public ClientAudioProvider()
{
_filters = new OnlineFilter[2];
_filters[0] =
OnlineFilter.CreateBandpass(ImpulseResponse.Finite, AudioManager.INPUT_SAMPLE_RATE, 560, 3900);
_filters[1] =
OnlineFilter.CreateBandpass(ImpulseResponse.Finite, AudioManager.INPUT_SAMPLE_RATE, 100, 4500);
JitterBufferProviderInterface =
new JitterBufferProviderInterface(new WaveFormat(AudioManager.INPUT_SAMPLE_RATE, 2));
SampleProvider = new Pcm16BitToSampleProvider(JitterBufferProviderInterface);
_decoder = OpusDecoder.Create(AudioManager.INPUT_SAMPLE_RATE, 1);
_decoder.ForwardErrorCorrection = false;
_highPassFilter = BiQuadFilter.HighPassFilter(AudioManager.INPUT_SAMPLE_RATE, 520, 0.97f);
_lowPassFilter = BiQuadFilter.LowPassFilter(AudioManager.INPUT_SAMPLE_RATE, 4130, 2.0f);
var effect = new CachedAudioEffect(CachedAudioEffect.AudioEffectTypes.NATO_TONE);
if (effect.AudioEffectBytes.Length > 0)
{
natoTone = ConversionHelpers.ByteArrayToShortArray(effect.AudioEffectBytes);
var vol = Settings.GlobalSettingsStore.Instance.GetClientSetting(GlobalSettingsKeys.NATOToneVolume)
.FloatValue;
for (int i = 0; i < natoTone.Length; i++)
{
natoTone[i] = (short)(natoTone[i] * vol);
}
}
}
public CachedLoopingAudioProvider(IWaveProvider source, WaveFormat waveFormat, CachedAudioEffect.AudioEffectTypes effectType)
{
this.WaveFormat = waveFormat;
var effect = new CachedAudioEffect(effectType);
_audioEffectShort = ConversionHelpers.ByteArrayToShortArray(effect.AudioEffectBytes);
this.source = source;
}
public void ConversionTestShortByteArray()
{
short[] shorts = new short[] { 1, short.MaxValue, short.MinValue, 0 };
var bytes = ConversionHelpers.ShortArrayToByteArray(shorts);
var result = ConversionHelpers.ByteArrayToShortArray(bytes);
var query = shorts.Where((b, i) => b == result[i]);
Assert.AreEqual(shorts.Length, query.Count());
}
public void ConversionTestByteShortArray()
{
byte[] bytes = new byte[] { 255, 253, 102, 0, 5, 0, 0, 0 };
var shorts = ConversionHelpers.ByteArrayToShortArray(bytes);
var result = ConversionHelpers.ShortArrayToByteArray(shorts);
var query = bytes.Where((b, i) => b == result[i]);
Assert.AreEqual(bytes.Length, query.Count());
}
private void CreateAudioEffectDouble(CachedAudioEffect effect)
{
if (effect.Loaded)
{
var effectShort = ConversionHelpers.ByteArrayToShortArray(effect.AudioEffectBytes);
var effectDouble = new double[effectShort.Length];
for (int i = 0; i < effectShort.Length; i++)
{
effectDouble[i] = ((effectShort[i] / 32768f));
}
effect.AudioEffectDouble = effectDouble;
}
}
public CachedLoopingAudioProvider(IWaveProvider source, WaveFormat waveFormat, CachedAudioEffect.AudioEffectTypes effectType)
{
this.WaveFormat = waveFormat;
var effect = new CachedAudioEffect(effectType);
_audioEffectShort = ConversionHelpers.ByteArrayToShortArray(effect.AudioEffectBytes);
var vol = Settings.GlobalSettingsStore.Instance.GetClientSetting(GlobalSettingsKeys.NATOToneVolume)
.FloatValue;
for (int i = 0; i < _audioEffectShort.Length; i++)
{
_audioEffectShort[i] = (short)(_audioEffectShort[i] * vol);
}
this.source = source;
}
private void ChangeVolumeOfEffect(CachedAudioEffect effect, GlobalSettingsKeys key)
{
if (effect.Loaded)
{
var effectShort = ConversionHelpers.ByteArrayToShortArray(effect.AudioEffectBytes);
var effectDouble = new double[effectShort.Length];
var vol = Settings.GlobalSettingsStore.Instance.GetClientSetting(key)
.FloatValue;
for (int i = 0; i < effectShort.Length; i++)
{
effectDouble[i] = ((effectShort[i] / 32768f) * vol);
}
effect.AudioEffectDouble = effectDouble;
}
}
public void AddClientAudioSamples(ClientAudio audio)
{
var newTransmission = LikelyNewTransmission();
if (newTransmission)
{
Logger.Debug($"{LogClientId}| Likely New Transmission");
}
var decoded = Decoder.Decode(audio.EncodedAudio,
audio.EncodedAudio.Length, out var decodedLength, newTransmission);
if (decodedLength > 0)
{
// for some reason if this is removed then it lags?!
//guess it makes a giant buffer and only uses a little?
//Answer: makes a buffer of 4000 bytes - so throw away most of it
var tmp = new byte[decodedLength];
Buffer.BlockCopy(decoded, 0, tmp, 0, decodedLength);
audio.PcmAudioShort = ConversionHelpers.ByteArrayToShortArray(tmp);
if (newTransmission)
{
// System.Diagnostics.Debug.WriteLine(audio.ClientGuid+"ADDED");
//append ms of silence - this functions as our jitter buffer??
var silencePad = AudioManager.InputSampleRate / 1000 * SilencePad;
var newAudio = new short[audio.PcmAudioShort.Length + silencePad];
Buffer.BlockCopy(audio.PcmAudioShort, 0, newAudio, silencePad, audio.PcmAudioShort.Length);
audio.PcmAudioShort = newAudio;
}
LastUpdate = DateTime.Now.Ticks;
var pcmAudio = ConversionHelpers.ShortArrayToByteArray(audio.PcmAudioShort);
_speechAudioProvider.AddSamples(pcmAudio, 0, pcmAudio.Length);
}
else
{
Logger.Debug($"{LogClientId}| Failed to decode audio from Packet for client");
}
}
private void UdpAudioDecode()
{
try
{
while (!_stop)
{
try
{
var encodedOpusAudio = new byte[0];
_encodedAudio.TryTake(out encodedOpusAudio, 100000, _stopFlag.Token);
var time = GetTickCount64(); //should add at the receive instead?
if ((encodedOpusAudio != null) && (encodedOpusAudio.Length > 36))
{
// process
// check if we should play audio
var myClient = IsClientMetaDataValid(_guid);
if ((myClient != null) && _clientStateSingleton.DcsPlayerRadioInfo.IsCurrent())
{
//Decode bytes
var udpVoicePacket = UDPVoicePacket.DecodeVoicePacket(encodedOpusAudio);
// check the radio
RadioReceivingState receivingState = null;
var receivingRadio =
_clientStateSingleton.DcsPlayerRadioInfo.CanHearTransmission(
udpVoicePacket.Frequency,
(RadioInformation.Modulation)udpVoicePacket.Modulation,
udpVoicePacket.UnitId, out receivingState);
//Check that we're not transmitting on this radio
double receivingPowerLossPercent = 0;
float lineOfSightLoss = 0;
if ((receivingRadio != null) && (receivingState != null)
&&
((receivingRadio.modulation == RadioInformation.Modulation.INTERCOM)
// INTERCOM Modulation is 2 so if its two dont bother checking LOS and Range
||
(
HasLineOfSight(udpVoicePacket, out lineOfSightLoss)
&&
InRange(udpVoicePacket, out receivingPowerLossPercent)
&&
!ShouldBlockRxAsTransmitting(receivingState.ReceivedOn)
)
)
)
{
// RadioReceivingState[receivingState.ReceivedOn] = receivingState;
//DECODE audio
int len1;
var decoded = _decoder.Decode(udpVoicePacket.AudioPart1Bytes,
udpVoicePacket.AudioPart1Bytes.Length, out len1);
if (len1 > 0)
{
// for some reason if this is removed then it lags?!
//guess it makes a giant buffer and only uses a little?
var tmp = new byte[len1];
Buffer.BlockCopy(decoded, 0, tmp, 0, len1);
//ALL GOOD!
//create marker for bytes
var audio = new ClientAudio
{
ClientGuid = udpVoicePacket.Guid,
PcmAudioShort = ConversionHelpers.ByteArrayToShortArray(tmp),
//Convert to Shorts!
ReceiveTime = GetTickCount64(),
Frequency = udpVoicePacket.Frequency,
Modulation = udpVoicePacket.Modulation,
Volume = receivingRadio.volume,
ReceivedRadio = receivingState.ReceivedOn,
UnitId = udpVoicePacket.UnitId,
Encryption = udpVoicePacket.Encryption,
Decryptable =
(udpVoicePacket.Encryption == receivingRadio.encKey) &&
receivingRadio.enc,
// mark if we can decrypt it
RadioReceivingState = receivingState,
RecevingPower =
receivingPowerLossPercent, //loss of 1.0 or greater is total loss
LineOfSightLoss = lineOfSightLoss, // Loss of 1.0 or greater is total loss
PacketNumber = udpVoicePacket.PacketNumber
};
//handle effects
var radioState = RadioReceivingState[audio.ReceivedRadio];
if ((radioState == null) || radioState.PlayedEndOfTransmission ||
!radioState.IsReceiving)
{
var decrytable = audio.Decryptable || (audio.Encryption == 0);
//mark that we have decrpyted encrypted audio for sound effects
if (decrytable && (audio.Encryption > 0))
{
_audioManager.PlaySoundEffectStartReceive(audio.ReceivedRadio,
true,
audio.Volume);
}
else
{
_audioManager.PlaySoundEffectStartReceive(audio.ReceivedRadio,
false,
audio.Volume);
}
}
RadioReceivingState[audio.ReceivedRadio] = new RadioReceivingState
{
IsSecondary = receivingState.IsSecondary,
LastReceviedAt = DateTime.Now.Ticks,
PlayedEndOfTransmission = false,
ReceivedOn = receivingState.ReceivedOn
};
_audioManager.AddClientAudio(audio);
}
else
{
Logger.Info("Failed to decode audio from Packet");
}
}
}
}
}
catch (Exception ex)
{
Logger.Info("Failed to decode audio from Packet");
}
}
}
catch (OperationCanceledException)
{
Logger.Info("Stopped DeJitter Buffer");
}
}
public void AddClientAudioSamples(ClientAudio audio)
{
//sort out volume
// var timer = new Stopwatch();
// timer.Start();
bool newTransmission = LikelyNewTransmission();
int decodedLength = 0;
var decoded = _decoder.Decode(audio.EncodedAudio,
audio.EncodedAudio.Length, out decodedLength, newTransmission);
if (decodedLength <= 0)
{
Logger.Info("Failed to decode audio from Packet for client");
return;
}
// for some reason if this is removed then it lags?!
//guess it makes a giant buffer and only uses a little?
//Answer: makes a buffer of 4000 bytes - so throw away most of it
var tmp = new byte[decodedLength];
Buffer.BlockCopy(decoded, 0, tmp, 0, decodedLength);
audio.PcmAudioShort = ConversionHelpers.ByteArrayToShortArray(tmp);
var decrytable = audio.Decryptable || (audio.Encryption == 0);
if (decrytable)
{
//adjust for LOS + Distance + Volume
AdjustVolumeForLoss(audio);
if (audio.ReceivedRadio == 0 ||
audio.Modulation == (short)RadioInformation.Modulation.MIDS)
{
if (profileSettings.GetClientSettingBool(ProfileSettingsKeys.RadioEffects))
{
AddRadioEffectIntercom(audio);
}
}
else
{
AddRadioEffect(audio);
}
//final adjust
AdjustVolume(audio);
}
else
{
AddEncryptionFailureEffect(audio);
AddRadioEffect(audio);
//final adjust
AdjustVolume(audio);
}
if (newTransmission)
{
// System.Diagnostics.Debug.WriteLine(audio.ClientGuid+"ADDED");
//append ms of silence - this functions as our jitter buffer??
var silencePad = (AudioManager.OUTPUT_SAMPLE_RATE / 1000) * SILENCE_PAD;
var newAudio = new short[audio.PcmAudioShort.Length + silencePad];
Buffer.BlockCopy(audio.PcmAudioShort, 0, newAudio, silencePad, audio.PcmAudioShort.Length);
audio.PcmAudioShort = newAudio;
}
_lastReceivedOn = audio.ReceivedRadio;
LastUpdate = DateTime.Now.Ticks;
JitterBufferProviderInterface.AddSamples(new JitterBufferAudio
{
Audio =
SeperateAudio(ConversionHelpers.ShortArrayToByteArray(audio.PcmAudioShort),
audio.ReceivedRadio),
PacketNumber = audio.PacketNumber
});
//timer.Stop();
}
public byte[] AddClientAudioSamples(ClientAudio audio, bool skipEffects = false)
{
//sort out volume
// var timer = new Stopwatch();
// timer.Start();
bool newTransmission = LikelyNewTransmission();
var decoded = _decoder.Decode(audio.EncodedAudio,
audio.EncodedAudio.Length, out var decodedLength, newTransmission);
if (decodedLength <= 0)
{
Logger.Info("Failed to decode audio from Packet for client");
return(null);
}
// for some reason if this is removed then it lags?!
//guess it makes a giant buffer and only uses a little?
//Answer: makes a buffer of 4000 bytes - so throw away most of it
var tmp = new byte[decodedLength];
Buffer.BlockCopy(decoded, 0, tmp, 0, decodedLength);
audio.PcmAudioShort = ConversionHelpers.ByteArrayToShortArray(tmp);
//only get settings every 3 seconds - and cache them - issues with performance
long now = DateTime.Now.Ticks;
if (TimeSpan.FromTicks(now - lastRefresh).TotalSeconds > 3) //3 seconds since last refresh
{
lastRefresh = now;
natoToneEnabled = profileSettings.GetClientSettingBool(ProfileSettingsKeys.NATOTone);
hqToneEnabled = profileSettings.GetClientSettingBool(ProfileSettingsKeys.HAVEQUICKTone);
radioEffectsEnabled = profileSettings.GetClientSettingBool(ProfileSettingsKeys.RadioEffects);
clippingEnabled = profileSettings.GetClientSettingBool(ProfileSettingsKeys.RadioEffectsClipping);
hqToneVolume = profileSettings.GetClientSettingFloat(ProfileSettingsKeys.HQToneVolume);
natoToneVolume = profileSettings.GetClientSettingFloat(ProfileSettingsKeys.NATOToneVolume);
fmVol = profileSettings.GetClientSettingFloat(ProfileSettingsKeys.FMNoiseVolume);
hfVol = profileSettings.GetClientSettingFloat(ProfileSettingsKeys.HFNoiseVolume);
uhfVol = profileSettings.GetClientSettingFloat(ProfileSettingsKeys.UHFNoiseVolume);
vhfVol = profileSettings.GetClientSettingFloat(ProfileSettingsKeys.VHFNoiseVolume);
}
var decrytable = audio.Decryptable /* || (audio.Encryption == 0) <--- this test has already been performed by all callers and would require another call to check for STRICT_AUDIO_ENCRYPTION */;
if (decrytable)
{
//adjust for LOS + Distance + Volume
AdjustVolumeForLoss(audio);
if (!skipEffects)
{
if (audio.ReceivedRadio == 0 ||
audio.Modulation == (short)RadioInformation.Modulation.MIDS)
{
if (profileSettings.GetClientSettingBool(ProfileSettingsKeys.RadioEffects))
{
AddRadioEffectIntercom(audio);
}
}
else
{
AddRadioEffect(audio);
}
}
//final adjust
AdjustVolume(audio);
}
else
{
AddEncryptionFailureEffect(audio);
AddRadioEffect(audio);
//final adjust
AdjustVolume(audio);
}
if (newTransmission)
{
// System.Diagnostics.Debug.WriteLine(audio.ClientGuid+"ADDED");
//append ms of silence - this functions as our jitter buffer??
var silencePad = (AudioManager.OUTPUT_SAMPLE_RATE / 1000) * SILENCE_PAD;
var newAudio = new short[audio.PcmAudioShort.Length + silencePad];
Buffer.BlockCopy(audio.PcmAudioShort, 0, newAudio, silencePad, audio.PcmAudioShort.Length);
audio.PcmAudioShort = newAudio;
}
_lastReceivedOn = audio.ReceivedRadio;
LastUpdate = DateTime.Now.Ticks;
if (!passThrough)
{
JitterBufferProviderInterface.AddSamples(new JitterBufferAudio
{
Audio =
SeperateAudio(ConversionHelpers.ShortArrayToByteArray(audio.PcmAudioShort),
audio.ReceivedRadio),
PacketNumber = audio.PacketNumber
});
return(null);
}
else
{
//return MONO PCM 16 as bytes
return(ConversionHelpers.ShortArrayToByteArray(audio.PcmAudioShort));
}
//timer.Stop();
}
private void UdpAudioDecode()
{
try
{
while (!_stop)
{
try
{
var encodedOpusAudio = new byte[0];
_encodedAudio.TryTake(out encodedOpusAudio, 100000, _stopFlag.Token);
var time = GetTickCount64(); //should add at the receive instead?
if (encodedOpusAudio != null && encodedOpusAudio.Length > 36)
{
// process
// check if we should play audio
var myClient = IsClientMetaDataValid(_guid);
if (myClient != null && RadioDCSSyncServer.DcsPlayerRadioInfo.IsCurrent())
{
//Decode bytes
var udpVoicePacket = UDPVoicePacket.DecodeVoicePacket(encodedOpusAudio);
// check the radio
RadioReceivingState receivingState = null;
var receivingRadio =
RadioDCSSyncServer.DcsPlayerRadioInfo.CanHearTransmission(udpVoicePacket.Frequency,
udpVoicePacket.Modulation,
udpVoicePacket.UnitId, out receivingState);
//Check that we're not transmitting on this radio
double receivingPower = 0;
float lineOfSightLoss = 0;
if (receivingRadio != null && receivingState != null
&&
(receivingRadio.modulation == 2
// INTERCOM Modulation is 2 so if its two dont bother checking LOS and Range
||
(
HasLineOfSight(udpVoicePacket, out lineOfSightLoss)
&&
InRange(udpVoicePacket, out receivingPower)
&&
!ShouldBlockRxAsTransmitting(receivingState.ReceivedOn)
)
)
)
{
// RadioReceivingState[receivingState.ReceivedOn] = receivingState;
//DECODE audio
int len1;
var decoded = _decoder.Decode(udpVoicePacket.AudioPart1Bytes,
udpVoicePacket.AudioPart1Bytes.Length, out len1);
int len2;
var decoded2 = _decoder.Decode(udpVoicePacket.AudioPart2Bytes,
udpVoicePacket.AudioPart2Bytes.Length, out len2);
if (len1 > 0 && len2 > 0)
{
// for some reason if this is removed then it lags?!
//guess it makes a giant buffer and only uses a little?
var tmp = new byte[len1 + len2];
Buffer.BlockCopy(decoded, 0, tmp, 0, len1);
Buffer.BlockCopy(decoded2, 0, tmp, len1, len2);
//ALL GOOD!
//create marker for bytes
var audio = new ClientAudio
{
ClientGuid = udpVoicePacket.Guid,
PcmAudioShort = ConversionHelpers.ByteArrayToShortArray(tmp),
//Convert to Shorts!
ReceiveTime = GetTickCount64(),
Frequency = udpVoicePacket.Frequency,
Modulation = udpVoicePacket.Modulation,
Volume = receivingRadio.volume,
ReceivedRadio = receivingState.ReceivedOn,
UnitId = udpVoicePacket.UnitId,
Encryption = udpVoicePacket.Encryption,
Decryptable =
udpVoicePacket.Encryption == receivingRadio.encKey && receivingRadio.enc,
// mark if we can decrypt it
RadioReceivingState = receivingState,
RecevingPower = receivingPower,
LineOfSightLoss = lineOfSightLoss // Loss of 1.0 or greater is total loss
};
//add to JitterBuffer!
lock (_lock)
{
_jitterBuffer.AddAudio(audio);
}
}
}
}
}
}
catch (Exception ex)
{
Logger.Info("Failed Decoding");
}
}
}
catch (OperationCanceledException)
{
Logger.Info("Stopped DeJitter Buffer");
}
}