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1 /*
2  *  Copyright (c) 2013 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 WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
12 #define WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
13 
14 #include "webrtc/common_types.h"
15 #include "webrtc/modules/include/module_common_types.h"
16 
17 //
18 // This class is for book keeping of calls to ACM. It is not useful to log API
19 // calls which are supposed to be called every 10ms, e.g. PlayoutData10Ms(),
20 // however, it is useful to know the number of such calls in a given time
21 // interval. The current implementation covers calls to PlayoutData10Ms() with
22 // detailed accounting of the decoded speech type.
23 //
24 // Thread Safety
25 // =============
26 // Please note that this class in not thread safe. The class must be protected
27 // if different APIs are called from different threads.
28 //
29 
30 namespace webrtc {
31 
32 namespace acm2 {
33 
34 class CallStatistics {
35  public:
CallStatistics()36   CallStatistics() {}
~CallStatistics()37   ~CallStatistics() {}
38 
39   // Call this method to indicate that NetEq engaged in decoding. |speech_type|
40   // is the audio-type according to NetEq.
41   void DecodedByNetEq(AudioFrame::SpeechType speech_type);
42 
43   // Call this method to indicate that a decoding call resulted in generating
44   // silence, i.e. call to NetEq is bypassed and the output audio is zero.
45   void DecodedBySilenceGenerator();
46 
47   // Get statistics for decoding. The statistics include the number of calls to
48   // NetEq and silence generator, as well as the type of speech pulled of off
49   // NetEq, c.f. declaration of AudioDecodingCallStats for detailed description.
50   const AudioDecodingCallStats& GetDecodingStatistics() const;
51 
52  private:
53   // Reset the decoding statistics.
54   void ResetDecodingStatistics();
55 
56   AudioDecodingCallStats decoding_stat_;
57 };
58 
59 }  // namespace acm2
60 
61 }  // namespace webrtc
62 
63 #endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
64