1 files changed, 88 insertions, 0 deletions
diff --git a/third_party/libwebrtc/logging/rtc_event_log/events/rtc_event_audio_playout.h b/third_party/libwebrtc/logging/rtc_event_log/events/rtc_event_audio_playout.h
new file mode 100644
index 0000000000..196c3ca247
--- /dev/null
+++ b/third_party/libwebrtc/logging/rtc_event_log/events/rtc_event_audio_playout.h
@@ -0,0 +1,88 @@
+/*
+ *  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_