My setup code is

_audioEndPoint = new SDL2AudioEndPoint(AudioOutputName, _audioEncoder);
_audioEndPoint.SetAudioSinkFormat(_audioFormat.Value);

int frameSize = _audioEncoder.GetFrameSize();
_audioSource = new SDL2AudioSource(AudioInputName, _audioEncoder, (uint)frameSize);
_audioSource.SetAudioSourceFormat(_audioFormat.Value);

MediaStreamTrack audioInputTrack = new MediaStreamTrack(_audioSource.GetAudioSourceFormats(),
MediaStreamStatusEnum.SendRecv);
_peerConnection.addTrack(audioInputTrack);

_peerConnection.OnAudioFormatsNegotiated += (audioFormats) =>
{
_audioSource.SetAudioSourceFormat(audioFormats.First());
_audioEndPoint.SetAudioSinkFormat(audioFormats.First());
};

_audioSource.OnAudioSourceEncodedSample += _audioSource_OnAudioSourceEncodedSample;
_peerConnection.OnRtpPacketReceived += _peerConnection_OnRtpPacketReceived;

private void _audioSource_OnAudioSourceEncodedSample (uint durationRtpUnits, byte[] sample)
{
_peerConnection.SendAudio(durationRtpUnits, sample);
}

Sometime I am getting error from _peerConnection.SendAudio(durationRtpUnits, sample) and Error is "Cannot get the audio sending format, missing either local or remote audio track.". But not always.

Would you please help me in this regard.

I had to manually download SDL2.dll and copy it to the same directory as my application in order for the library to be properly found by the SipSorceryMedia functions.

Otherwise I received the following error:
Unhandled exception: System.DllNotFoundException: Unable to load DLL 'SDL2' or one of its dependencies: The specified module could not be found. (0x8007007E)

_audioSource = new SDL2AudioSource(AudioInputName, new AudioEncoder());
_audioSink = new SDL2AudioEndPoint(AudioOutputName, new AudioEncoder());

        MediaStreamTrack audioInputTrack = new MediaStreamTrack(_audioSource.GetAudioSourceFormats(),
            MediaStreamStatusEnum.SendOnly);
        _peerConnection.addTrack(audioInputTrack);

        _audioSource.OnAudioSourceEncodedSample += _peerConnection.SendAudio;
        _peerConnection.OnRtpPacketReceived += (IPEndPoint rep, SDPMediaTypesEnum media, RTPPacket rtpPkt) =>
        {
            if (media == SDPMediaTypesEnum.audio)
            {
                _audioSink.GotAudioRtp(rep, rtpPkt.Header.SyncSource,
                    rtpPkt.Header.SequenceNumber, rtpPkt.Header.Timestamp,
                    rtpPkt.Header.PayloadType, rtpPkt.Header.MarkerBit == 1,
                    rtpPkt.Payload);
            }
        };

        _peerConnection.OnAudioFormatsNegotiated += (audioFormats) =>
        {
            _audioSource.SetAudioSourceFormat(audioFormats.First());
            _audioSink.SetAudioSinkFormat(audioFormats.First());
        };

Is there any way two get an event when source or endpoint change in client app?

Is there any way to change source or endpoint at runtime?

Is there any way to change audio source and end point in already connected RTCPeerConnection that audio call is running in a webrtc environment.

Is there anyway to replace Track?

Hi @sipsorcery ,

Could you send me an invitation for this repository so I could push my code into it ?

Thx,
Christophe

I was trying to use SIPSorceryMedia.SDL2 library in a simple console application:

using SIPSorcery.Media;
using SIPSorceryMedia.SDL2;

SDL2Helper.InitSDL();
SDL2AudioEndPoint endpoint = new("Speakers (Realtek High Definition Audio)", new AudioEncoder());

but on the fifth line I get this exception:

System.MissingMethodException
HResult=0x80131513
Message=Method not found: 'Microsoft.Extensions.Logging.ILogger SIPSorcery.LogFactory.CreateLogger()'.
Source=SIPSorceryMedia.SDL2
StackTrace:
at SIPSorceryMedia.SDL2.SDL2AudioEndPoint..ctor(String audioOutDeviceName, IAudioEncoder audioEncoder)
at Program.Main(String[] args) in C:\Users<UserName>\source\repos\ConsoleApp4\ConsoleApp4\Program.cs:line 5

I am using .NET 8, and I tried both SDL2-2.0.20-win32-x64 (as mentioned in the README) and SDL-2.30.5-win32-x64 (lates version)
I have also tried .NET 9.0-Preview and I have tried to install Microsoft.Extensions.Logging, Microsoft.Extensions.Logging.Console and I got the same exception on the same line.

Edit:

The Exception is only thrown when SIPSorcery package is installed. when I remove it no exception is thrown.

Hi
I am trying to capture windows desktop sound using WasapiLoopbackCapture
waveIn = new WasapiLoopbackCapture();
waveIn.WaveFormat = new WaveFormat(48000 , 16 , 1);
waveIn.DataAvailable += LocalAudioSampleAvailable;
and I want to encode the received buffers in LocalAudioSampleAvailable with opus audio encoder , so I made some changes in WindowsAudioEndPoint
https://github.com/sipsorcery-org/SIPSorceryMedia.Windows/blob/master/src/WindowsAudioEndPoint.cs
WindowsAudioEndPoint audioSource = new WindowsAudioEndPoint(new OpusAudioEncoder());
audioSource.RestrictFormats(format => format.Codec == AudioCodecsEnum.OPUS);
but when I run my app , I get error in "chrome and Edge"
"Uncaught (in promise) DOMException: Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to set remote offer sdp: Failed to set remote audio description send parameters for m-section with mid='0'."
I fixed that by changing this line inside "OpusAudioEncoder" class
https://github.com/sipsorcery-org/SIPSorceryMedia.SDL2/blob/main/test/CheckCodec/OpusAudioEncoder.cs
public static readonly AudioFormat MEDIA_FORMAT_OPUS = new AudioFormat(111, "opus", SAMPLE_RATE, SAMPLE_RATE, 2, "a=fmtp:111 minptime=10;useinbandfec=1");
with this line
public static readonly AudioFormat MEDIA_FORMAT_OPUS = new AudioFormat(AudioCodecsEnum.OPUS, 111, SAMPLE_RATE, SAMPLE_RATE, 2, "a=fmtp:111 minptime=10;useinbandfec=1");
the problem is , I cant get the sound at the end correctly and I don't know why , the following code shows how I am encoding the buffers when I get them from WasapiLoopbackCapture inside LocalAudioSampleAvailable

private void LocalAudioSampleAvailable(object sender, WaveInEventArgs args)
        {
            // Note NAudio.Wave.WaveBuffer.ShortBuffer does not take into account little endian.
            // https://github.com/naudio/NAudio/blob/master/NAudio/Wave/WaveOutputs/WaveBuffer.cs
            //WaveBuffer wavBuffer = new WaveBuffer(args.Buffer.Take(args.BytesRecorded).ToArray());
            //byte[] encodedSample = _audioEncoder.EncodeAudio(wavBuffer.ShortBuffer, _audioFormatManager.SelectedFormat);
            try
            {
                byte[] buffer = args.Buffer.Take(args.BytesRecorded).ToArray();
                short[] pcm = buffer.Where((x, i) => i % 2 == 0).Select((y, i) => BitConverter.ToInt16(buffer, i * 2)).ToArray();
                byte[] encodedSample = _audioEncoder.EncodeAudio(pcm, _audioFormatManager.SelectedFormat);

                //OnAudioSourceEncodedSample?.Invoke((uint)_audioFormatManager.SelectedFormat.RtpClockRate / 1000 * AUDIO_SAMPLE_PERIOD_MILLISECONDS, encodedSample);
                OnAudioSourceEncodedSample?.Invoke((uint)encodedSample.Length, encodedSample);
                //OnAudioSourceEncodedSample?.Invoke((uint)(pcm.Length * _audioFormatManager.SelectedFormat.RtpClockRate / _audioFormatManager.SelectedFormat.ClockRate), encodedSample);

            }
            catch (Exception ex)
            {

            }
        }

I am getting sound but it contains too much noise and it is not clear at all and the quality is very low , can you help me with this ?
many thanks

sudo apt-get install libsdl2 gives following error:

E: Unable to locate package libsdl2.

WSL:
Windows 10 Linux Subsystem
RELEASE: Ubuntu 20.04.4 LTS
KERNEL: Linux 5.10.102.1-microsoft-standard-WSL2

sipsorcery-org/sipsorcery#864 (comment)

also project not running same problem
how i can install and use SDL2 in MAUI?
thank you

Application: ssss.exe
CoreCLR Version: 7.0.923.32018
.NET Version: 7.0.9
Description: The process was terminated due to an unhandled exception.
Exception Info: System.AccessViolationException: Attempted to read or write protected memory. This is often an indication that other memory is corrupt.
Stack:
at SDL2.SDL.SDL_CloseAudioDevice(UInt32)
at SDL2.SDL.SDL_CloseAudioDevice(UInt32)
at SIPSorceryMedia.SDL2.SDL2Helper.CloseAudioDevice(UInt32)
at SIPSorceryMedia.SDL2.SDL2Helper.CloseAudioPlaybackDevice(UInt32)
at SIPSorceryMedia.SDL2.SDL2AudioEndPoint.CloseAudioSink()
at SIPSorceryMedia.SDL2.SDL2AudioEndPoint.InitPlaybackDevice()
at SIPSorceryMedia.SDL2.SDL2AudioEndPoint.SetAudioSinkFormat(SIPSorceryMedia.Abstractions.AudioFormat)
at SIPSorcery.Media.VoIPMediaSession.AudioFormatsNegotiated(System.Collections.Generic.List1<SIPSorceryMedia.Abstractions.AudioFormat>) at SIPSorcery.Net.RTPSession.RaisedOnAudioFormatsNegotiated(Int32, System.Collections.Generic.List1<SIPSorceryMedia.Abstractions.AudioFormat>)
at SIPSorcery.net.RTP.AudioStream.CheckAudioFormatsNegotiation()
at SIPSorcery.Net.RTPSession.SetRemoteDescription(SIPSorcery.SIP.App.SdpType, SIPSorcery.Net.SDP)
at SIPSorcery.SIP.App.SIPUserAgent+d__135.MoveNext()
at System.Runtime.CompilerServices.AsyncMethodBuilderCore.Start[[SIPSorcery.SIP.App.SIPUserAgent+d__135, SIPSorcery, Version=6.0.12.0, Culture=neutral, PublicKeyToken=null]](d__135 ByRef)
at System.Runtime.CompilerServices.AsyncVoidMethodBuilder.Start[[SIPSorcery.SIP.App.SIPUserAgent+d__135, SIPSorcery, Version=6.0.12.0, Culture=neutral, PublicKeyToken=null]](d__135 ByRef)
at SIPSorcery.SIP.App.SIPUserAgent.ClientCallAnsweredHandler(SIPSorcery.SIP.App.ISIPClientUserAgent, SIPSorcery.SIP.SIPResponse)
at SIPSorcery.SIP.App.SIPClientUserAgent.ServerFinalResponseReceived(SIPSorcery.SIP.SIPEndPoint, SIPSorcery.SIP.SIPEndPoint, SIPSorcery.SIP.SIPTransaction, SIPSorcery.SIP.SIPResponse)
at SIPSorcery.SIP.UACInviteTransaction+<UACInviteTransaction_TransactionFinalResponseReceived>d__19.MoveNext()
at System.Runtime.CompilerServices.AsyncMethodBuilderCore.Start[[SIPSorcery.SIP.UACInviteTransaction+<UACInviteTransaction_TransactionFinalResponseReceived>d__19, SIPSorcery, Version=6.0.12.0, Culture=neutral, PublicKeyToken=null]](<UACInviteTransaction_TransactionFinalResponseReceived>d__19 ByRef)
at System.Runtime.CompilerServices.AsyncTaskMethodBuilder`1[[System.Net.Sockets.SocketError, System.Net.Primitives, Version=7.0.0.0, Culture=neutral, PublicKeyToken=b03f5f7f11d50a3a]].Start[[SIPSorcery.SIP.UACInviteTransaction+<UACInviteTransaction_TransactionFinalResponseReceived>d__19, SIPSorcery, Version=6.0.12.0, Culture=neutral, PublicKeyToken=null]](<UACInviteTransaction_TransactionFinalResponseReceived>d__19 ByRef)
at SIPSorcery.SIP.UACInviteTransaction.UACInviteTransaction_TransactionFinalResponseReceived(SIPSorcery.SIP.SIPEndPoint, SIPSorcery.SIP.SIPEndPoint, SIPSorcery.SIP.SIPTransaction, SIPSorcery.SIP.SIPResponse)
at SIPSorcery.SIP.SIPTransaction.GotResponse(SIPSorcery.SIP.SIPEndPoint, SIPSorcery.SIP.SIPEndPoint, SIPSorcery.SIP.SIPResponse)
at SIPSorcery.SIP.SIPTransport.SIPMessageReceived(SIPSorcery.SIP.SIPChannel, SIPSorcery.SIP.SIPEndPoint, SIPSorcery.SIP.SIPEndPoint, Byte[])
at SIPSorcery.SIP.SIPTransport.ProcessReceiveQueue()
at System.Threading.ExecutionContext.RunInternal(System.Threading.ExecutionContext, System.Threading.ContextCallback, System.Object)
at System.Threading.Tasks.Task.ExecuteWithThreadLocal(System.Threading.Tasks.Task ByRef, System.Threading.Thread)

In Linux (WSL-Ubuntu) "sudo apt-get install libsdl2" says "Unable to locate package libsdl2"

But "sudo apt install libsdl2-dev libsdl2-2.0-0 -y" is working. After installation "sudo apt install libsdl2-dev libsdl2-2.0-0 -y", SDL2 does not providing any choice to select audio input or output device. It only shows "RDP Source" as Audio Input and "RDP Sink" as audio output.

My built in mic is not working. Using external USB Mic. But SDL2 cannot recognize it.

Please help me solve this issue.

When using our app, if an audio input or output device is unplugged, I can switch to another available device on all platforms (Windows, MacOS, Linux). However, after plugging the device back in, the app can only switch back to it on Windows.

On MacOS, I encounter the following error: "Failed to open audio recording device [name]. SDL Error: CoreAudio error (AudioQueueStart): -66681".

This indicates that the unplugged device is still in use or blocked by SDL2 or MacOS. I have tried the following code in the "InitRecordingDevice" method after calling "CloseAudio()", but it still does not work:

SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDL_InitSubSystem(SDL_INIT_AUDIO);

I have also added "SDL_ClearQueuedAudio(dev)" in the "CloseAudio" method before calling "CloseAudioRecordingDevice", but this has not resolved the issue either.

Are there any solutions for this problem?

I am using SIPSorceryMedia.SDL2, version 1.0.0 from nuget.

As per Document, 'No additional steps are required for an x64 build. The nuget package includes the SDL v2.0.20 x64 binaries'. But without manually download the library, I am getting error (Unable to load DLL 'SDL2').

Both GetAudioPlaybackDevices() & GetAudioPlaybackDevices of SDL2Helper class returns empty list.

I can hear sound but it is not sending audio. Code is as follow:

OpusAudioEncoder audioEncoder = new OpusAudioEncoder();
AudioFormat? audioFormat = OpusAudioEncoder.MEDIA_FORMAT_OPUS;

        _audioEndPoint = new SDL2AudioEndPoint(AudioOutputName, audioEncoder);
        _audioEndPoint.SetAudioSinkFormat(audioFormat.Value);

        int frameSize = audioEncoder.GetFrameSize();
        _audioSource = new SDL2AudioSource(AudioInputName, audioEncoder, (uint)frameSize);
        _audioSource.SetAudioSourceFormat(audioFormat.Value);

        MediaStreamTrack audioInputTrack = new MediaStreamTrack(_audioSource.GetAudioSourceFormats(),
            MediaStreamStatusEnum.SendRecv);
        _peerConnection.addTrack(audioInputTrack);

        _audioSource.OnAudioSourceEncodedSample += _peerConnection.SendAudio;

        _peerConnection.OnRtpPacketReceived += (IPEndPoint rep, SDPMediaTypesEnum media, RTPPacket rtpPkt) =>
        {
            if (media == SDPMediaTypesEnum.audio)
            {
                _audioEndPoint.GotAudioRtp(rep, rtpPkt.Header.SyncSource,
                    rtpPkt.Header.SequenceNumber, rtpPkt.Header.Timestamp,
                    rtpPkt.Header.PayloadType, rtpPkt.Header.MarkerBit == 1,
                    rtpPkt.Payload);
            }
        };

        _peerConnection.OnAudioFormatsNegotiated += (audioFormats) =>
        {
            _audioSource.SetAudioSourceFormat(audioFormats.First());
            _audioEndPoint.SetAudioSinkFormat(audioFormats.First());
        };

When using the laptop's built-in speaker and microphone, there is a tendency for echo to occur. However, this issue does not arise when headphones are used.

Is there any echo/noise cancellation feature in SIPSorceryMedia.SDL2?

Code:
_audioEncoder = new OpusAudioEncoder();
_audioFormat = OpusAudioEncoder.MEDIA_FORMAT_OPUS;

_frameSize = _audioEncoder.GetFrameSize();
_audioSource = new SDL2AudioSource(AudioInputName, _audioEncoder, (uint)_frameSize);
_audioEndPoint = new SDL2AudioEndPoint(AudioOutputName, _audioEncoder);