/// <summary>
/// Event handler for receiving RTP packets from the remote party.
/// </summary>
/// <param name="remoteEP">The remote end point the RTP was received from.</param>
/// <param name="format">The audio format of the encoded packets.</param>
/// <param name="rtpPacket">The RTP packet with the media sample.</param>
public short[] DecodeAudio(byte[] encodedSample, AudioFormat format)
{
if (format.Codec == AudioCodecsEnum.G722)
{
if (_g722Decoder == null)
{
_g722Decoder = new G722Codec();
_g722DecoderState = new G722CodecState(G722_BIT_RATE, G722Flags.None);
}
short[] decodedPcm = new short[encodedSample.Length * 2];
int decodedSampleCount =
_g722Decoder.Decode(_g722DecoderState, decodedPcm, encodedSample, encodedSample.Length);
return(decodedPcm.Take(decodedSampleCount).ToArray());
}
else if (format.Codec == AudioCodecsEnum.PCMA)
{
return(encodedSample.Select(x => ALawDecoder.ALawToLinearSample(x)).ToArray());
}
else if (format.Codec == AudioCodecsEnum.PCMU)
{
return(encodedSample.Select(x => MuLawDecoder.MuLawToLinearSample(x)).ToArray());
}
else if (format.Codec == AudioCodecsEnum.L16)
{
// Samples are on the wire as big endian.
return(encodedSample.Where((x, i) => i % 2 == 0)
.Select((y, i) => (short)(encodedSample[i * 2] << 8 | encodedSample[i * 2 + 1])).ToArray());
}
else if (format.Codec == AudioCodecsEnum.PCM_S16LE)
{
// Samples are on the wire as little endian (well unlikely to be on the wire in this case but when they
// arrive from somewhere like the SkypeBot SDK they will be in little endian format).
return(encodedSample.Where((x, i) => i % 2 == 0)
.Select((y, i) => (short)(encodedSample[i * 2 + 1] << 8 | encodedSample[i * 2])).ToArray());
}
else
{
throw new ApplicationException($"Audio format {format.Codec} cannot be decoded.");
}
}
public byte[] EncodeAudio(short[] pcm, AudioFormat format)
{
if (format.Codec == AudioCodecsEnum.G722)
{
if (_g722Codec == null)
{
_g722Codec = new G722Codec();
_g722CodecState = new G722CodecState(G722_BIT_RATE, G722Flags.None);
}
int outputBufferSize = pcm.Length / 2;
byte[] encodedSample = new byte[outputBufferSize];
int res = _g722Codec.Encode(_g722CodecState, encodedSample, pcm, pcm.Length);
return(encodedSample);
}
else if (format.Codec == AudioCodecsEnum.PCMA)
{
return(pcm.Select(x => ALawEncoder.LinearToALawSample(x)).ToArray());
}
else if (format.Codec == AudioCodecsEnum.PCMU)
{
return(pcm.Select(x => MuLawEncoder.LinearToMuLawSample(x)).ToArray());
}
else if (format.Codec == AudioCodecsEnum.L16)
{
// When netstandard2.1 can be used.
//return MemoryMarshal.Cast<short, byte>(pcm)
// Put on the wire in network byte order (big endian).
return(pcm.SelectMany(x => new byte[] { (byte)(x >> 8), (byte)(x) }).ToArray());
}
else if (format.Codec == AudioCodecsEnum.PCM_S16LE)
{
// Put on the wire as little endian.
return(pcm.SelectMany(x => new byte[] { (byte)(x), (byte)(x >> 8) }).ToArray());
}
else
{
throw new ApplicationException($"Audio format {format.Codec} cannot be encoded.");
}
}
/// <summary>
/// Initialises the audio source as required.
/// </summary>
public override Task Start()
{
lock (this)
{
if (!IsStarted)
{
if (AudioLocalTrack == null || AudioLocalTrack.Capabilities == null ||
AudioLocalTrack.Capabilities.Count == 0)
{
throw new ApplicationException(
"Cannot start audio session without a local audio track being available.");
}
else if (AudioRemoteTrack == null || AudioRemoteTrack.Capabilities == null ||
AudioRemoteTrack.Capabilities.Count == 0)
{
throw new ApplicationException(
"Cannot start audio session without a remote audio track being available.");
}
_sendingFormat = base.GetSendingFormat(SDPMediaTypesEnum.audio);
_sendingAudioSampleRate = SDPMediaFormatInfo.GetClockRate(_sendingFormat.FormatCodec);
_sendingAudioRtpRate = SDPMediaFormatInfo.GetRtpClockRate(_sendingFormat.FormatCodec);
Log.LogDebug($"RTP audio session selected sending codec {_sendingFormat.FormatCodec}.");
if (_sendingFormat.FormatCodec == SDPMediaFormatsEnum.G722)
{
_g722Codec = new G722Codec();
_g722CodecState = new G722CodecState(G722_BIT_RATE, G722Flags.None);
_g722Decoder = new G722Codec();
_g722DecoderState = new G722CodecState(G722_BIT_RATE, G722Flags.None);
}
// If required start the audio source.
if (_audioOpts != null && _audioOpts.AudioSource != AudioSourcesEnum.None)
{
if (_audioOpts.AudioSource == AudioSourcesEnum.Silence)
{
_audioStreamTimer = new Timer(SendSilenceSample, null, 0, AUDIO_SAMPLE_PERIOD_MILLISECONDS);
}
else if (_audioOpts.AudioSource == AudioSourcesEnum.PinkNoise ||
_audioOpts.AudioSource == AudioSourcesEnum.WhiteNoise ||
_audioOpts.AudioSource == AudioSourcesEnum.SineWave)
{
_signalGenerator = new SignalGenerator(_sendingAudioSampleRate, 1);
switch (_audioOpts.AudioSource)
{
case AudioSourcesEnum.PinkNoise:
_signalGenerator.Type = SignalGeneratorType.Pink;
break;
case AudioSourcesEnum.SineWave:
_signalGenerator.Type = SignalGeneratorType.Sin;
break;
case AudioSourcesEnum.WhiteNoise:
default:
_signalGenerator.Type = SignalGeneratorType.White;
break;
}
_audioStreamTimer = new Timer(SendSignalGeneratorSample, null, 0,
AUDIO_SAMPLE_PERIOD_MILLISECONDS);
}
else if (_audioOpts.AudioSource == AudioSourcesEnum.Music)
{
if (_audioOpts.SourceFiles == null ||
!_audioOpts.SourceFiles.ContainsKey(_sendingFormat.FormatCodec))
{
Log.LogWarning($"Source file not set for codec {_sendingFormat.FormatCodec}.");
}
else
{
string sourceFile = _audioOpts.SourceFiles[_sendingFormat.FormatCodec];
if (String.IsNullOrEmpty(sourceFile) || !File.Exists(sourceFile))
{
Log.LogWarning(
"Could not start audio music source as the source file does not exist.");
}
else
{
_audioStreamReader = new StreamReader(sourceFile);
_audioStreamTimer = new Timer(SendMusicSample, null, 0,
AUDIO_SAMPLE_PERIOD_MILLISECONDS);
}
}
}
}
base.OnRtpPacketReceived += RtpPacketReceived;
}
return(base.Start());
}
}
/// <summary>
/// Creates a new RTP audio visual session with audio/video capturing and rendering capabilities.
/// </summary>
/// <param name="addrFamily">The address family to create the underlying socket on (IPv4 or IPv6).</param>
/// <param name="audioOptions">Options for the send and receive audio streams on this session.</param>
/// <param name="videoOptions">Options for the send and receive video streams on this session</param>
/// <param name="bindAddress">Optional. If specified this address will be used as the bind address for any RTP
/// and control sockets created. Generally this address does not need to be set. The default behaviour
/// is to bind to [::] or 0.0.0.0, depending on system support, which minimises network routing
/// causing connection issues.</param>
/// <param name="disableExternalAudioSource">If true then no attempt will be made to use an external audio
/// source, e.g. microphone.</param>
public RtpAVSession(AudioOptions audioOptions, VideoOptions videoOptions, IPAddress bindAddress = null, bool disableExternalAudioSource = false)
: base(false, false, false, bindAddress)
{
_audioOpts = audioOptions ?? DefaultAudioOptions;
_videoOpts = videoOptions ?? DefaultVideoOptions;
_disableExternalAudioSource = disableExternalAudioSource;
if (_audioOpts != null && _audioOpts.AudioCodecs != null &&
_audioOpts.AudioCodecs.Any(x => !(x == SDPMediaFormatsEnum.PCMU || x == SDPMediaFormatsEnum.PCMA || x == SDPMediaFormatsEnum.G722)))
{
throw new ApplicationException("Only PCMA, PCMU and G722 are supported for audio codec options.");
}
// Initialise the video decoding objects. Even if we are not sourcing video
// we need to be ready to receive and render.
_vpxDecoder = new VpxEncoder();
int res = _vpxDecoder.InitDecoder();
if (res != 0)
{
throw new ApplicationException("VPX decoder initialisation failed.");
}
_imgConverter = new ImageConvert();
if (_audioOpts.AudioSource != AudioSourcesEnum.None)
{
var pcmu = new SDPMediaFormat(SDPMediaFormatsEnum.PCMU);
//// RTP event support.
//int clockRate = pcmu.GetClockRate();
//SDPMediaFormat rtpEventFormat = new SDPMediaFormat(DTMF_EVENT_PAYLOAD_ID);
//rtpEventFormat.SetFormatAttribute($"{SDP.TELEPHONE_EVENT_ATTRIBUTE}/{clockRate}");
//rtpEventFormat.SetFormatParameterAttribute("0-16");
var audioCapabilities = new List <SDPMediaFormat>();
if (_audioOpts.AudioCodecs == null || _audioOpts.AudioCodecs.Count == 0)
{
audioCapabilities.Add(pcmu);
}
else
{
foreach (var codec in _audioOpts.AudioCodecs)
{
audioCapabilities.Add(new SDPMediaFormat(codec));
}
}
//audioCapabilities.Add(rtpEventFormat);
if (audioCapabilities.Any(x => x.FormatCodec == SDPMediaFormatsEnum.G722))
{
_g722Encode = new G722Codec();
_g722EncodeState = new G722CodecState(64000, G722Flags.None);
_g722Decode = new G722Codec();
_g722DecodeState = new G722CodecState(64000, G722Flags.None);
}
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, audioCapabilities);
addTrack(audioTrack);
}
if (_videoOpts.VideoSource != VideoSourcesEnum.None)
{
MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
addTrack(videoTrack);
}
// Where the magic (for processing received media) happens.
base.OnRtpPacketReceived += RtpPacketReceived;
}
/// <summary>
/// Encodes a buffer of G722
/// </summary>
/// <param name="state">Codec state</param>
/// <param name="outputBuffer">Output buffer (to contain encoded G722)</param>
/// <param name="inputBuffer">PCM 16 bit samples to encode</param>
/// <param name="inputBufferCount">Number of samples in the input buffer to encode</param>
/// <returns>Number of encoded bytes written into output buffer</returns>
public int Encode(G722CodecState state, byte[] outputBuffer, short[] inputBuffer, int inputBufferCount)
{
int dlow;
int dhigh;
int el;
int wd;
int wd1;
int ril;
int wd2;
int il4;
int ih2;
int wd3;
int eh;
int mih;
int i;
int j;
// Low and high band PCM from the QMF
int xlow;
int xhigh;
int g722_bytes;
// Even and odd tap accumulators
int sumeven;
int sumodd;
int ihigh;
int ilow;
int code;
g722_bytes = 0;
xhigh = 0;
for (j = 0; j < inputBufferCount;)
{
if (state.ItuTestMode)
{
xlow =
xhigh = inputBuffer[j++] >> 1;
}
else
{
if (state.EncodeFrom8000Hz)
{
xlow = inputBuffer[j++] >> 1;
}
else
{
// Apply the transmit QMF
// Shuffle the buffer down
for (i = 0; i < 22; i++)
{
state.QmfSignalHistory[i] = state.QmfSignalHistory[i + 2];
}
state.QmfSignalHistory[22] = inputBuffer[j++];
state.QmfSignalHistory[23] = inputBuffer[j++];
// Discard every other QMF output
sumeven = 0;
sumodd = 0;
for (i = 0; i < 12; i++)
{
sumodd += state.QmfSignalHistory[2 * i] * qmf_coeffs[i];
sumeven += state.QmfSignalHistory[2 * i + 1] * qmf_coeffs[11 - i];
}
xlow = (sumeven + sumodd) >> 14;
xhigh = (sumeven - sumodd) >> 14;
}
}
// Block 1L, SUBTRA
el = Saturate(xlow - state.Band[0].s);
// Block 1L, QUANTL
wd = (el >= 0) ? el : -(el + 1);
for (i = 1; i < 30; i++)
{
wd1 = (q6[i] * state.Band[0].det) >> 12;
if (wd < wd1)
{
break;
}
}
ilow = (el < 0) ? iln[i] : ilp[i];
// Block 2L, INVQAL
ril = ilow >> 2;
wd2 = qm4[ril];
dlow = (state.Band[0].det * wd2) >> 15;
// Block 3L, LOGSCL
il4 = rl42[ril];
wd = (state.Band[0].nb * 127) >> 7;
state.Band[0].nb = wd + wl[il4];
if (state.Band[0].nb < 0)
{
state.Band[0].nb = 0;
}
else if (state.Band[0].nb > 18432)
{
state.Band[0].nb = 18432;
}
// Block 3L, SCALEL
wd1 = (state.Band[0].nb >> 6) & 31;
wd2 = 8 - (state.Band[0].nb >> 11);
wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
state.Band[0].det = wd3 << 2;
Block4(state, 0, dlow);
if (state.EncodeFrom8000Hz)
{
// Just leave the high bits as zero
code = (0xC0 | ilow) >> (8 - state.BitsPerSample);
}
else
{
// Block 1H, SUBTRA
eh = Saturate(xhigh - state.Band[1].s);
// Block 1H, QUANTH
wd = (eh >= 0) ? eh : -(eh + 1);
wd1 = (564 * state.Band[1].det) >> 12;
mih = (wd >= wd1) ? 2 : 1;
ihigh = (eh < 0) ? ihn[mih] : ihp[mih];
// Block 2H, INVQAH
wd2 = qm2[ihigh];
dhigh = (state.Band[1].det * wd2) >> 15;
// Block 3H, LOGSCH
ih2 = rh2[ihigh];
wd = (state.Band[1].nb * 127) >> 7;
state.Band[1].nb = wd + wh[ih2];
if (state.Band[1].nb < 0)
{
state.Band[1].nb = 0;
}
else if (state.Band[1].nb > 22528)
{
state.Band[1].nb = 22528;
}
// Block 3H, SCALEH
wd1 = (state.Band[1].nb >> 6) & 31;
wd2 = 10 - (state.Band[1].nb >> 11);
wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
state.Band[1].det = wd3 << 2;
Block4(state, 1, dhigh);
code = ((ihigh << 6) | ilow) >> (8 - state.BitsPerSample);
}
if (state.Packed)
{
// Pack the code bits
state.OutBuffer |= (uint)(code << state.OutBits);
state.OutBits += state.BitsPerSample;
if (state.OutBits >= 8)
{
outputBuffer[g722_bytes++] = (byte)(state.OutBuffer & 0xFF);
state.OutBits -= 8;
state.OutBuffer >>= 8;
}
}
else
{
outputBuffer[g722_bytes++] = (byte)code;
}
}
return(g722_bytes);
}
/// <summary>
/// Decodes a buffer of G722
/// </summary>
/// <param name="state">Codec state</param>
/// <param name="outputBuffer">Output buffer (to contain decompressed PCM samples)</param>
/// <param name="inputG722Data"></param>
/// <param name="inputLength">Number of bytes in input G722 data to decode</param>
/// <returns>Number of samples written into output buffer</returns>
public int Decode(G722CodecState state, short[] outputBuffer, byte[] inputG722Data, int inputLength)
{
int dlowt;
int rlow;
int ihigh;
int dhigh;
int rhigh;
int xout1;
int xout2;
int wd1;
int wd2;
int wd3;
int code;
int outlen;
int i;
int j;
outlen = 0;
rhigh = 0;
for (j = 0; j < inputLength;)
{
if (state.Packed)
{
// Unpack the code bits
if (state.InBits < state.BitsPerSample)
{
state.InBuffer |= (uint)(inputG722Data[j++] << state.InBits);
state.InBits += 8;
}
code = (int)state.InBuffer & ((1 << state.BitsPerSample) - 1);
state.InBuffer >>= state.BitsPerSample;
state.InBits -= state.BitsPerSample;
}
else
{
code = inputG722Data[j++];
}
switch (state.BitsPerSample)
{
default:
case 8:
wd1 = code & 0x3F;
ihigh = (code >> 6) & 0x03;
wd2 = qm6[wd1];
wd1 >>= 2;
break;
case 7:
wd1 = code & 0x1F;
ihigh = (code >> 5) & 0x03;
wd2 = qm5[wd1];
wd1 >>= 1;
break;
case 6:
wd1 = code & 0x0F;
ihigh = (code >> 4) & 0x03;
wd2 = qm4[wd1];
break;
}
// Block 5L, LOW BAND INVQBL
wd2 = (state.Band[0].det * wd2) >> 15;
// Block 5L, RECONS
rlow = state.Band[0].s + wd2;
// Block 6L, LIMIT
if (rlow > 16383)
{
rlow = 16383;
}
else if (rlow < -16384)
{
rlow = -16384;
}
// Block 2L, INVQAL
wd2 = qm4[wd1];
dlowt = (state.Band[0].det * wd2) >> 15;
// Block 3L, LOGSCL
wd2 = rl42[wd1];
wd1 = (state.Band[0].nb * 127) >> 7;
wd1 += wl[wd2];
if (wd1 < 0)
{
wd1 = 0;
}
else if (wd1 > 18432)
{
wd1 = 18432;
}
state.Band[0].nb = wd1;
// Block 3L, SCALEL
wd1 = (state.Band[0].nb >> 6) & 31;
wd2 = 8 - (state.Band[0].nb >> 11);
wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
state.Band[0].det = wd3 << 2;
Block4(state, 0, dlowt);
if (!state.EncodeFrom8000Hz)
{
// Block 2H, INVQAH
wd2 = qm2[ihigh];
dhigh = (state.Band[1].det * wd2) >> 15;
// Block 5H, RECONS
rhigh = dhigh + state.Band[1].s;
// Block 6H, LIMIT
if (rhigh > 16383)
{
rhigh = 16383;
}
else if (rhigh < -16384)
{
rhigh = -16384;
}
// Block 2H, INVQAH
wd2 = rh2[ihigh];
wd1 = (state.Band[1].nb * 127) >> 7;
wd1 += wh[wd2];
if (wd1 < 0)
{
wd1 = 0;
}
else if (wd1 > 22528)
{
wd1 = 22528;
}
state.Band[1].nb = wd1;
// Block 3H, SCALEH
wd1 = (state.Band[1].nb >> 6) & 31;
wd2 = 10 - (state.Band[1].nb >> 11);
wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
state.Band[1].det = wd3 << 2;
Block4(state, 1, dhigh);
}
if (state.ItuTestMode)
{
outputBuffer[outlen++] = (short)(rlow << 1);
outputBuffer[outlen++] = (short)(rhigh << 1);
}
else
{
if (state.EncodeFrom8000Hz)
{
outputBuffer[outlen++] = (short)(rlow << 1);
}
else
{
// Apply the receive QMF
for (i = 0; i < 22; i++)
{
state.QmfSignalHistory[i] = state.QmfSignalHistory[i + 2];
}
state.QmfSignalHistory[22] = rlow + rhigh;
state.QmfSignalHistory[23] = rlow - rhigh;
xout1 = 0;
xout2 = 0;
for (i = 0; i < 12; i++)
{
xout2 += state.QmfSignalHistory[2 * i] * qmf_coeffs[i];
xout1 += state.QmfSignalHistory[2 * i + 1] * qmf_coeffs[11 - i];
}
outputBuffer[outlen++] = (short)(xout1 >> 11);
outputBuffer[outlen++] = (short)(xout2 >> 11);
}
}
}
return(outlen);
}
static void Block4(G722CodecState s, int band, int d)
{
int wd1;
int wd2;
int wd3;
int i;
// Block 4, RECONS
s.Band[band].d[0] = d;
s.Band[band].r[0] = Saturate(s.Band[band].s + d);
// Block 4, PARREC
s.Band[band].p[0] = Saturate(s.Band[band].sz + d);
// Block 4, UPPOL2
for (i = 0; i < 3; i++)
{
s.Band[band].sg[i] = s.Band[band].p[i] >> 15;
}
wd1 = Saturate(s.Band[band].a[1] << 2);
wd2 = (s.Band[band].sg[0] == s.Band[band].sg[1]) ? -wd1 : wd1;
if (wd2 > 32767)
{
wd2 = 32767;
}
wd3 = (s.Band[band].sg[0] == s.Band[band].sg[2]) ? 128 : -128;
wd3 += (wd2 >> 7);
wd3 += (s.Band[band].a[2] * 32512) >> 15;
if (wd3 > 12288)
{
wd3 = 12288;
}
else if (wd3 < -12288)
{
wd3 = -12288;
}
s.Band[band].ap[2] = wd3;
// Block 4, UPPOL1
s.Band[band].sg[0] = s.Band[band].p[0] >> 15;
s.Band[band].sg[1] = s.Band[band].p[1] >> 15;
wd1 = (s.Band[band].sg[0] == s.Band[band].sg[1]) ? 192 : -192;
wd2 = (s.Band[band].a[1] * 32640) >> 15;
s.Band[band].ap[1] = Saturate(wd1 + wd2);
wd3 = Saturate(15360 - s.Band[band].ap[2]);
if (s.Band[band].ap[1] > wd3)
{
s.Band[band].ap[1] = wd3;
}
else if (s.Band[band].ap[1] < -wd3)
{
s.Band[band].ap[1] = -wd3;
}
// Block 4, UPZERO
wd1 = (d == 0) ? 0 : 128;
s.Band[band].sg[0] = d >> 15;
for (i = 1; i < 7; i++)
{
s.Band[band].sg[i] = s.Band[band].d[i] >> 15;
wd2 = (s.Band[band].sg[i] == s.Band[band].sg[0]) ? wd1 : -wd1;
wd3 = (s.Band[band].b[i] * 32640) >> 15;
s.Band[band].bp[i] = Saturate(wd2 + wd3);
}
// Block 4, DELAYA
for (i = 6; i > 0; i--)
{
s.Band[band].d[i] = s.Band[band].d[i - 1];
s.Band[band].b[i] = s.Band[band].bp[i];
}
for (i = 2; i > 0; i--)
{
s.Band[band].r[i] = s.Band[band].r[i - 1];
s.Band[band].p[i] = s.Band[band].p[i - 1];
s.Band[band].a[i] = s.Band[band].ap[i];
}
// Block 4, FILTEP
wd1 = Saturate(s.Band[band].r[1] + s.Band[band].r[1]);
wd1 = (s.Band[band].a[1] * wd1) >> 15;
wd2 = Saturate(s.Band[band].r[2] + s.Band[band].r[2]);
wd2 = (s.Band[band].a[2] * wd2) >> 15;
s.Band[band].sp = Saturate(wd1 + wd2);
// Block 4, FILTEZ
s.Band[band].sz = 0;
for (i = 6; i > 0; i--)
{
wd1 = Saturate(s.Band[band].d[i] + s.Band[band].d[i]);
s.Band[band].sz += (s.Band[band].b[i] * wd1) >> 15;
}
s.Band[band].sz = Saturate(s.Band[band].sz);
// Block 4, PREDIC
s.Band[band].s = Saturate(s.Band[band].sp + s.Band[band].sz);
}
/// <summary>
/// Initialises the audio source as required.
/// </summary>
public override Task Start()
{
lock (this)
{
if (!_isStarted)
{
_isStarted = true;
if (AudioLocalTrack == null || AudioLocalTrack.Capabilties == null || AudioLocalTrack.Capabilties.Count == 0)
{
throw new ApplicationException("Cannot start audio session without a local audio track being available.");
}
else if (AudioRemoteTrack == null || AudioRemoteTrack.Capabilties == null || AudioRemoteTrack.Capabilties.Count == 0)
{
throw new ApplicationException("Cannot start audio session without a remote audio track being available.");
}
// Choose which codec to use.
//_sendingFormat = AudioLocalTrack.Capabilties
// .Where(x => x.FormatID != DTMF_EVENT_PAYLOAD_ID.ToString() && int.TryParse(x.FormatID, out _))
// .OrderBy(x => int.Parse(x.FormatID)).First();
_sendingFormat = base.GetSendingFormat(SDPMediaTypesEnum.audio);
Log.LogDebug($"RTP audio session selected sending codec {_sendingFormat.FormatCodec}.");
if (_sendingFormat.FormatCodec == SDPMediaFormatsEnum.G722)
{
_g722Codec = new G722Codec();
_g722CodecState = new G722CodecState(64000, G722Flags.None);
}
// If required start the audio source.
if (_audioOpts != null && _audioOpts.AudioSource != DummyAudioSourcesEnum.None)
{
if (_audioOpts.AudioSource == DummyAudioSourcesEnum.Silence)
{
_audioStreamTimer = new Timer(SendSilenceSample, null, 0, AUDIO_SAMPLE_PERIOD_MILLISECONDS);
}
else if (_audioOpts.AudioSource == DummyAudioSourcesEnum.PinkNoise ||
_audioOpts.AudioSource == DummyAudioSourcesEnum.WhiteNoise ||
_audioOpts.AudioSource == DummyAudioSourcesEnum.SineWave)
{
_signalGenerator = new SignalGenerator(SAMPLE_RATE, 1);
switch (_audioOpts.AudioSource)
{
case DummyAudioSourcesEnum.PinkNoise:
_signalGenerator.Type = SignalGeneratorType.Pink;
break;
case DummyAudioSourcesEnum.SineWave:
_signalGenerator.Type = SignalGeneratorType.Sin;
break;
case DummyAudioSourcesEnum.WhiteNoise:
default:
_signalGenerator.Type = SignalGeneratorType.White;
break;
}
_audioStreamTimer = new Timer(SendNoiseSample, null, 0, AUDIO_SAMPLE_PERIOD_MILLISECONDS);
}
else if (_audioOpts.AudioSource == DummyAudioSourcesEnum.Music)
{
if (_audioOpts.SourceFiles == null || !_audioOpts.SourceFiles.ContainsKey(_sendingFormat.FormatCodec))
{
Log.LogWarning($"Source file not set for codec {_sendingFormat.FormatCodec}.");
}
else
{
string sourceFile = _audioOpts.SourceFiles[_sendingFormat.FormatCodec];
if (String.IsNullOrEmpty(sourceFile) || !File.Exists(sourceFile))
{
Log.LogWarning("Could not start audio music source as the source file does not exist.");
}
else
{
_audioStreamReader = new StreamReader(sourceFile);
_audioStreamTimer = new Timer(SendMusicSample, null, 0, AUDIO_SAMPLE_PERIOD_MILLISECONDS);
}
}
}
}
}
return(base.Start());
}
}
public byte[] EncodeAudio(short[] pcm, AudioCodecsEnum codec, AudioSamplingRatesEnum sampleRate)
{
byte[] encodedSample = null;
if (codec == AudioCodecsEnum.G722)
{
if (_g722Codec == null)
{
_g722Codec = new G722Codec();
_g722CodecState = new G722CodecState(G722_BIT_RATE, G722Flags.None);
}
if (sampleRate == AudioSamplingRatesEnum.Rate16KHz)
{
// No up sampling required.
int outputBufferSize = pcm.Length / 2;
encodedSample = new byte[outputBufferSize];
int res = _g722Codec.Encode(_g722CodecState, encodedSample, pcm, pcm.Length);
}
else
{
// Up sample the supplied PCM signal by doubling each sample.
int outputBufferSize = pcm.Length;
encodedSample = new byte[outputBufferSize];
short[] pcmUpsampled = new short[pcm.Length * 2];
for (int i = 0; i < pcm.Length; i++)
{
pcmUpsampled[i * 2] = pcm[i];
pcmUpsampled[i * 2 + 1] = pcm[i];
}
_g722Codec.Encode(_g722CodecState, encodedSample, pcmUpsampled, pcmUpsampled.Length);
}
return(encodedSample);
}
else if (codec == AudioCodecsEnum.PCMA ||
codec == AudioCodecsEnum.PCMU)
{
Func <short, byte> encode = (codec == AudioCodecsEnum.PCMA) ?
(Func <short, byte>)ALawEncoder.LinearToALawSample : MuLawEncoder.LinearToMuLawSample;
if (sampleRate == AudioSamplingRatesEnum.Rate8KHz)
{
// No down sampling required.
int outputBufferSize = pcm.Length;
encodedSample = new byte[outputBufferSize];
for (int index = 0; index < pcm.Length; index++)
{
encodedSample[index] = encode(pcm[index]);
}
}
else
{
// Down sample the supplied PCM signal by skipping every second sample.
int outputBufferSize = pcm.Length / 2;
encodedSample = new byte[outputBufferSize];
int encodedIndex = 0;
// Skip every second sample.
for (int index = 0; index < pcm.Length; index += 2)
{
encodedSample[encodedIndex++] = encode(pcm[index]);
}
}
return(encodedSample);
}
else
{
throw new ApplicationException($"Audio format {codec} cannot be encoded.");
}
}
/// <summary>
/// Event handler for receiving RTP packets from the remote party.
/// </summary>
/// <param name="remoteEP">The remote end point the RTP was received from.</param>
/// <param name="codec">The encoding codec of the packets.</param>
/// <param name="rtpPacket">The RTP packet with the media sample.</param>
public byte[] DecodeAudio(byte[] encodedSample, AudioCodecsEnum codec, AudioSamplingRatesEnum sampleRate)
{
bool wants8kSamples = sampleRate == AudioSamplingRatesEnum.Rate8KHz;
bool wants16kSamples = sampleRate == AudioSamplingRatesEnum.Rate16KHz;
if (codec == AudioCodecsEnum.G722)
{
if (_g722Decoder == null)
{
_g722Decoder = new G722Codec();
_g722DecoderState = new G722CodecState(G722_BIT_RATE, G722Flags.None);
}
short[] decodedPcm16k = new short[encodedSample.Length * 2];
int decodedSampleCount = _g722Decoder.Decode(_g722DecoderState, decodedPcm16k, encodedSample, encodedSample.Length);
// The decoder provides short samples but streams and devices generally seem to want
// byte samples so convert them.
byte[] pcm8kBuffer = (wants8kSamples) ? new byte[decodedSampleCount] : null;
byte[] pcm16kBuffer = (wants16kSamples) ? new byte[decodedSampleCount * 2] : null;
for (int i = 0; i < decodedSampleCount; i++)
{
var bufferSample = BitConverter.GetBytes(decodedPcm16k[i]);
// For 8K samples the crude re-sampling to get from 16K to 8K is to skip
// every second sample.
if (pcm8kBuffer != null && i % 2 == 0)
{
pcm8kBuffer[(i / 2) * 2] = bufferSample[0];
pcm8kBuffer[(i / 2) * 2 + 1] = bufferSample[1];
}
// G722 provides 16k samples.
if (pcm16kBuffer != null)
{
pcm16kBuffer[i * 2] = bufferSample[0];
pcm16kBuffer[i * 2 + 1] = bufferSample[1];
}
}
return(pcm8kBuffer ?? pcm16kBuffer);
}
else if (codec == AudioCodecsEnum.PCMA ||
codec == AudioCodecsEnum.PCMU)
{
Func <byte, short> decode = (codec == AudioCodecsEnum.PCMA) ?
(Func <byte, short>)ALawDecoder.ALawToLinearSample : MuLawDecoder.MuLawToLinearSample;
byte[] pcm8kBuffer = (wants8kSamples) ? new byte[encodedSample.Length * 2] : null;
byte[] pcm16kBuffer = (wants16kSamples) ? new byte[encodedSample.Length * 4] : null;
for (int i = 0; i < encodedSample.Length; i++)
{
var bufferSample = BitConverter.GetBytes(decode(encodedSample[i]));
// G711 samples at 8KHz.
if (pcm8kBuffer != null)
{
pcm8kBuffer[i * 2] = bufferSample[0];
pcm8kBuffer[i * 2 + 1] = bufferSample[1];
}
// The crude up-sampling approach to get 16K samples from G711 is to
// duplicate each 8K sample.
// TODO: This re-sampling approach introduces artifacts. Applying a low pass
// filter seems to be recommended.
if (pcm16kBuffer != null)
{
pcm16kBuffer[i * 4] = bufferSample[0];
pcm16kBuffer[i * 4 + 1] = bufferSample[1];
pcm16kBuffer[i * 4 + 2] = bufferSample[0];
pcm16kBuffer[i * 4 + 3] = bufferSample[1];
}
}
return(pcm8kBuffer ?? pcm16kBuffer);
}
else
{
throw new ApplicationException($"Audio format {codec} cannot be decoded.");
}
}