Im trying to Make a radio Like Auto Dj to Play List Of Mp3 Files in series Like What Happen In Radio.
I tried a lot of work around but finally i thought of sending mp3 files to shoutcast server and play the output of that server my problem is i don't how to do that
i have tried bass.radio to use bass.net and that's my code
private int _recHandle;
private BroadCast _broadCast;
EncoderLAME l;
IStreamingServer server = null;
// Init Bass
Bass.BASS_Init(-1, 44100, BASSInit.BASS_DEVICE_DEFAULT,IntPtr.Zero);
// create the stream
int _stream = Bass.BASS_StreamCreateFile("1.mp3", 0, 0,
BASSFlag.BASS_SAMPLE_FLOAT | BASSFlag.BASS_STREAM_PRESCAN);
l= new EncoderLAME(_stream);
l.InputFile = null; //STDIN
l.OutputFile = null;
l.Start(null, IntPtr.Zero, false);
// decode the stream (if not using a decoding channel, simply call "Bass.BASS_ChannelPlay" here)
byte[] encBuffer = new byte[65536]; // our dummy encoder buffer
while (Bass.BASS_ChannelIsActive(_stream) == BASSActive.BASS_ACTIVE_PLAYING)
{
// getting sample data will automatically feed the encoder
int len = Bass.BASS_ChannelGetData(_stream, encBuffer, encBuffer.Length);
}
//l.Stop(); // finish
//Bass.BASS_StreamFree(_stream);
//Server
SHOUTcast shoutcast = new SHOUTcast(l);
shoutcast.ServerAddress = "50.22.219.37";
shoutcast.ServerPort = 12904;
shoutcast.Password = "01008209907";
shoutcast.PublicFlag = true;
shoutcast.Genre = "Hörspiel";
shoutcast.StationName = "Kravis Server";
shoutcast.Url = "";
shoutcast.Aim = "";
shoutcast.Icq = "";
shoutcast.Irc = "";
server = shoutcast;
server.SongTitle = "BASS.NET";
// disconnect, if connected
if (_broadCast != null && _broadCast.IsConnected)
{
_broadCast.Disconnect();
}
_broadCast = null;
GC.Collect();
_broadCast = new BroadCast(server);
_broadCast.Notification += OnBroadCast_Notification;
_broadCast.AutoReconnect = true;
_broadCast.ReconnectTimeout = 5;
_broadCast.AutoConnect();
but i don't get my File Streamed to streamed to the server even the _broadCast Is Connected.
so if any solution of code or any other thing i can do.
I haven't used BASS in many years, so I can't give you specific advice on the code you have there. But, I wanted to give you the gist of the process of what you need to do... it might help you get started.
As your file is in MP3, it is possible to send it directly to the server and hear it on the receiving end. However, there are a few problems with that. The first is rate control. If you simply transmit the file data, you'll send say 5 minutes of data in perhaps a 10 second time period. This will eventually cause failures as the clients aren't going to buffer much data, and they will disconnect. Another problem is that your MP3 files often have extra data in them in the form of ID3 tags. Some players will ignore this, others won't. Finally, some of your files might be in different sample rates than others, so even if you rate limit your sending, the players will break when they hit a file in a different sample rate.
What needs to happen is the generation of a fresh stream. The pipeline looks something like this:
[Source File] -> [Codec] -> [Raw PCM Audio] -> [Codec] -> [MP3 Stream] -> [SHOUTcast Server] -> [Clients]
Additionally, that raw PCM audio step needs to run in at a realtime rate. While your computer can definitely decode and encode faster than realtime, it needs to be ran at realtime so that the players can listen in realtime.
Related
I am building a recording software for recording all connected devices to PC into mp3 format.
Here is my code:
IWaveIn _captureInstance = inputDevice.DataFlow == DataFlow.Render ?
new WasapiLoopbackCapture(inputDevice) : new WasapiCapture(inputDevice)
var waveFormatToUse = _captureInstance.WaveFormat;
var sampleRateToUse = waveFormatToUse.SampleRate;
var channelsToUse = waveFormatToUse.Channels;
if (sampleRateToUse > 48000) // LameMP3FileWriter doesn't support a rate more than 48000Hz
{
sampleRateToUse = 48000;
}
else if (sampleRateToUse < 8000) // LameMP3FileWriter doesn't support a rate less than 8000Hz
{
sampleRateToUse = 8000;
}
if (channelsToUse > 2) // LameMP3FileWriter doesn't support a number of channels more than 2
{
channelsToUse = 2;
}
waveFormatToUse = WaveFormat.CreateCustomFormat(_captureInstance.WaveFormat.Encoding,
sampleRateToUse,
channelsToUse,
_captureInstance.WaveFormat.AverageBytesPerSecond,
_captureInstance.WaveFormat.BlockAlign,
_captureInstance.WaveFormat.BitsPerSample);
_mp3FileWriter = new LameMP3FileWriter(_currentStream, waveFormatToUse, 32);
This code works properly, except the cases when a connected device (also virtual as SteelSeries Sonar) has more than 2 channels.
In the case with more than 2 channels all recordings with noise only.
How can I solve this issue? It isn't required to use only LameMP3FileWriter, I only need it will mp3 or any format with good compression. Also if it's possible without saving intermediate files on the disk (all processing in memory), only the final file with audio.
My recording code:
// When the capturer receives audio, start writing the buffer into the mentioned file
_captureInstance.DataAvailable += (s, a) =>
{
lock (_writerLock)
{
// Write buffer into the file of the writer instance
_mp3FileWriter?.Write(a.Buffer, 0, a.BytesRecorded);
}
};
// When the Capturer Stops, dispose instances of the capturer and writer
_captureInstance.RecordingStopped += (s, a) =>
{
lock (_writerLock)
{
_mp3FileWriter?.Dispose();
}
_captureInstance?.Dispose();
};
// Start audio recording
_captureInstance.StartRecording();
If LAME doesn't support more than 2 channels, you can't use this encoder for your purpose. Have you tried it with the Fraunhofer surround MP3 encoder?
Link: https://download.cnet.com/mp3-surround-encoder/3000-2140_4-165541.html
Also, here's a nice article discussing how to convert between most audio formats (with C# code samples): https://www.codeproject.com/articles/501521/how-to-convert-between-most-audio-formats-in-net
I'm trying to use NAudio to play shoutcast streams. This is doable (apparently) with a few lines of code:
var url = "http://dance.radiomonster.fm/320.mp3";
using(var radioStream = new MediaFoundationReader(url))
using(var wo = new WaveOutEvent())
{
wo.Init(radioStream);
wo.Play();
while (wo.PlaybackState == PlaybackState.Playing)
{
Thread.Sleep(1000);
}
}
Works fine for playback. However I need to take that as float samples converted to 48 KHz mono to be sent off as a bytestream elsewhere. So I do this:
//Convert the wave to 48 khz, mono
str1 = new WaveFormatConversionProvider(new WaveFormat(48000,1), radioStream);
//Convert the converted wave to floats
str = new Wave16ToFloatProvider(str1);
Under previous iterations of this I would do something like this to retrieve the data:
while ((readResultCount = str.Read(result, 0, result.Length)) > 0)
{
float[] output = new float[result.Length / sizeof(float)];
Buffer.BlockCopy(result, 0, output, 0, sizeof(float) * output.Length);
au.EnqueueEncodeVoiceData(output);
}
However, this results in audio output that skips worse than a CD that's been ran over.
I've tried to see if there was any way to determine if there was data enough in the buffer and avoid reading the stream if not, but there doesn't seem to be any facilities to do this.
What am I doing wrong?
Ah, here's a matter of hidden knowledge.
The loop itself was being called in a timer event tick, which meant that multiple instances of the loop could (and would) get called, since the Read operation is apparently blocking until either the requested data is received or the stream ends for whatever reason.
Since the timer ticked indiscriminately of whether the loop had exited or not, it was running over its own toes.
I resolved this by removing the timer tick and shoving the routine into a Task.
//Declare string for application temp path and tack on the file extension
string fileName = string.Format ("Myfile{0}.wav", DateTime.Now.ToString ("yyyyMMddHHmmss"));
string audioFilePath = Path.Combine (Path.GetTempPath (), fileName);
Console.WriteLine("Audio File Path: " + audioFilePath);
url = NSUrl.FromFilename(audioFilePath);
//set up the NSObject Array of values that will be combined with the keys to make the NSDictionary
NSObject[] values = new NSObject[]
{
NSNumber.FromFloat (44100.0f), //Sample Rate
NSNumber.FromInt32 ((int)AudioToolbox.AudioFormatType.LinearPCM), //AVFormat
NSNumber.FromInt32 (2), //Channels
NSNumber.FromInt32 (16), //PCMBitDepth
NSNumber.FromBoolean (false), //IsBigEndianKey
NSNumber.FromBoolean (false) //IsFloatKey
};
//Set up the NSObject Array of keys that will be combined with the values to make the NSDictionary
NSObject[] keys = new NSObject[]
{
AVAudioSettings.AVSampleRateKey,
AVAudioSettings.AVFormatIDKey,
AVAudioSettings.AVNumberOfChannelsKey,
AVAudioSettings.AVLinearPCMBitDepthKey,
AVAudioSettings.AVLinearPCMIsBigEndianKey,
AVAudioSettings.AVLinearPCMIsFloatKey
};
//Set Settings with the Values and Keys to create the NSDictionary
settings = NSDictionary.FromObjectsAndKeys (values, keys);
//Set recorder parameters
recorder = AVAudioRecorder.Create(url, new AudioSettings(settings), out error);
//Set Recorder to Prepare To Record
recorder.PrepareToRecord();
This code works well, but how can you keep a record from a microphone directly to stream?
I did not find any information in the Internet, i hope you can help me
You are looking for buffered access to the audio stream (recording or playback), iOS provides it via Audio Queue Services (AVAudioRecorder is too high level), so then as audio buffers are filled, iOS calls your callback with a filled buffer from the queue, you do something with it (save it to disk, write it to a C#-based Stream, send to a playback audio queue [speakers], etc...) and, normally, place it back into the queue for reuse.
Something like this starts recording to queue of audio buffers:
var recordFormat = new AudioStreamBasicDescription() {
SampleRate = 8000,
Format = AudioFormatType.LinearPCM,
FormatFlags = AudioFormatFlags.LinearPCMIsSignedInteger | AudioFormatFlags.LinearPCMIsPacked,
FramesPerPacket = 1,
ChannelsPerFrame = 1,
BitsPerChannel = 16,
BytesPerPacket = 2,
BytesPerFrame = 2,
Reserved = 0
};
recorder = new InputAudioQueue (recordFormat);
for (int count = 0; count < BufferCount; count++) {
IntPtr bufferPointer;
recorder.AllocateBuffer(AudioBufferSize, out bufferPointer);
recorder.EnqueueBuffer(bufferPointer, AudioBufferSize, null);
}
recorder.InputCompleted += HandleInputCompleted;
recorder.Start ();
So assuming a AudioBufferSize of 8k and a BufferCount of 3 in this example, so once the first of three buffers is filled, our handler HandleInputCompleted is called (since there are 2 buffers still in the queue recording continues to them.
Our InputCompleted handler:
private void HandleInputCompleted (object sender, InputCompletedEventArgs e)
{
// We received a new buffer of audio, do something with it....
// Some unsafe code will be required to rip the buffer...
// Place the buffer back into the queue so iOS knows you are done with it
recorder.EnqueueBuffer(e.IntPtrBuffer, AudioBufferSize, null);
// At some point you need to call `recorder.Stop();` ;-)
}
(I ripped out our code from the handler as it is a custom audio-2-text learning neutral network as we use really small buffers in a very large queue to reduce feedback latency and load that audio data within single TCP/UDP packets for cloud processing (think Siri ;-)
In this handler you have access to the Pointer to the buffer that is currently filled via InputCompletedEventArgs.IntPtrBuffer, Using that pointer you could peek each byte in the buffer and poke them to your C#-based Stream if that is your goal.
Apple has a great tech article concerning Audio Queue: https://developer.apple.com/library/ios/documentation/MusicAudio/Conceptual/AudioQueueProgrammingGuide/AboutAudioQueues/AboutAudioQueues.html
Im currently a bit stuck with my c# project.
I have 2 applications, they both have a common class definition I call a NetMessage
a NetMessage contains a MessageType string property, as well as 2 List lists.
The idea is that I can pack this class with classes, and data to send across the network as a byte[].
Because Network Streams do not advertise the amount of data they are receiving, I modified my Send method to send the size of the NetMessage byte[] ahead of the actual byte[].
private static byte[] ReceivedBytes(NetworkStream MainStream)
{
try
{
//byte[] myReadBuffer = new byte[1024];
int receivedDataLength = 0;
byte[] data = { };
long len = 0;
int i = 0;
MainStream.ReadTimeout = 60000;
//MainStream.CanTimeout = false;
if (MainStream.CanRead)
{
//Read the length of the incoming message
byte[] byteLen = new byte[8];
MainStream.Read(byteLen, 0, 8);
len = BitConverter.ToInt64(byteLen, 0);
data = new byte[len];
//data is now set to the appropriate size for the expected message
//While we have not got the full message
//Read each individual byte and append to data.
//This method, seems to work, but is ridiculously slow,
while (receivedDataLength < data.Length)
{
receivedDataLength += MainStream.Read(data, receivedDataLength, 1);
}
//receivedDataLength += MainStream.Read(data, receivedDataLength, data.Length);
return data;
}
}
catch (Exception E)
{
//System.Windows.Forms.MessageBox.Show("Exception:" + E.ToString());
}
return null;
}
I have tried to change the size argument below to something like 1024 or to be the data.Length, but I get funky results.
receivedDataLength += MainStream.Read(data, receivedDataLength, 1);
setting it to data.Length seems to cause problems when the Class being sent is a few mb in size.
Setting the buffer size to be 1024 like I have seen in other examples, causes failures when the size of the incoming message is small, like 843 bytes, it errors out saying that I tried to read out of bounds or something.
Below is the type of method being used to send the data in the first place.
public static void SendBytesToStream(NetworkStream TheStream, byte[] TheMessage)
{
//IAsyncResult r = TheStream.BeginWrite(TheMessage, 0, TheMessage.Length, null, null);
// r.AsyncWaitHandle.WaitOne(10000);
//TheStream.EndWrite(r);
try
{
long len = TheMessage.Length;
byte[] Bytelen = BitConverter.GetBytes(len);
TheStream.Write(Bytelen, 0, Bytelen.Length);
TheStream.Flush();
// <-- I've tried putting thread sleeps in this spot to see if it helps
//I've also tried writing each byte of the message individually
//takes longer, but seems more accurate as far as network transmission goes?
TheStream.Write(TheMessage, 0, TheMessage.Length);
TheStream.Flush();
}
catch (Exception e)
{
//System.Windows.Forms.MessageBox.Show(e.ToString());
}
}
I'd like to get these two methods setup to the point where they are reliably sending and receiving data.
The application I am using this for, monitors a screenshots folder in a game directory,
when it detects a screenshot in TGA format, it converts it to PNG, then takes its byte[] and sends it to the receiver.
The receiver then posts it to Facebook (I don't want my FB tokens distributed in my client application), hence the server / proxy idea.
Its strange, but when I step through the code, the transfer is invariably successful.
But if I run it full speed, no breakpoint, it typically tells me that the connection was closed by the remote host etc.
The client typically finishes sending the data almost instantly, even though its a 4mb file.
The receiver spends about 2 minutes reading from the Network Stream, which doesnt make sense, if the client finished sending the data, does that mean the data is just floating in cyber space, and being pulled down?
Surely it should be synchronous?
I suspect I know where my code was going wrong.
It turns out that the scope I was creating my TCPClient that was doing the sending, was declared and instantiated within a method.
This being the case, the GAC was disposing of it, even though the Receiving Server had not finished downloading the stream.
I managed to resolve it by creating a method that can detect when the Client has disconnected on the server end, and until it has actually disconnected, it will keep looping/waiting until disconnected.
This way, we are waiting until the server lets go of us.
In my application I can download some media files from web. Normally I used WebClient.OpenReadCompleted method to download, decrypt and save the file to IsolatedStorage. It worked well and looked like that:
private void downloadedSong_OpenReadCompleted(object sender, OpenReadCompletedEventArgs e, SomeOtherValues someOtherValues) // delegate, uses additional values
{
// Some preparations
try
{
if (e.Result != null)
{
using (isolatedStorageFile = IsolatedStorageFile.GetUserStoreForApplication())
{
// working with the gained stream, decryption
// saving the decrypted file to isolatedStorage
isolatedStorageFileStream = new IsolatedStorageFileStream("SomeFileNameHere", FileMode.OpenOrCreate, isolatedStorageFile);
// and use it for MediaElement
mediaElement.SetSource(isolatedStorageFileStream);
mediaElement.Position = new TimeSpan(0);
mediaElement.MediaOpened += new RoutedEventHandler(mediaFile_MediaOpened);
// and some other work
}
}
}
catch(Exception ex)
{
// try/catch stuff
}
}
But after some investigation I found out that with large files(for me it's more than 100 MB) I'm getting OutOfMemory exception during downloading this file. I suppose that's because WebClient.OpenReadCompleted loads the whole stream into RAM and chokes... And I will need more memory to decrypt this stream.
After another investigation, I've found how to divide large file into chunks after OpenReadCompleted event at saving this file to IsolatedStorage(or decryption and then saving in my ocasion), but this would help with only a part of problem... The primary problem is how to prevent phone chokes during download process. Is there a way to download large file in chunks? Then I could use the found solution to pass through decryption process. (and still I'd need to find a way to load such big file into mediaElement, but that would be another question)
Answer:
private WebHeaderCollection headers;
private int iterator = 0;
private int delta = 1048576;
private string savedFile = "testFile.mp3";
// some preparations
// Start downloading first piece
using (IsolatedStorageFile isolatedStorageFile = IsolatedStorageFile.GetUserStoreForApplication())
{
if (isolatedStorageFile.FileExists(savedFile))
isolatedStorageFile.DeleteFile(savedFile);
}
headers = new WebHeaderCollection();
headers[HttpRequestHeader.Range] = "bytes=" + iterator.ToString() + '-' + (iterator + delta).ToString();
webClientReadCompleted = new WebClient();
webClientReadCompleted.Headers = headers;
webClientReadCompleted.OpenReadCompleted += downloadedSong_OpenReadCompleted;
webClientReadCompleted.OpenReadAsync(new Uri(song.Link));
// song.Link was given earlier
private void downloadedSong_OpenReadCompleted(object sender, OpenReadCompletedEventArgs e)
{
try
{
if (e.Cancelled == false)
{
if (e.Result != null)
{
((WebClient)sender).OpenReadCompleted -= downloadedSong_OpenReadCompleted;
using (IsolatedStorageFile myIsolatedStorage = IsolatedStorageFile.GetUserStoreForApplication())
{
using (IsolatedStorageFileStream fileStream = new IsolatedStorageFileStream(savedFile, FileMode.Append, FileAccess.Write, myIsolatedStorage))
{
int mediaFileLength = (int)e.Result.Length;
byte[] byteFile = new byte[mediaFileLength];
e.Result.Read(byteFile, 0, byteFile.Length);
fileStream.Write(byteFile, 0, byteFile.Length);
// If there's something left, download it recursively
if (byteFile.Length > delta)
{
iterator = iterator + delta + 1;
headers = new WebHeaderCollection();
headers[HttpRequestHeader.Range] = "bytes=" + iterator.ToString() + '-' + (iterator + delta).ToString();
webClientReadCompleted.Headers = headers;
webClientReadCompleted.OpenReadCompleted += downloadedSong_OpenReadCompleted;
webClientReadCompleted.OpenReadAsync(new Uri(song.Link));
}
}
}
}
}
}
To download a file in chunks you'll need to make multiple requests. One for each chunk.
Unfortunately it's not possible to say "get me this file and return it in chunks of size X";
Assuming that the server supports it, you can use the HTTP Range header to specify which bytes of a file the server should return in response to a request.
You then make multiple requests to get the file in pieces and then put it all back together on the device. You'll probably find it simplest to make sequential calls and start the next one once you've got and verified the previous chunk.
This approach makes it simple to resume a download when the user returns to the app. You just look at how much was downloaded previously and then get the next chunk.
I've written an app which downloads movies (up to 2.6GB) in 64K chunks and then played them back from IsolatedStorage with the MediaPlayerLauncher. Playing via the MediaElement should work too but I haven't verified. You can test this by loading a large file directly into IsolatedStorage (via Isolated Storage Explorer, or similar) and check the memory implications of playing that way.
Confirmed: You can use BackgroundTransferRequest to download multi-GB files but you must set TransferPreferences to None to force the download to happen while connected to an external power supply and while connected to wi-fi, else the BackgroundTransferRequest will fail.
I wonder if it's possible to use a BackgroundTransferRequest to download large files easily and let the phone worry about the implementation details? The documentation seems to suggest that file downloads over 100 MB are possible, and the "Range" verb is reserved for it's own use, so it probably uses this automatically if it can behind the scenes.
From the documentation regarding files over 100 MB:
For files larger than 100 MB, you must set the TransferPreferences
property of the transfer to None or the transfer will fail. If you do
not know the size of a transfer and it is possible that it could
exceed this limit, you should set the value to None, meaning that the
transfer will only proceed when the phone is connected to external
power and has a Wi-Fi connection.
From the documentation regarding use of the "Range" verb:
The Headers property of the BackgroundTransferRequest object is used
to set the HTTP headers for a transfer request. The following headers
are reserved for use by the system and cannot be used by calling
applications. Adding one of the following headers to the Headers
collection will cause a NotSupportedException to be thrown when the
Add(BackgroundTransferRequest) method is used to queue the transfer
request:
If-Modified-Since
If-None-Match
If-Range
Range
Unless-Modified-Since
Here's the documentation:
http://msdn.microsoft.com/en-us/library/windowsphone/develop/hh202955(v=vs.105).aspx