Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(160)

Side by Side Diff: webrtc/modules/audio_coding/main/acm2/call_statistics.h

Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
(Empty)
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_MAIN_ACM2_CALL_STATISTICS_H_
12 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_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 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_MAIN_ACM2_CALL_STATISTICS_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698