public AecPc(int filterLengthInSamples, int samplesPerFrame, int samplesPerSecond)
{
// Default settings.
aecConfig = new AecConfig(filterLengthInSamples, samplesPerFrame, samplesPerSecond);
aecConfig.NlpMode = AecNlpMode.KAecNlpModerate;
aecConfig.SkewMode = false;
aecConfig.MetricsMode = false;
farendOld = new List <short[]> {
new short[aecConfig.SamplesPerFrame], new short[aecConfig.SamplesPerFrame]
};
set_config(aecConfig);
aec = new AecCore(aecConfig);
farendBuf = new RingBuffer(aecConfig.BufSizeSamp);
//if (WebRtcAec_CreateResampler(&aecpc->resampler) == -1)
sampFreq = 16000;
scSampFreq = 48000;
//WebRtcAec_InitResampler(aecpc->resampler, aecpc->scSampFreq)
splitSampFreq = sampFreq;
skewFrCtr = 0;
activity = 0;
delayChange = 1;
delayCtr = 0;
sum = 0;
counter = 0;
checkBuffSize = true;
firstVal = 0;
ECstartup = true;
bufSizeStart = 0;
checkBufSizeCtr = 0;
filtDelay = 0;
timeForDelayChange = 0;
knownDelay = 0;
lastDelayDiff = 0;
skew = 0;
resample = false;
highSkewCtr = 0;
sampFactor = (scSampFreq * 1.0f) / splitSampFreq;
}
/// <summary>
/// Constructs an instance of the WebRtcFilter.
/// </summary>
/// <param name="expectedAudioLatency">The expected audio latency in milliseconds</param>
/// <param name="filterLength">The length of the echo cancellation filter in milliseconds</param>
/// <param name="recordedAudioFormat">The audio format into which the recorded audio has been transformed prior to encoding (e.g., not the raw audio)</param>
/// <param name="playedAudioFormat">The audio format which the audio must be transformed after decoding and before playing</param>
/// <param name="enableAec">Whether to enable acoustic echo cancellation</param>
/// <param name="enableDenoise">Whether to enable denoising</param>
/// <param name="enableAgc">Whether to enable automatic gain control</param>
/// <param name="playedResampler">The resampler which should be used on played audio</param>
/// <param name="recordedResampler">The resampler which should be used on recorded audio</param>
public WebRtcFilter(int expectedAudioLatency, int filterLength,
AudioFormat recordedAudioFormat, AudioFormat playedAudioFormat,
bool enableAec, bool enableDenoise, bool enableAgc,
IAudioFilter playedResampler = null, IAudioFilter recordedResampler = null) :
base(expectedAudioLatency, filterLength, recordedAudioFormat, playedAudioFormat, playedResampler, recordedResampler)
{
// Default settings.
var aecConfig = new AecConfig(FilterLength, recordedAudioFormat.SamplesPerFrame, recordedAudioFormat.SamplesPerSecond)
{
NlpMode = AecNlpMode.KAecNlpModerate,
SkewMode = false,
MetricsMode = false
};
_ns = new NoiseSuppressor(recordedAudioFormat);
_aec = new AecCore(aecConfig);
if (aecConfig.NlpMode != AecNlpMode.KAecNlpConservative &&
aecConfig.NlpMode != AecNlpMode.KAecNlpModerate &&
aecConfig.NlpMode != AecNlpMode.KAecNlpAggressive)
{
throw new ArgumentException();
}
_aec.targetSupp = WebRtcConstants.targetSupp[(int)aecConfig.NlpMode];
_aec.minOverDrive = WebRtcConstants.minOverDrive[(int)aecConfig.NlpMode];
if (aecConfig.MetricsMode && aecConfig.MetricsMode != true)
{
throw new ArgumentException();
}
_aec.metricsMode = aecConfig.MetricsMode;
if (_aec.metricsMode)
{
_aec.InitMetrics();
}
_enableAec = enableAec;
_enableDenoise = enableDenoise;
_enableAgc = enableAgc;
_agc = new Agc(0, 255, Agc.AgcMode.AgcModeAdaptiveDigital, (uint)recordedAudioFormat.SamplesPerSecond);
}
