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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.
*/
#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
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