xref: /aosp_15_r20/external/webrtc/modules/audio_processing/test/api_call_statistics.h (revision d9f758449e529ab9291ac668be2861e7a55c2422)
1 /*
2  *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef MODULES_AUDIO_PROCESSING_TEST_API_CALL_STATISTICS_H_
12 #define MODULES_AUDIO_PROCESSING_TEST_API_CALL_STATISTICS_H_
13 
14 #include <vector>
15 
16 #include "absl/strings/string_view.h"
17 
18 namespace webrtc {
19 namespace test {
20 
21 // Collects statistics about the API call durations.
22 class ApiCallStatistics {
23  public:
24   enum class CallType { kRender, kCapture };
25 
26   // Adds a new datapoint.
27   void Add(int64_t duration_nanos, CallType call_type);
28 
29   // Prints out a report of the statistics.
30   void PrintReport() const;
31 
32   // Writes the call information to a file.
33   void WriteReportToFile(absl::string_view filename) const;
34 
35  private:
36   struct CallData {
37     CallData(int64_t duration_nanos, CallType call_type);
38     int64_t duration_nanos;
39     CallType call_type;
40   };
41   std::vector<CallData> calls_;
42 };
43 
44 }  // namespace test
45 }  // namespace webrtc
46 
47 #endif  // MODULES_AUDIO_PROCESSING_TEST_API_CALL_STATISTICS_H_
48