ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=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 -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
64 lines
2.2 KiB
C++
64 lines
2.2 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_MAIN_ACM2_CALL_STATISTICS_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_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_MAIN_ACM2_CALL_STATISTICS_H_
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