3e6db2321c
This is the last piece of the old directory layout of the modules. Duplicated header files are left in audio_coding/main/include until downstream code is updated to the new location. They have pragma warnings added to them and identical header guards as the new headers to avoid breaking things. BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc NOTRY=True NOPRESUBMIT=True Review URL: https://codereview.webrtc.org/1481493004 Cr-Commit-Position: refs/heads/master@{#10803}
64 lines
2.1 KiB
C++
64 lines
2.1 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
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#define WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
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#include "webrtc/common_types.h"
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#include "webrtc/modules/include/module_common_types.h"
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//
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// This class is for book keeping of calls to ACM. It is not useful to log API
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// calls which are supposed to be called every 10ms, e.g. PlayoutData10Ms(),
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// however, it is useful to know the number of such calls in a given time
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// interval. The current implementation covers calls to PlayoutData10Ms() with
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// detailed accounting of the decoded speech type.
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//
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// Thread Safety
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// =============
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// Please note that this class in not thread safe. The class must be protected
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// if different APIs are called from different threads.
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//
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namespace webrtc {
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namespace acm2 {
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class CallStatistics {
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public:
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CallStatistics() {}
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~CallStatistics() {}
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// Call this method to indicate that NetEq engaged in decoding. |speech_type|
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// is the audio-type according to NetEq.
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void DecodedByNetEq(AudioFrame::SpeechType speech_type);
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// Call this method to indicate that a decoding call resulted in generating
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// silence, i.e. call to NetEq is bypassed and the output audio is zero.
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void DecodedBySilenceGenerator();
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// Get statistics for decoding. The statistics include the number of calls to
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// NetEq and silence generator, as well as the type of speech pulled of off
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// NetEq, c.f. declaration of AudioDecodingCallStats for detailed description.
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const AudioDecodingCallStats& GetDecodingStatistics() const;
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private:
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// Reset the decoding statistics.
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void ResetDecodingStatistics();
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AudioDecodingCallStats decoding_stat_;
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};
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} // namespace acm2
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
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