Files
platform-external-webrtc/webrtc/modules/audio_coding/acm2/call_statistics.cc
kjellander 3e6db2321c audio_coding: remove "main" directory
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}
2015-11-26 12:45:01 +00:00

56 lines
1.4 KiB
C++

/*
* 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.
*/
#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
#include <assert.h>
namespace webrtc {
namespace acm2 {
void CallStatistics::DecodedByNetEq(AudioFrame::SpeechType speech_type) {
++decoding_stat_.calls_to_neteq;
switch (speech_type) {
case AudioFrame::kNormalSpeech: {
++decoding_stat_.decoded_normal;
break;
}
case AudioFrame::kPLC: {
++decoding_stat_.decoded_plc;
break;
}
case AudioFrame::kCNG: {
++decoding_stat_.decoded_cng;
break;
}
case AudioFrame::kPLCCNG: {
++decoding_stat_.decoded_plc_cng;
break;
}
case AudioFrame::kUndefined: {
// If the audio is decoded by NetEq, |kUndefined| is not an option.
assert(false);
}
}
}
void CallStatistics::DecodedBySilenceGenerator() {
++decoding_stat_.calls_to_silence_generator;
}
const AudioDecodingCallStats& CallStatistics::GetDecodingStatistics() const {
return decoding_stat_;
}
} // namespace acm2
} // namespace webrtc