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https://github.com/oxen-io/session-android.git
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d83a3d71bc
Merge in RedPhone // FREEBIE
56 lines
1.4 KiB
C++
56 lines
1.4 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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#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
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#include <assert.h>
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namespace webrtc {
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namespace acm2 {
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void CallStatistics::DecodedByNetEq(AudioFrame::SpeechType speech_type) {
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++decoding_stat_.calls_to_neteq;
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switch (speech_type) {
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case AudioFrame::kNormalSpeech: {
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++decoding_stat_.decoded_normal;
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break;
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}
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case AudioFrame::kPLC: {
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++decoding_stat_.decoded_plc;
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break;
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}
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case AudioFrame::kCNG: {
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++decoding_stat_.decoded_cng;
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break;
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}
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case AudioFrame::kPLCCNG: {
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++decoding_stat_.decoded_plc_cng;
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break;
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}
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case AudioFrame::kUndefined: {
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// If the audio is decoded by NetEq, |kUndefined| is not an option.
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assert(false);
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}
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}
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}
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void CallStatistics::DecodedBySilenceGenerator() {
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++decoding_stat_.calls_to_silence_generator;
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}
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const AudioDecodingCallStats& CallStatistics::GetDecodingStatistics() const {
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return decoding_stat_;
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}
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} // namespace acm2
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} // namespace webrtc
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