I'm currently getting a float array using directsound to record audio.
Now I would like to play that float array using XAudio2 (SlimDX also), but I'm not sure what to do since the sample example from SlimDX plays a .wav file.
here is how they do this:
XAudio2 device = new XAudio2();
MasteringVoice masteringVoice = new MasteringVoice(device);
var s = System.IO.File.OpenRead(fileName);
WaveStream stream = new WaveStream(s);
s.Close();
AudioBuffer buffer = new AudioBuffer();
buffer.AudioData = stream;
buffer.AudioBytes = (int)stream.Length;
buffer.Flags = BufferFlags.EndOfStream;
SourceVoice sourceVoice = new SourceVoice(device, stream.Format);
sourceVoice.SubmitSourceBuffer(buffer);
sourceVoice.Start();
// loop until the sound is done playing
while (sourceVoice.State.BuffersQueued > 0)
{
if (GetAsyncKeyState(VK_ESCAPE) != 0)
break;
Thread.Sleep(10);
}
// wait until the escape key is released
while (GetAsyncKeyState(VK_ESCAPE) != 0)
Thread.Sleep(10);
// cleanup the voice
buffer.Dispose();
sourceVoice.Dispose();
stream.Dispose();
Basically, what I would like to know is how to play a float array using slimDX?
Thanks in advance
I'm not an expert on audio stuff, but I do know that you can create a WaveFormat of IeeeFloat. Fill in all the other information, and then write your data to a DataStream and give that to the AudioBuffer. Then you can call Submit as normal.
Related
I'm using NAudio to output audio files to both the speaker and a headset on a window 10 laptop. I created two WaveOut and assigned the corresponding device number. But I cannot here the audio from the speaker when the headset is plugged in. Can anyone let me know how to solve this? Here's my code (it works fine on the headset or the speaker separately, but I want to hear the sound from both of them at the same time):
var input1 = new Mp3FileReader(PATH + "left.mp3");
var input2 = new Mp3FileReader(PATH + "right.mp3");
var waveProvider = new MultiplexingWaveProvider(new IWaveProvider[] { input1, input2 }, 2);
var input3 = new Mp3FileReader(PATH + "left.mp3");
int channel = ((Mp3FileReader)input1).Mp3WaveFormat.Channels;
Debug.WriteLine(channel);
waveProvider.ConnectInputToOutput(0, 0);
waveProvider.ConnectInputToOutput(3, 1);
WaveOut wave = new WaveOut();
wave.DeviceNumber = 1;
wave.Init(waveProvider);
WaveOut wave1 = new WaveOut();
wave1.DeviceNumber = 0;
wave1.Init(input3);
wave.Play();
wave1.Play();
I think the issue is that you don't have two soundcards, you have one soundcard that is switching between playing sound out of the speakers and headphones. If you bought a USB headset then you'd have two soundcards, and should be able to play different sounds through each one separately.
This post is also posted on The Amazing Audio Engine forum.
Hi everyone, I am new to The Amazing Audio Engine and iOS dev, and have been trying to figure out how to get the BPM of a track.
So far I have found two articles on offline rendering on the forum:
http://forum.theamazingaudioengine.com/discussion/comment/1743/#Comment_1743
http://forum.theamazingaudioengine.com/discussion/comment/649#Comment_649
As far as I know the AEAudioControllerRenderMainOutput function is only correctly implemented in this fork.
I am trying to do offline rendering to process a track and then use the algorithm described here (JavaScript) and implemented here.
So far I'm loading this fork, and I am using Swift (I am part of Make School Summer Academy at the moment, which teaches Swift).
When playing a track this code works for me (No offline rendering!)
let file = NSBundle.mainBundle().URLForResource("track", withExtension:
"m4a")
let channel: AnyObject! = AEAudioFilePlayer.audioFilePlayerWithURL(file, audioController: audioController, error: nil)
audioController = AEAudioController(audioDescription: AEAudioController.nonInterleavedFloatStereoAudioDescription())
let receiver = AEBlockAudioReceiver { (source, time, frames, audioBufferList) -> Void in
let leftSamples = UnsafeMutablePointer<Float>(audioBufferList[0].mBuffers.mData)
// Advance the buffer sizeof(float) * 512
let rightSamples = UnsafeMutablePointer<Float>(audioBufferList[0].mBuffers.mData) + 512
println("leftSamples: \(leftSamples) rightSamples: \(rightSamples)")
}
audioController.addChannels([channel])
audioController.addOutputReceiver(receiver)
audioController.start()
Trying offline rendering
This is the code I am trying to run while I am using this fork
audioController = AEAudioController(audioDescription: AEAudioController.nonInterleaved16BitStereoAudioDescription())
let file = NSBundle.mainBundle().URLForResource("track", withExtension: "mp3")
let channel: AnyObject! = AEAudioFilePlayer.audioFilePlayerWithURL(file, audioController: audioController, error: nil)
audioController.addChannels([channel])
audioController.start(nil)
audioController.stop()
var t = AudioTimeStamp()
let bufferLength: UInt32 = 4096
var buffer = AEAllocateAndInitAudioBufferList(audioController.audioDescription, Int32(bufferLength))
AEAudioControllerRenderMainOutput(audioController, t, bufferLength, buffer)
var renderDuration: NSTimeInterval = channel.duration
var sampleRate: Float64 = audioController.audioDescription.mSampleRate
var lengthInFrames: UInt32 = UInt32(renderDuration * sampleRate)
var songBuffer: [Float64]
t.mFlags = UInt32(kAudioTimeStampSampleTimeValid)
var frequencyAnalyzer = FrequencyAnalyzer()
println("renderDuration \(renderDuration)")
var outIsOpen = Boolean()
AUGraphClose(audioController.audioGraph)
AUGraphIsOpen(audioController.audioGraph, &outIsOpen)
println("AUGraphIsOpen: \(outIsOpen)")
for (var i: UInt32 = 0; i < lengthInFrames; i += bufferLength) {
AEAudioControllerRenderMainOutput(audioController, t, bufferLength, buffer);
t.mSampleTime += Float64(bufferLength)
println(t.mSampleTime)
let leftSamples = UnsafeMutablePointer<Int16>(buffer[0].mBuffers.mData)
let rightSamples = UnsafeMutablePointer<Int16>(buffer[0].mBuffers.mData) + 512
println("leftSamples: \(leftSamples.memory) rightSamples: \(rightSamples.memory)")
}
AEFreeAudioBufferList(buffer)
AUGraphOpen(audioController.audioGraph)
audioController.start(nil)
audioController.stop()
Offline rendering is not working for me ATM. The second example is not working it's getting me a lot of mixed errors which I don't understand.
A very common one is inside the channelAudioProducer function on this line:
// Tell mixer/mixer's converter unit to render into audio status = AudioUnitRender(group->converterUnit ? group->converterUnit : group->mixerAudioUnit, arg->ioActionFlags, &arg->originalTimeStamp, 0, *frames, audio);
It gives me EXC_BAD_ACCESS (code=EXC_I386_GPFLT). Among other errors this one is very common.
I am sorry I am a total noob on this field but some stuff I don't really understand. Should I use nonInterleaved16BitStereoAudioDescription or nonInterleavedFloatStereoAudioDescription? How does this implement the mData?
I would love to get some help on this since I'm kind of lost at the moment. Please when you answer me try to explain it as fully as you can, I am new to this stuff.
NOTE: Posting code in Objective-C is fine if you don't know Swift.
I'm starting to write a Windows Service that will convert G.722 audio files into WAV files and I'm planning on using the NAudio library.
After looking at the NAudio demos, I've found that I will need to use the G722Codec to decode the audio data from the file but I'm having trouble figuring out how to read the G722 file. Which reader should I use?\
The G722 files are 7 kHz.
I'm working my way through the Pluralsight course for NAudio but it would be great to get a small code sample.
I got it working with the RawSourceWaveStreambut then tried to simply read the bytes of the file, decode using the G722 Codec and write the bytes out to a wave file. It worked.
private readonly G722CodecState _state = new G722CodecState(64000, G722Flags.SampleRate8000);
private readonly G722Codec _codec = new G722Codec();
private readonly WaveFormat _waveFormat = new WaveFormat(8000, 1);
public MainWindow()
{
InitializeComponent();
var data = File.ReadAllBytes(#"C:\Recordings\000-06Z_chunk00000.g722");
var output = Decode(data, 0, data.Length);
using (WaveFileWriter waveFileWriter = new WaveFileWriter(#"C:\Recordings\000-06Z_chunk00000.wav", _waveFormat))
{
waveFileWriter.Write(output, 0, output.Length);
}
}
private byte[] Decode(byte[] data, int offset, int length)
{
if (offset != 0)
{
throw new ArgumentException("G722 does not yet support non-zero offsets");
}
int decodedLength = length * 4;
var outputBuffer = new byte[decodedLength];
var wb = new WaveBuffer(outputBuffer);
int decoded = _codec.Decode(_state, wb.ShortBuffer, data, length);
return outputBuffer;
}
I have a game with a preloader in scene 1, with the following code on the time line.
stop();
loadingBar._xscale = 1;
var loadingCall:Number = setInterval(preloadSite, 50);
function preloadSite():Void {
var siteLoaded:Number = _root.getBytesLoaded();
var siteTotal:Number = _root.getBytesTotal();
var percentage:Number = Math.round(siteLoaded/siteTotal*100);
loadingBar._xscale = percentage;
bytesDisplay.text = percentage + "%";
if (siteLoaded >= siteTotal) {
clearInterval(loadingCall);
gotoAndPlay("StartMenu", 1);
}
}
The code works fine when there are no music files linked to frame 1. If there are music files linked, then everything loads before the preloader shows up.
I found this great webpage about preloaders, which speaks about the linkage issue, and suggests I put all the big files on frame 2, after the preloader, then skip them. I put my large files on frame 2 as suggested and the preloader worked again.
My question is, is there a better way to do this. This solution seems like a hack.
The only better option I can think of, is to NOT store the MP3 file in your Flash file, but rather load it in your preloader with your flash file's content. This is provided that you're storing your MP3 file somewhere else online (like on a server).
stop();
loadingBar._xscale = 1;
var sound:Sound = new Sound();
sound.loadSound("http://www.example.com/sound.mp3", false);
var loadingCall:Number = setInterval(preloadSite, 50);
function preloadSite():Void {
var siteLoaded:Number = _root.getBytesLoaded()+sound.getBytesLoaded();
var siteTotal:Number = _root.getBytesTotal()+sound.getBytesTotal();
var percentage:Number = Math.round(siteLoaded / siteTotal * 100);
loadingBar._xscale = percentage;
bytesDisplay.text = percentage + "%";
if (siteLoaded >= siteTotal) {
clearInterval(loadingCall);
gotoAndPlay("StartMenu", 1);
sound.start();
}
}
The app saves the camera output into a mov. file, then turn it to flv format that sent by AVPacket to rtmp server.
It switch every time between two files, one is written by the camera output and the other one is sent.
My problem is that the audio/video is getting out of sync after a while.
The first buffer sent is allways 100% sync but after awhile it get messed.
I belive its a DTS-PTS problem..
if(isVideo)
{
packet->stream_index = VIDEO_STREAM;
packet->dts = packet->pts = videoPosition;
videoPosition += packet->duration = FLV_TIMEBASE * packet->duration * videoCodec->ticks_per_frame * videoCodec->time_base.num / videoCodec->time_base.den;
}
else
{
packet->stream_index = AUDIO_STREAM;
packet->dts = packet->pts = audioPosition;
audioPosition += packet->duration = FLV_TIMEBASE * packet->duration / audioRate;
//NSLog(#"audio position = %lld", audioPosition);
}
packet->pos = -1;
packet->convergence_duration = AV_NOPTS_VALUE;
// This sometimes fails without being a critical error, so no exception is raised
if((code = av_interleaved_write_frame(file, packet)))
{
NSLog(#"Streamer::Couldn't write frame");
}
av_free_packet(packet);
You can research this sample: http://unick-soft.ru/art/files/ffmpegEncoder-vs2008.zip
But this sample is for Windows.
In this sample I use pts only for audio stream:
if (pVideoCodec->coded_frame->pts != AV_NOPTS_VALUE)
{
pkt.pts = av_rescale_q(pVideoCodec->coded_frame->pts,
pVideoCodec->time_base, pVideoStream->time_base);
}
I was experiencing a similar issue when switching out the AVAssetWriters, and noticed that it went way if I only started using the new AVAssetWriter when I got a video sample
https://medium.com/#brandon.kobel/ios-seamless-video-chunks-4383a5a3a874