| /* |
| * 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/array_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" |
| #include "logging/rtc_event_log/events/rtc_event_field_encoding.h" |
| #include "logging/rtc_event_log/events/rtc_event_log_parse_status.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_ |