/* * Copyright (c) 2013 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 MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_ #define MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_ #include "api/audio/audio_frame.h" #include "modules/audio_coding/include/audio_coding_module_typedefs.h" // // This class is for book keeping of calls to ACM. It is not useful to log API // calls which are supposed to be called every 10ms, e.g. PlayoutData10Ms(), // however, it is useful to know the number of such calls in a given time // interval. The current implementation covers calls to PlayoutData10Ms() with // detailed accounting of the decoded speech type. // // Thread Safety // ============= // Please note that this class in not thread safe. The class must be protected // if different APIs are called from different threads. // namespace webrtc { namespace acm2 { class CallStatistics { public: CallStatistics() {} ~CallStatistics() {} // Call this method to indicate that NetEq engaged in decoding. `speech_type` // is the audio-type according to NetEq, and `muted` indicates if the decoded // frame was produced in muted state. void DecodedByNetEq(AudioFrame::SpeechType speech_type, bool muted); // Call this method to indicate that a decoding call resulted in generating // silence, i.e. call to NetEq is bypassed and the output audio is zero. void DecodedBySilenceGenerator(); // Get statistics for decoding. The statistics include the number of calls to // NetEq and silence generator, as well as the type of speech pulled of off // NetEq, c.f. declaration of AudioDecodingCallStats for detailed description. const AudioDecodingCallStats& GetDecodingStatistics() const; private: // Reset the decoding statistics. void ResetDecodingStatistics(); AudioDecodingCallStats decoding_stat_; }; } // namespace acm2 } // namespace webrtc #endif // MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_