コード例 #1
0
ファイル: Stream.cs プロジェクト: mehmetcanbudak/LiveSPICE
        private static void ConvertSamples(IntPtr In, ASIOSampleType InType, Audio.SampleBuffer Out)
        {
            switch (InType)
            {
            //case ASIOSampleType.Int16MSB:
            //case ASIOSampleType.Int24MSB:
            //case ASIOSampleType.Int32MSB:
            //case ASIOSampleType.Float32MSB:
            //case ASIOSampleType.Float64MSB:
            //case ASIOSampleType.Int32MSB16:
            //case ASIOSampleType.Int32MSB18:
            //case ASIOSampleType.Int32MSB20:
            //case ASIOSampleType.Int32MSB24:
            case ASIOSampleType.Int16LSB: Audio.Util.LEi16ToLEf64(In, Out.Raw, Out.Count); break;

            //case ASIOSampleType.Int24LSB:
            case ASIOSampleType.Int32LSB: Audio.Util.LEi32ToLEf64(In, Out.Raw, Out.Count); break;

            case ASIOSampleType.Float32LSB: Audio.Util.LEf32ToLEf64(In, Out.Raw, Out.Count); break;

            case ASIOSampleType.Float64LSB: Audio.Util.CopyMemory(Out.Raw, In, Out.Count * sizeof(double)); break;

            //case ASIOSampleType.Int32LSB16:
            //case ASIOSampleType.Int32LSB18:
            //case ASIOSampleType.Int32LSB20:
            //case ASIOSampleType.Int32LSB24:
            default: throw new NotImplementedException("Unsupported sample type");
            }
        }
コード例 #2
0
ファイル: Stream.cs プロジェクト: mikeoliphant/LiveSPICE
        private void OnBufferSwitch(int Index, ASIOBool Direct)
        {
            Audio.SampleBuffer[] a = new Audio.SampleBuffer[input.Length];
            for (int i = 0; i < input.Length; ++i)
            {
                a[i] = input[i].Samples;

                ConvertSamples(
                    input[i].Info.buffers[Index],
                    input[i].Type,
                    a[i].Raw,
                    a[i].Count);
            }

            Audio.SampleBuffer[] b = new Audio.SampleBuffer[output.Length];
            for (int i = 0; i < output.Length; ++i)
            {
                b[i] = output[i].Samples;
            }

            callback(buffer, a, b, sampleRate);

            for (int i = 0; i < output.Length; ++i)
            {
                ConvertSamples(
                    b[i].Raw,
                    output[i].Info.buffers[Index],
                    output[i].Type,
                    b[i].Count);
            }
        }
コード例 #3
0
        private void OnBufferSwitch(int Index, ASIOBool Direct)
        {
            Audio.SampleBuffer[] a = new Audio.SampleBuffer[input.Length];
            for (int i = 0; i < input.Length; ++i)
            {
                a[i] = input[i].Samples;

                using (Audio.RawLock l = new Audio.RawLock(a[i], false, true))
                    ConvertSamples(
                        input[i].Info.buffers[Index],
                        input[i].Type,
                        l,
                        l.Count);
            }

            Audio.SampleBuffer[] b = new Audio.SampleBuffer[output.Length];
            for (int i = 0; i < output.Length; ++i)
            {
                b[i] = output[i].Samples;
            }

            callback(buffer, a, b, sampleRate);

            for (int i = 0; i < output.Length; ++i)
            {
                using (Audio.RawLock l = new Audio.RawLock(b[i], true, false))
                    ConvertSamples(
                        l,
                        output[i].Info.buffers[Index],
                        output[i].Type,
                        l.Count);
            }
        }
コード例 #4
0
ファイル: Stream.cs プロジェクト: mehmetcanbudak/LiveSPICE
        public Stream(Guid DeviceId, Audio.Stream.SampleHandler Callback, Channel[] Input, Channel[] Output)
            : base(Input, Output)
        {
            Log.Global.WriteLine(MessageType.Info, "Instantiating ASIO stream with {0} input channels and {1} output channels.", Input.Length, Output.Length);
            asio = new AsioObject(DeviceId);
            asio.Init(IntPtr.Zero);
            callback = Callback;

            // Just use the driver's preferred buffer size.
            bufferSize = asio.BufferSize.Preferred;

            ASIOBufferInfo[] infos = new ASIOBufferInfo[Input.Length + Output.Length];
            for (int i = 0; i < Input.Length; ++i)
            {
                infos[i].isInput    = ASIOBool.True;
                infos[i].channelNum = Input[i].Index;
            }
            for (int i = 0; i < Output.Length; ++i)
            {
                infos[Input.Length + i].isInput    = ASIOBool.False;
                infos[Input.Length + i].channelNum = Output[i].Index;
            }

            ASIOCallbacks callbacks = new ASIOCallbacks()
            {
                bufferSwitch         = OnBufferSwitch,
                sampleRateDidChange  = OnSampleRateChange,
                asioMessage          = OnAsioMessage,
                bufferSwitchTimeInfo = OnBufferSwitchTimeInfo
            };

            asio.CreateBuffers(infos, bufferSize, callbacks);

            // Create input buffers.
            input        = new BufferInfo[Input.Length];
            inputBuffers = new Audio.SampleBuffer[Input.Length];
            for (int i = 0; i < Input.Length; ++i)
            {
                input[i]        = new BufferInfo(infos[i], Input[i].Type);
                inputBuffers[i] = new Audio.SampleBuffer(bufferSize);
            }

            // Create output buffers.
            output        = new BufferInfo[Output.Length];
            outputBuffers = new Audio.SampleBuffer[Output.Length];
            for (int i = 0; i < Output.Length; ++i)
            {
                output[i]        = new BufferInfo(infos[Input.Length + i], Output[i].Type);
                outputBuffers[i] = new Audio.SampleBuffer(bufferSize);
            }

            sampleRate = asio.SampleRate;

            asio.Start();
        }
コード例 #5
0
        private static double AmplifySignal(Audio.SampleBuffer Samples, double Gain)
        {
            double peak = 0.0;

            for (int i = 0; i < Samples.Count; ++i)
            {
                double v = Samples[i];
                v         *= Gain;
                peak       = Math.Max(peak, Math.Abs(v));
                Samples[i] = v;
            }
            return(peak);
        }
コード例 #6
0
ファイル: Buffer.cs プロジェクト: alexbv16/LiveSPICE
        public Buffer(WAVEFORMATEX Format, int Count)
        {
            handle = GCHandle.Alloc(this);

            size = BlockAlignedSize(Format, Count);
            samples = new Audio.SampleBuffer(Count) { Tag = this };

            header = new WAVEHDR();
            header.lpData = Marshal.AllocHGlobal(size);
            header.dwUser = (IntPtr)handle;
            header.dwBufferLength = (uint)size;
            header.dwFlags = 0;
            pin = GCHandle.Alloc(header, GCHandleType.Pinned);
        }
コード例 #7
0
        private static double AmplifySignal(Audio.SampleBuffer Signal, double Gain)
        {
            double peak = 0.0;

            using (Audio.SamplesLock samples = new Audio.SamplesLock(Signal, true, true))
            {
                for (int i = 0; i < samples.Count; ++i)
                {
                    double v = samples[i];
                    v         *= Gain;
                    peak       = Math.Max(peak, Math.Abs(v));
                    samples[i] = v;
                }
            }
            return(peak);
        }
コード例 #8
0
ファイル: Buffer.cs プロジェクト: mikeoliphant/LiveSPICE
        public Buffer(WAVEFORMATEX Format, int Count)
        {
            samples = new Audio.SampleBuffer(Count)
            {
                Tag = this
            };

            int size = BlockAlignedSize(Format, Count);

            data                  = new byte[size];
            dataPin               = GCHandle.Alloc(data, GCHandleType.Pinned);
            header                = new WAVEHDR();
            headerPin             = GCHandle.Alloc(header, GCHandleType.Pinned);
            header.lpData         = dataPin.AddrOfPinnedObject();
            header.dwBufferLength = (uint)size;
            header.dwFlags        = 0;
        }
コード例 #9
0
        public Buffer(WAVEFORMATEX Format, int Count)
        {
            handle = GCHandle.Alloc(this);

            size    = BlockAlignedSize(Format, Count);
            samples = new Audio.SampleBuffer(Count)
            {
                Tag = this
            };

            header                = new WAVEHDR();
            header.lpData         = Marshal.AllocHGlobal(size);
            header.dwUser         = (IntPtr)handle;
            header.dwBufferLength = (uint)size;
            header.dwFlags        = 0;
            pin = GCHandle.Alloc(header, GCHandleType.Pinned);
        }
コード例 #10
0
ファイル: NullStream.cs プロジェクト: mikeoliphant/LiveSPICE
        private void Proc()
        {
            Audio.SampleBuffer[] input  = new Audio.SampleBuffer[] { };
            Audio.SampleBuffer[] output = new Audio.SampleBuffer[] { };

            long     samples = 0;
            DateTime start   = DateTime.Now;

            while (run)
            {
                // Run at ~50 callbacks/second. This doesn't need to be super precise. In
                // practice, Thread.Sleep is going to be +/- 10s of ms, but we'll still deliver
                // the right number of samples on average.
                Thread.Sleep(20);
                double elapsed        = (DateTime.Now - start).TotalSeconds;
                int    needed_samples = (int)(Math.Round(elapsed * SampleRate) - samples);
                callback(needed_samples, input, output, SampleRate);
                samples += needed_samples;
            }
        }
コード例 #11
0
ファイル: NullStream.cs プロジェクト: alexbv16/LiveSPICE
        private void Proc()
        {
            // Send 60 chunks/second. This code won't be perfectly accurate if 60 doesn't divide SampleRate.
            int count = (int)(SampleRate / 60);
            Audio.SampleBuffer[] input = new Audio.SampleBuffer[] { };
            Audio.SampleBuffer[] output = new Audio.SampleBuffer[] { };

            long t0 = Util.Timer.Counter;
            while (run)
            {
                long t1 = Util.Timer.Counter;
                if ((t1 - t0) / Util.Timer.Frequency > count / SampleRate)
                {
                    callback(count, input, output, SampleRate);
                    t0 = t1;
                }
                else
                {
                    Thread.Sleep(0);
                }
            }
        }
コード例 #12
0
        private void Proc()
        {
            // Send 60 chunks/second. This code won't be perfectly accurate if 60 doesn't divide SampleRate.
            int count = (int)(SampleRate / 60);

            Audio.SampleBuffer[] input  = new Audio.SampleBuffer[] { };
            Audio.SampleBuffer[] output = new Audio.SampleBuffer[] { };

            long t0 = Util.Timer.Counter;

            while (run)
            {
                long t1 = Util.Timer.Counter;
                if ((t1 - t0) / Util.Timer.Frequency > count / SampleRate)
                {
                    callback(count, input, output, SampleRate);
                    t0 = t1;
                }
                else
                {
                    Thread.Sleep(0);
                }
            }
        }
コード例 #13
0
ファイル: Stream.cs プロジェクト: yo1frenchtoast/LiveSPICE
        private void Proc()
        {
            Thread.CurrentThread.Name = "WaveAudio Stream";

            try
            {
                Log.Global.WriteLine(MessageType.Info, "Entering streaming thread");

                EventWaitHandle[] events = waveIn.Select(i => i.Callback).Concat(waveOut.Select(i => i.Callback)).ToArray();

                Audio.SampleBuffer[] input  = new Audio.SampleBuffer[waveIn.Length];
                Audio.SampleBuffer[] output = new Audio.SampleBuffer[waveOut.Length];
                while (!stop)
                {
                    // TODO: Why can't we use this?
                    //if (!WaitHandle.WaitAll(events, 100))
                    //    continue;

                    // Read from the inputs.
                    for (int i = 0; i < waveIn.Length; ++i)
                    {
                        InBuffer b = waveIn[i].GetBuffer();
                        if (b == null)
                        {
                            return;
                        }
                        using (Audio.RawLock l = new Audio.RawLock(b.Samples, false, true))
                            ConvertSamples(b.Data, format, l, l.Count);
                        b.Record();
                        input[i] = b.Samples;
                    }

                    // Get an available buffer from the outputs.
                    for (int i = 0; i < waveOut.Length; ++i)
                    {
                        OutBuffer b = waveOut[i].GetBuffer();
                        if (b == null)
                        {
                            return;
                        }
                        output[i] = b.Samples;
                    }

                    // Call the callback.
                    callback(buffer, input, output, format.nSamplesPerSec);

                    // Play the results.
                    for (int i = 0; i < output.Length; ++i)
                    {
                        OutBuffer b = (OutBuffer)output[i].Tag;
                        using (Audio.RawLock l = new Audio.RawLock(b.Samples, true, false))
                            ConvertSamples(l, b.Data, format, l.Count);
                        b.Play();
                    }
                }
            }
            catch (Exception Ex)
            {
                Log.Global.WriteLine(MessageType.Error, "Unhandled exception on streaming thread '{0}': {1}", Ex.GetType().FullName, Ex.ToString());
            }
            Log.Global.WriteLine(MessageType.Info, "Exiting streaming thread");
        }
コード例 #14
0
 public Buffer(ASIOBufferInfo Info, ASIOSampleType Type, int Count)
 {
     info    = Info;
     type    = Type;
     samples = new Audio.SampleBuffer(Count);
 }
コード例 #15
0
ファイル: Stream.cs プロジェクト: alexbv16/LiveSPICE
 public Buffer(ASIOBufferInfo Info, ASIOSampleType Type, int Count)
 {
     info = Info;
     type = Type;
     samples = new Audio.SampleBuffer(Count);
 }
コード例 #16
0
ファイル: Stream.cs プロジェクト: mikeoliphant/LiveSPICE
        private void Proc()
        {
            Thread.CurrentThread.Name = "WaveAudio Stream";

            try
            {
                Log.Global.WriteLine(MessageType.Info, "Entering streaming thread");

                Audio.SampleBuffer[] input  = new Audio.SampleBuffer[waveIn.Length];
                Audio.SampleBuffer[] output = new Audio.SampleBuffer[waveOut.Length];
                while (!stop)
                {
                    // Read from the inputs.
                    for (int i = 0; i < waveIn.Length; ++i)
                    {
                        InBuffer b = null;
                        do
                        {
                            b = waveIn[i].GetBuffer();
                        } while (b == null && !stop);
                        if (b != null)
                        {
                            ConvertSamples(b.Data, format, b.Samples.Raw, b.Samples.Count);
                            b.Record();
                            input[i] = b.Samples;
                        }
                    }

                    // Get an available buffer from the outputs.
                    for (int i = 0; i < waveOut.Length; ++i)
                    {
                        OutBuffer b = null;
                        do
                        {
                            b = waveOut[i].GetBuffer();
                        } while (b == null && !stop);
                        if (b != null)
                        {
                            output[i] = b.Samples;
                        }
                    }

                    if (!stop)
                    {
                        Debug.Assert(input.All(i => i != null));
                        Debug.Assert(output.All(i => i != null));

                        // Call the callback.
                        callback(buffer, input, output, format.nSamplesPerSec);

                        // Play the results.
                        for (int i = 0; i < output.Length; ++i)
                        {
                            OutBuffer b = (OutBuffer)output[i].Tag;
                            ConvertSamples(b.Samples.Raw, b.Data, format, b.Samples.Count);
                            b.Play();
                        }
                    }
                }
            }
            catch (Exception Ex)
            {
                Log.Global.WriteLine(MessageType.Error, "Unhandled exception on streaming thread '{0}': {1}", Ex.GetType().FullName, Ex.ToString());
            }
            Log.Global.WriteLine(MessageType.Info, "Exiting streaming thread");
        }
コード例 #17
0
ファイル: Stream.cs プロジェクト: alexbv16/LiveSPICE
        private void OnBufferSwitch(int Index, ASIOBool Direct)
        {
            Audio.SampleBuffer[] a = new Audio.SampleBuffer[input.Length];
            for (int i = 0; i < input.Length; ++i)
            {
                a[i] = input[i].Samples;

                using (Audio.RawLock l = new Audio.RawLock(a[i], false, true))
                    ConvertSamples(
                        input[i].Info.buffers[Index],
                        input[i].Type,
                        l,
                        l.Count);
            }

            Audio.SampleBuffer[] b = new Audio.SampleBuffer[output.Length];
            for (int i = 0; i < output.Length; ++i)
                b[i] = output[i].Samples;

            callback(buffer, a, b, sampleRate);

            for (int i = 0; i < output.Length; ++i)
            {
                using (Audio.RawLock l = new Audio.RawLock(b[i], true, false))
                    ConvertSamples(
                        l,
                        output[i].Info.buffers[Index],
                        output[i].Type,
                        l.Count);
            }
        }