Line data    Source code

       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:
      36           0 :   CallStatistics() {}
      37           0 :   ~CallStatistics() {}
      38             : 
      39             :   // Call this method to indicate that NetEq engaged in decoding. |speech_type|
      40             :   // is the audio-type according to NetEq, and |muted| indicates if the decoded
      41             :   // frame was produced in muted state.
      42             :   void DecodedByNetEq(AudioFrame::SpeechType speech_type, bool muted);
      43             : 
      44             :   // Call this method to indicate that a decoding call resulted in generating
      45             :   // silence, i.e. call to NetEq is bypassed and the output audio is zero.
      46             :   void DecodedBySilenceGenerator();
      47             : 
      48             :   // Get statistics for decoding. The statistics include the number of calls to
      49             :   // NetEq and silence generator, as well as the type of speech pulled of off
      50             :   // NetEq, c.f. declaration of AudioDecodingCallStats for detailed description.
      51             :   const AudioDecodingCallStats& GetDecodingStatistics() const;
      52             : 
      53             :  private:
      54             :   // Reset the decoding statistics.
      55             :   void ResetDecodingStatistics();
      56             : 
      57             :   AudioDecodingCallStats decoding_stat_;
      58             : };
      59             : 
      60             : }  // namespace acm2
      61             : 
      62             : }  // namespace webrtc
      63             : 
      64             : #endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_