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/*
* 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_