encode(rtc::ArrayView<const RtcEvent*> batch) {
return RtcEventAudioPlayout::definition_.EncodeBatch(batch);
}
static RtcEventLogParseStatus Parse(
absl::string_view encoded_bytes,
bool batched,
std::map<uint32_t, std::vector<LoggedAudioPlayoutEvent>>& output) {
std::vector<LoggedAudioPlayoutEvent> temp_output;
auto status = RtcEventAudioPlayout::definition_.ParseBatch(
encoded_bytes, batched, temp_output);
for (const LoggedAudioPlayoutEvent& event : temp_output) {
output[event.ssrc].push_back(event);
}
return status;
}
private:
RtcEventAudioPlayout(const RtcEventAudioPlayout& other);
const uint32_t ssrc_;
static constexpr RtcEventDefinition<RtcEventAudioPlayout,
LoggedAudioPlayoutEvent,
uint32_t>
definition_{{"AudioPlayout", RtcEventAudioPlayout::kType},
{&RtcEventAudioPlayout::ssrc_,
&LoggedAudioPlayoutEvent::ssrc,
{"ssrc", /*id=*/1, FieldType::kFixed32, /*width=*/32}}};
};
} // namespace webrtc
#endif // LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_AUDIO_PLAYOUT_H_