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  • Windows 7 does not recognize hdmi audio output

    - by user30038
    I recently upgraded to Windows 7 on a Dell 545 MT with audio/video output via nVidia 9800 GTX+ to a flat screen television. I have installed/uninstalled/reinstalled all the most recent drivers from nVidia & realtek, but the machine will not recognize the hdmi output as a sound device (as it did previously in Vista). I troubleshot the problem with Dell for over an hour and was then directed to their software specialists who want $130 to fix the issue! Can anyone offer some insights on this? I've searched nVidia and Windows 7 forums w/o success and would really like to get sound coming back through my tv speakers. Many Thanks.

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  • Digital audio does not work on MacBook Pro

    - by mathk
    I have a MacBook Pro (8,2). Using a TOSLINK cable I have no digital output. Oddly enough, sometime I can hear a glitch when I plug in the cable or when I give it a gentle wiggle. My guess is that the output is not correctly detecting that I have a digital link. So is there a way to force digital audio output on a MacBook Pro? Some say that in the Audio MIDI Setup there is an option but I can't find it. I am running OS X 10.7.5.

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  • Unity 5.1 audio issues (no sound in back channels)

    - by N0xus
    I've trying to bring in surround sound audio into my project. I've set my computer up to run in 5.1 and when I play a 6 channel audio through windows media player (it's a test audio that does left speaker, right speaker etc) it works fine. However, when I run it through Unity, all I get is the front 3 channels. I've set it in the Edit - project settings - audio to be 5.1 in there. I even set it in code with following: void Start() { AudioSettings.speakerMode = AudioSpeakerMode.Mode5point1; } How ever, when I run a debug line of: print ( AudioSettings.driverCaps); It tells me that Unity is only playing in stereo. Is there something I'm still not doing? I should also add I've ran 10 different tests using the 3D audio pan and spread options. I've set both to either being fully off, half way on and full. Still the same results.

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  • How do I record audio through M-Audio Keystudio?

    - by interstar
    Hi, I'm trying to get my M-Audio Keystudio (which has an audio input as well as the keyboard) to record audio to Audacity. I'm in Ubuntu 10.10. When I look at the Sound Preferences I can select "M-Audio RunTime DFU Analog Stereo" as my input device. However, when I try to record in Audacity, Audacity remains frozen. The program seems to be running and recording, but the recording cursor won't advance. If I reset the audio input to the internal sound card, recording works normally. Any ideas what to look for?

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  • Stream audio to mobile device

    - by blackn1ght
    I'd like to stream the audio from Ubuntu 10.10 to my HTC Desire HD (Android 2.2). I've seen solutions so far for streaming from audio players, but I'd like to stream any audio output from the PC to my phone. My use case is for watching TV/Films in VLC or online (BBC iPlayer) in bed, without having to use my surround sound system which is likely to wake up my house mates. I'm not just talking about music from Banshee, but any audio that the system makes. I was thinking that PulseAudio is pretty powerful, is it possible to route audio through that to a mobile device? Can it be done through bluetooth? Cheers in advance!

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  • USB Audio Device Loopback Through Speakers

    - by matto1990
    I have a USB turntable which when plugged in to my ubuntu 10.10 machine appears in the audio settings as an input device (USB PnP Audio Device Analog Stereo) like a microphone. What I'd like to be able to do it to have the sound for that audio device played back through the audio output (speaker or whatever). I'm not too worried if there's a slight delay between the audio coming in and it being played out through the speakers. As far as I'm aware this is refereed to as software loopback. I can achieve exactly what I want if I open Audacity, enable software loopback and press record. Obvious this isn't ideal as I don't really want it recording what I'm playing all the time. I know this is possible because of the Audacity example however I'd like to know if there's a way to do it without it recording. I've search around for a while for a piece of software that does this, however I couldn't get anything even close. Any help would be greatly appreciated.

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  • Play audio in javascript with a good performance

    - by João
    I'm developing a browser game where the player can shoot. Everytime he shoots it play a sound. Currently i'm using this code to play sounds in JavaScript: var audio = document.createElement("audio"); audio.src = "my_sound.mp3"; audio.play(); I'm worried about performance here. Will 10 simultaneous sounds impact my game performance too much? Will all audio objects stay in memory even after they are played?

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  • How can I select an audio output device in directshow

    - by Vibhore Tanwer
    I was wondering how I can select the output device for audio in directshow. I am able to get available audio output devices in directshow. But how can I make one of these to be audio output device. Its always going for the default audio device. I want to be able to output audio on my choice of device. I have been struggling through google but couldn't find anything useful. All I could get was this link but it doesn't really solve my problem. Any help will be really helpful for me.

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  • Diverting sound output of MCE to SPDIF

    - by Saxtus
    I have an ASUS SupremeFX II audio card (which in fact is an onboard audio riser slot) with the default drivers that are pre-installed by Windows 7 x64 for this card. I am able to manually switch between analog output and SPDIF output by the means of control panel (or external utilities like STADS), a change that affects all applications. The problem is that by doing that every time I am about to launch Windows Media Center, except that it's not that elegant, also makes all other Windows application's sounds to pass through SPDIF too, bypassing analog output completely, blending with what I am watching at Windows Media Center. Is there a way to make SPDIF as the default playback device for Windows Media Center? I know other programs that have a setting like that (foobar2000 for example) working like charm, allowing me to even have different outputs working at the same time (tested with my current card successfully). But when comes to Windows Media Center... it just use what the default playback device is all the time. The only setting that I know of, is under: Settings General Windows Media Center Setup Set Up Your Speakers and what it does is to just change the default playback device for entire system. Please help!

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  • Do you have any additions or alterations to this list of popular audio formats?

    - by roja
    All, I am trying to compile a list of common audio file formats used in both personal storage and peer transmission. I have compiled the following list, do you think that there are any significant formats missing? Are any of them not actually common formats? Any advice/alterations are highly useful. advanced audio coding, apple lossless audio file, atrac3 audio file, atrac audio file, audio interchange file format, core audio file, free lossless audio codec file, mpeg 1 audio layer 3, mpeg 2 audio, mpeg 4 audio book file, musical instrument digital interface, ogg vorbis compressed audio file, open media framework file, real audio, real audio media, waveform audio file format, windows media audio Kind regards, Roja

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  • Boost Audio Input on OS X?

    - by alanstorm
    I'm using my 13" Mac Book Pro's audio input functionality with an external microphone (recent vintage, bought around Thanksgiving). I've increased my input volume to the maximum in system preference, but the resulting recorded volume (using iShowU HD) is very low. Is there anyway to increase the input volume/sensitivity beyond Apple's default settings? I've found plenty on google about increasing the OUTPUT volume, but I want to increase the input volume.

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  • JavaScript audio not playing outside of jQuery function

    - by user1814016
    I know the question title doesn't make much sense, but I can't think of a better way to put it. I am a newbie to jQuery and I'm using this code to fade in a <div> and play a sound: $(document).ready(function(){ $('#speech').fadeIn('medium', function() { play('msg_appear'); var sptx = $('<p class="stext">').text('There is nothing here.'); $('#speech').append(sptx); $('.stext').typeOut({marker: '', delay: 22}); }); }); This code runs fine however the sound plays after the fade-in is complete. I wanted it to play while it was fading in, so I tried placing the play() call outside of the fade-in function like this: $(document).ready(function(){ play('msg_appear'); $('#speech').fadeIn('medium', function() { However, now it's not playing at all. There's no errors on the JavaScript console so I'm unsure if it's a syntax error, and probably something obvious, but I don't know what. play() is a function I found to play audio, here it is if it matters at all. I placed it in the same file the above code is; right above the $(document).ready(). function play(sound) { if (window.HTMLAudioElement) { var snd = new Audio(''); if(snd.canPlayType('audio/ogg')) { snd = new Audio(sound + '.ogg'); } else if(snd.canPlayType('audio/mp3')) { snd = new Audio(sound + '.mp3'); } snd.play(); } else { alert('HTML5 Audio is not supported by your browser!'); } }

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  • Capturing and Transforming ASP.NET Output with Response.Filter

    - by Rick Strahl
    During one of my Handlers and Modules session at DevConnections this week one of the attendees asked a question that I didn’t have an immediate answer for. Basically he wanted to capture response output completely and then apply some filtering to the output – effectively injecting some additional content into the page AFTER the page had completely rendered. Specifically the output should be captured from anywhere – not just a page and have this code injected into the page. Some time ago I posted some code that allows you to capture ASP.NET Page output by overriding the Render() method, capturing the HtmlTextWriter() and reading its content, modifying the rendered data as text then writing it back out. I’ve actually used this approach on a few occasions and it works fine for ASP.NET pages. But this obviously won’t work outside of the Page class environment and it’s not really generic – you have to create a custom page class in order to handle the output capture. [updated 11/16/2009 – updated ResponseFilterStream implementation and a few additional notes based on comments] Enter Response.Filter However, ASP.NET includes a Response.Filter which can be used – well to filter output. Basically Response.Filter is a stream through which the OutputStream is piped back to the Web Server (indirectly). As content is written into the Response object, the filter stream receives the appropriate Stream commands like Write, Flush and Close as well as read operations although for a Response.Filter that’s uncommon to be hit. The Response.Filter can be programmatically replaced at runtime which allows you to effectively intercept all output generation that runs through ASP.NET. A common Example: Dynamic GZip Encoding A rather common use of Response.Filter hooking up code based, dynamic  GZip compression for requests which is dead simple by applying a GZipStream (or DeflateStream) to Response.Filter. The following generic routines can be used very easily to detect GZip capability of the client and compress response output with a single line of code and a couple of library helper routines: WebUtils.GZipEncodePage(); which is handled with a few lines of reusable code and a couple of static helper methods: /// <summary> ///Sets up the current page or handler to use GZip through a Response.Filter ///IMPORTANT:  ///You have to call this method before any output is generated! /// </summary> public static void GZipEncodePage() {     HttpResponse Response = HttpContext.Current.Response;     if(IsGZipSupported())     {         stringAcceptEncoding = HttpContext.Current.Request.Headers["Accept-Encoding"];         if(AcceptEncoding.Contains("deflate"))         {             Response.Filter = newSystem.IO.Compression.DeflateStream(Response.Filter,                                        System.IO.Compression.CompressionMode.Compress);             Response.AppendHeader("Content-Encoding", "deflate");         }         else        {             Response.Filter = newSystem.IO.Compression.GZipStream(Response.Filter,                                       System.IO.Compression.CompressionMode.Compress);             Response.AppendHeader("Content-Encoding", "gzip");                            }     }     // Allow proxy servers to cache encoded and unencoded versions separately    Response.AppendHeader("Vary", "Content-Encoding"); } /// <summary> /// Determines if GZip is supported /// </summary> /// <returns></returns> public static bool IsGZipSupported() { string AcceptEncoding = HttpContext.Current.Request.Headers["Accept-Encoding"]; if (!string.IsNullOrEmpty(AcceptEncoding) && (AcceptEncoding.Contains("gzip") || AcceptEncoding.Contains("deflate"))) return true; return false; } GZipStream and DeflateStream are streams that are assigned to Response.Filter and by doing so apply the appropriate compression on the active Response. Response.Filter content is chunked So to implement a Response.Filter effectively requires only that you implement a custom stream and handle the Write() method to capture Response output as it’s written. At first blush this seems very simple – you capture the output in Write, transform it and write out the transformed content in one pass. And that indeed works for small amounts of content. But you see, the problem is that output is written in small buffer chunks (a little less than 16k it appears) rather than just a single Write() statement into the stream, which makes perfect sense for ASP.NET to stream data back to IIS in smaller chunks to minimize memory usage en route. Unfortunately this also makes it a more difficult to implement any filtering routines since you don’t directly get access to all of the response content which is problematic especially if those filtering routines require you to look at the ENTIRE response in order to transform or capture the output as is needed for the solution the gentleman in my session asked for. So in order to address this a slightly different approach is required that basically captures all the Write() buffers passed into a cached stream and then making the stream available only when it’s complete and ready to be flushed. As I was thinking about the implementation I also started thinking about the few instances when I’ve used Response.Filter implementations. Each time I had to create a new Stream subclass and create my custom functionality but in the end each implementation did the same thing – capturing output and transforming it. I thought there should be an easier way to do this by creating a re-usable Stream class that can handle stream transformations that are common to Response.Filter implementations. Creating a semi-generic Response Filter Stream Class What I ended up with is a ResponseFilterStream class that provides a handful of Events that allow you to capture and/or transform Response content. The class implements a subclass of Stream and then overrides Write() and Flush() to handle capturing and transformation operations. By exposing events it’s easy to hook up capture or transformation operations via single focused methods. ResponseFilterStream exposes the following events: CaptureStream, CaptureString Captures the output only and provides either a MemoryStream or String with the final page output. Capture is hooked to the Flush() operation of the stream. TransformStream, TransformString Allows you to transform the complete response output with events that receive a MemoryStream or String respectively and can you modify the output then return it back as a return value. The transformed output is then written back out in a single chunk to the response output stream. These events capture all output internally first then write the entire buffer into the response. TransformWrite, TransformWriteString Allows you to transform the Response data as it is written in its original chunk size in the Stream’s Write() method. Unlike TransformStream/TransformString which operate on the complete output, these events only see the current chunk of data written. This is more efficient as there’s no caching involved, but can cause problems due to searched content splitting over multiple chunks. Using this implementation, creating a custom Response.Filter transformation becomes as simple as the following code. To hook up the Response.Filter using the MemoryStream version event: ResponseFilterStream filter = new ResponseFilterStream(Response.Filter); filter.TransformStream += filter_TransformStream; Response.Filter = filter; and the event handler to do the transformation: MemoryStream filter_TransformStream(MemoryStream ms) { Encoding encoding = HttpContext.Current.Response.ContentEncoding; string output = encoding.GetString(ms.ToArray()); output = FixPaths(output); ms = new MemoryStream(output.Length); byte[] buffer = encoding.GetBytes(output); ms.Write(buffer,0,buffer.Length); return ms; } private string FixPaths(string output) { string path = HttpContext.Current.Request.ApplicationPath; // override root path wonkiness if (path == "/") path = ""; output = output.Replace("\"~/", "\"" + path + "/").Replace("'~/", "'" + path + "/"); return output; } The idea of the event handler is that you can do whatever you want to the stream and return back a stream – either the same one that’s been modified or a brand new one – which is then sent back to as the final response. The above code can be simplified even more by using the string version events which handle the stream to string conversions for you: ResponseFilterStream filter = new ResponseFilterStream(Response.Filter); filter.TransformString += filter_TransformString; Response.Filter = filter; and the event handler to do the transformation calling the same FixPaths method shown above: string filter_TransformString(string output) { return FixPaths(output); } The events for capturing output and capturing and transforming chunks work in a very similar way. By using events to handle the transformations ResponseFilterStream becomes a reusable component and we don’t have to create a new stream class or subclass an existing Stream based classed. By the way, the example used here is kind of a cool trick which transforms “~/” expressions inside of the final generated HTML output – even in plain HTML controls not HTML controls – and transforms them into the appropriate application relative path in the same way that ResolveUrl would do. So you can write plain old HTML like this: <a href=”~/default.aspx”>Home</a>  and have it turned into: <a href=”/myVirtual/default.aspx”>Home</a>  without having to use an ASP.NET control like Hyperlink or Image or having to constantly use: <img src=”<%= ResolveUrl(“~/images/home.gif”) %>” /> in MVC applications (which frankly is one of the most annoying things about MVC especially given the path hell that extension-less and endpoint-less URLs impose). I can’t take credit for this idea. While discussing the Response.Filter issues on Twitter a hint from Dylan Beattie who pointed me at one of his examples which does something similar. I thought the idea was cool enough to use an example for future demos of Response.Filter functionality in ASP.NET next I time I do the Modules and Handlers talk (which was great fun BTW). How practical this is is debatable however since there’s definitely some overhead to using a Response.Filter in general and especially on one that caches the output and the re-writes it later. Make sure to test for performance anytime you use Response.Filter hookup and make sure it' doesn’t end up killing perf on you. You’ve been warned :-}. How does ResponseFilterStream work? The big win of this implementation IMHO is that it’s a reusable  component – so for implementation there’s no new class, no subclassing – you simply attach to an event to implement an event handler method with a straight forward signature to retrieve the stream or string you’re interested in. The implementation is based on a subclass of Stream as is required in order to handle the Response.Filter requirements. What’s different than other implementations I’ve seen in various places is that it supports capturing output as a whole to allow retrieving the full response output for capture or modification. The exception are the TransformWrite and TransformWrite events which operate only active chunk of data written by the Response. For captured output, the Write() method captures output into an internal MemoryStream that is cached until writing is complete. So Write() is called when ASP.NET writes to the Response stream, but the filter doesn’t pass on the Write immediately to the filter’s internal stream. The data is cached and only when the Flush() method is called to finalize the Stream’s output do we actually send the cached stream off for transformation (if the events are hooked up) and THEN finally write out the returned content in one big chunk. Here’s the implementation of ResponseFilterStream: /// <summary> /// A semi-generic Stream implementation for Response.Filter with /// an event interface for handling Content transformations via /// Stream or String. /// <remarks> /// Use with care for large output as this implementation copies /// the output into a memory stream and so increases memory usage. /// </remarks> /// </summary> public class ResponseFilterStream : Stream { /// <summary> /// The original stream /// </summary> Stream _stream; /// <summary> /// Current position in the original stream /// </summary> long _position; /// <summary> /// Stream that original content is read into /// and then passed to TransformStream function /// </summary> MemoryStream _cacheStream = new MemoryStream(5000); /// <summary> /// Internal pointer that that keeps track of the size /// of the cacheStream /// </summary> int _cachePointer = 0; /// <summary> /// /// </summary> /// <param name="responseStream"></param> public ResponseFilterStream(Stream responseStream) { _stream = responseStream; } /// <summary> /// Determines whether the stream is captured /// </summary> private bool IsCaptured { get { if (CaptureStream != null || CaptureString != null || TransformStream != null || TransformString != null) return true; return false; } } /// <summary> /// Determines whether the Write method is outputting data immediately /// or delaying output until Flush() is fired. /// </summary> private bool IsOutputDelayed { get { if (TransformStream != null || TransformString != null) return true; return false; } } /// <summary> /// Event that captures Response output and makes it available /// as a MemoryStream instance. Output is captured but won't /// affect Response output. /// </summary> public event Action<MemoryStream> CaptureStream; /// <summary> /// Event that captures Response output and makes it available /// as a string. Output is captured but won't affect Response output. /// </summary> public event Action<string> CaptureString; /// <summary> /// Event that allows you transform the stream as each chunk of /// the output is written in the Write() operation of the stream. /// This means that that it's possible/likely that the input /// buffer will not contain the full response output but only /// one of potentially many chunks. /// /// This event is called as part of the filter stream's Write() /// operation. /// </summary> public event Func<byte[], byte[]> TransformWrite; /// <summary> /// Event that allows you to transform the response stream as /// each chunk of bytep[] output is written during the stream's write /// operation. This means it's possibly/likely that the string /// passed to the handler only contains a portion of the full /// output. Typical buffer chunks are around 16k a piece. /// /// This event is called as part of the stream's Write operation. /// </summary> public event Func<string, string> TransformWriteString; /// <summary> /// This event allows capturing and transformation of the entire /// output stream by caching all write operations and delaying final /// response output until Flush() is called on the stream. /// </summary> public event Func<MemoryStream, MemoryStream> TransformStream; /// <summary> /// Event that can be hooked up to handle Response.Filter /// Transformation. Passed a string that you can modify and /// return back as a return value. The modified content /// will become the final output. /// </summary> public event Func<string, string> TransformString; protected virtual void OnCaptureStream(MemoryStream ms) { if (CaptureStream != null) CaptureStream(ms); } private void OnCaptureStringInternal(MemoryStream ms) { if (CaptureString != null) { string content = HttpContext.Current.Response.ContentEncoding.GetString(ms.ToArray()); OnCaptureString(content); } } protected virtual void OnCaptureString(string output) { if (CaptureString != null) CaptureString(output); } protected virtual byte[] OnTransformWrite(byte[] buffer) { if (TransformWrite != null) return TransformWrite(buffer); return buffer; } private byte[] OnTransformWriteStringInternal(byte[] buffer) { Encoding encoding = HttpContext.Current.Response.ContentEncoding; string output = OnTransformWriteString(encoding.GetString(buffer)); return encoding.GetBytes(output); } private string OnTransformWriteString(string value) { if (TransformWriteString != null) return TransformWriteString(value); return value; } protected virtual MemoryStream OnTransformCompleteStream(MemoryStream ms) { if (TransformStream != null) return TransformStream(ms); return ms; } /// <summary> /// Allows transforming of strings /// /// Note this handler is internal and not meant to be overridden /// as the TransformString Event has to be hooked up in order /// for this handler to even fire to avoid the overhead of string /// conversion on every pass through. /// </summary> /// <param name="responseText"></param> /// <returns></returns> private string OnTransformCompleteString(string responseText) { if (TransformString != null) TransformString(responseText); return responseText; } /// <summary> /// Wrapper method form OnTransformString that handles /// stream to string and vice versa conversions /// </summary> /// <param name="ms"></param> /// <returns></returns> internal MemoryStream OnTransformCompleteStringInternal(MemoryStream ms) { if (TransformString == null) return ms; //string content = ms.GetAsString(); string content = HttpContext.Current.Response.ContentEncoding.GetString(ms.ToArray()); content = TransformString(content); byte[] buffer = HttpContext.Current.Response.ContentEncoding.GetBytes(content); ms = new MemoryStream(); ms.Write(buffer, 0, buffer.Length); //ms.WriteString(content); return ms; } /// <summary> /// /// </summary> public override bool CanRead { get { return true; } } public override bool CanSeek { get { return true; } } /// <summary> /// /// </summary> public override bool CanWrite { get { return true; } } /// <summary> /// /// </summary> public override long Length { get { return 0; } } /// <summary> /// /// </summary> public override long Position { get { return _position; } set { _position = value; } } /// <summary> /// /// </summary> /// <param name="offset"></param> /// <param name="direction"></param> /// <returns></returns> public override long Seek(long offset, System.IO.SeekOrigin direction) { return _stream.Seek(offset, direction); } /// <summary> /// /// </summary> /// <param name="length"></param> public override void SetLength(long length) { _stream.SetLength(length); } /// <summary> /// /// </summary> public override void Close() { _stream.Close(); } /// <summary> /// Override flush by writing out the cached stream data /// </summary> public override void Flush() { if (IsCaptured && _cacheStream.Length > 0) { // Check for transform implementations _cacheStream = OnTransformCompleteStream(_cacheStream); _cacheStream = OnTransformCompleteStringInternal(_cacheStream); OnCaptureStream(_cacheStream); OnCaptureStringInternal(_cacheStream); // write the stream back out if output was delayed if (IsOutputDelayed) _stream.Write(_cacheStream.ToArray(), 0, (int)_cacheStream.Length); // Clear the cache once we've written it out _cacheStream.SetLength(0); } // default flush behavior _stream.Flush(); } /// <summary> /// /// </summary> /// <param name="buffer"></param> /// <param name="offset"></param> /// <param name="count"></param> /// <returns></returns> public override int Read(byte[] buffer, int offset, int count) { return _stream.Read(buffer, offset, count); } /// <summary> /// Overriden to capture output written by ASP.NET and captured /// into a cached stream that is written out later when Flush() /// is called. /// </summary> /// <param name="buffer"></param> /// <param name="offset"></param> /// <param name="count"></param> public override void Write(byte[] buffer, int offset, int count) { if ( IsCaptured ) { // copy to holding buffer only - we'll write out later _cacheStream.Write(buffer, 0, count); _cachePointer += count; } // just transform this buffer if (TransformWrite != null) buffer = OnTransformWrite(buffer); if (TransformWriteString != null) buffer = OnTransformWriteStringInternal(buffer); if (!IsOutputDelayed) _stream.Write(buffer, offset, buffer.Length); } } The key features are the events and corresponding OnXXX methods that handle the event hookups, and the Write() and Flush() methods of the stream implementation. All the rest of the members tend to be plain jane passthrough stream implementation code without much consequence. I do love the way Action<t> and Func<T> make it so easy to create the event signatures for the various events – sweet. A few Things to consider Performance Response.Filter is not great for performance in general as it adds another layer of indirection to the ASP.NET output pipeline, and this implementation in particular adds a memory hit as it basically duplicates the response output into the cached memory stream which is necessary since you may have to look at the entire response. If you have large pages in particular this can cause potentially serious memory pressure in your server application. So be careful of wholesale adoption of this (or other) Response.Filters. Make sure to do some performance testing to ensure it’s not killing your app’s performance. Response.Filter works everywhere A few questions came up in comments and discussion as to capturing ALL output hitting the site and – yes you can definitely do that by assigning a Response.Filter inside of a module. If you do this however you’ll want to be very careful and decide which content you actually want to capture especially in IIS 7 which passes ALL content – including static images/CSS etc. through the ASP.NET pipeline. So it is important to filter only on what you’re looking for – like the page extension or maybe more effectively the Response.ContentType. Response.Filter Chaining Originally I thought that filter chaining doesn’t work at all due to a bug in the stream implementation code. But it’s quite possible to assign multiple filters to the Response.Filter property. So the following actually works to both compress the output and apply the transformed content: WebUtils.GZipEncodePage(); ResponseFilterStream filter = new ResponseFilterStream(Response.Filter); filter.TransformString += filter_TransformString; Response.Filter = filter; However the following does not work resulting in invalid content encoding errors: ResponseFilterStream filter = new ResponseFilterStream(Response.Filter); filter.TransformString += filter_TransformString; Response.Filter = filter; WebUtils.GZipEncodePage(); In other words multiple Response filters can work together but it depends entirely on the implementation whether they can be chained or in which order they can be chained. In this case running the GZip/Deflate stream filters apparently relies on the original content length of the output and chokes when the content is modified. But if attaching the compression first it works fine as unintuitive as that may seem. Resources Download example code Capture Output from ASP.NET Pages © Rick Strahl, West Wind Technologies, 2005-2010Posted in ASP.NET  

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  • Monitoring an audio line.

    - by Stefan Liebenberg
    I need to monitor my audio line-in in linux, and in the event that audio is played, the sound must be recorded and saved to a file. Similiar to how motion monitors the video feed. Is it possible to do this with bash? something along the lines of: #!/bin/bash # audio device device=/dev/audio-line-in # below this threshold audio will not be recorded. noise_threshold=10 # folder where recordings are stored storage_folder=~/recordings # run indefenitly, until Ctrl-C is pressed while true; do # noise_level() represents a function to determine # the noise level from device if noise_level( $device ) > $noise_threshold; then # stream from device to file, can be encoded to mp3 later. cat $device > $storage_folder/`date`.raw fi; done;

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  • PhP/HTML play button [migrated]

    - by Marian
    I'm wanting to make my own small webpage, I've got a domain Saoo.eu As you see there is a small play button in the corner witch plays a playlist. Is there anyway to have that playbutton on each page I'd add in the future without resetting every time the page changes? Am I forced to use iFrames for that? This is my player code <button id="audioControl" style="width:30px;height:25px;"></button> <audio id="aud" src="" autoplay autobuffer /> Script: $(document).ready(function() { $('#audioControl').html('II'); if(Modernizr.audio && Modernizr.audio.mp3) { audio.setAttribute("src",'http://daokun.webs.com/play0.mp3'); } else if(Modernizr.audio && Modernizr.audio.wav) { audio.setAttribute("src", 'http://daokun.webs.com/play0.ogg'); } }); var audio = document.getElementById('aud'), count = 0; $('#audioControl').toggle( function () { audio.pause(); $('#audioControl').html('>'); }, function () { audio.play(); $('#audioControl').html('II'); } ); audio.addEventListener("ended", function() { count++; if(count == 4){count = 0;} if(Modernizr.audio && Modernizr.audio.mp3) { audio.setAttribute("src",'http://daokun.webs.com/play'+count+'.mp3'); } else if(Modernizr.audio && Modernizr.audio.wav) { audio.setAttribute("src", 'http://daokun.webs.com/play'+count+'.ogg'); } audio.load(); });

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  • Web Audio API and mobile browsers

    - by Michael
    I've run into a problem while implementing sound and music into an HTML game that I'm building. I'm using the Web Audio API, loading all the sound files with XMLHttpRequests and decoding them into an AudioBufferSourceNode with AudioContext.prototype.decodeAudioData(). It looks something like this: var request = new XMLHttpRequest(); request.open("GET", "soundfile.ogg", true); request.responseType = "arraybuffer"; request.onload = function() { context.decodeAudioData(request.response) } request.send(); Everything plays fine, but on mobile the decodeAudioData takes an absurdly long time for the background music. I then tried using AudioContext.prototype.createMediaElementSource() to load the music from an HTML Audio object, since they support streaming and don't have to load the whole file into memory at once. It looked something like this: var audio = new Audio('soundfile.ogg'); var source = context.createMediaElementSource(audio); var mainVolume = context.createGain(); source.connect(mainVolume); mainVolume.connect(context.destination); This loads much faster, but the audio volume isn't affected by the gain node. Works fine on desktop, so I'm assuming this is a bug/limitation of mobile Chrome (testing on Android). Is there actually no good, well-performing way to handle sound on mobile browsers or am just I doing something stupid?

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  • Audio programming resources

    - by rashleighp
    I've been very interested in the last few months about getting in to audio programming (I'm from a musical background). I've been a .NET developer for two years and have also done some objective c for an iPhone app recently. I realise I would probably need to work on my C++ chops and have been having a play around with FMOD EX and doing a lot of research into the industry. I was just wondering if anyone could suggest some good resources for audio programming (be they websites, podcasts, books, videos, online courses etc). Anything from Fourier analysis, low level coding, audio engine creation to audio APIs. I just want to learn as much as possible! Thanks in advance.

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  • Setting Up Audio on a Server Install

    - by tdcrenshaw
    I'm running on a clean install of 10.10 Server edition and have alsa-base, alsa-tools, alsa-utils, alsaplayer, and alsa-firmware-loader installed. At one point I installed pulseaudio, but I have since removed it. I've tried the following lspci | grep audio 00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 01) 01:06.0 Multimedia audio controller: Creative Labs [SB Live! Value] EMU10k1X aplay -l aplay: device_list:235: no soundcards found... alsamixer can not open mixer: No such file or directory When I search for modules with find /lib/modules/`uname -r` | grep snd I do get a list of modules I'm not very experienced with alsa setup, so I'm not sure where to go from here

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  • How to stream semi-live audio over internet

    - by Thomas Tempelmann
    I want to write something like Skype, i.e. I have a constant audio stream on one computer and then recompress it in a format that's suitable for a latent internet connection, receive it on the other end and play it. Let's also assume that the internet connection is fairly modern and fast, i.e. DSL or alike, no slow connections over phone and such. The involved computers will also be rather modern (Dual Core Intel CPUs at 2GHz or more). I know how to handle the audio on the machines. What I don't know is how to transmit the audio in an efficient way. The challenges are: I'd like get good audio quality across the line. The stream should be received without drops. The stream may, however, be received with a little delay (a second delay is acceptable). I imagine that the transport software could first determine the average (and max) latency, then start the stream and tell the receiver to wait for that max latency before starting to play the audio. With that, if the latency doesn't get any higher, the entire stream will be playable on the other side without stutter or drops. If, due to unexpected IP latencies or blockages, the stream does get cut off, I want to be able to notice this so that I can take actions (e.g. abort the stream) and eventually start a new transmission. What are my options if I want do use ready-made software for the compression and tranmission? I have no intention to write my own audio compression engine, really. OTOH, I plan to sell the solution in a vertical market, meaning I can afford a few dollars of license fees per copy, but not $100s. I guess the simplest solution would be to just open a TCP stream, send a few packets back and forth to determine their running time (or even use UDP for that), then use the results as the guide for my max latency value, then simply fire the audio data in its raw form (uncompressed 16 bit stereo), along with a timing code over the TCP connection. The receiver reads the data and plays it with the pre-determined delay. That might just work with the type of fast connection I expect. I just wonder if there are better solutions to reach this goal, with better performance (lower latency) and less data (compressed). BTW, I first try to implement this on OS X, but might want to do it on Windows, too, if it proves successful.

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  • Record audio via MediaRecorder

    - by Isuru Madusanka
    I am trying to record audio by MediaRecorder, and I get an error, I tried to change everything and nothing works. Last two hours I try to find the error, I used Log class too and I found out that error occurred when it call recorder.start() method. What could be the problem? public class AudioRecorderActivity extends Activity { MediaRecorder recorder; File audioFile = null; private static final String TAG = "AudioRecorderActivity"; private View startButton; private View stopButton; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); startButton = findViewById(R.id.start); stopButton = findViewById(R.id.stop); setContentView(R.layout.main); } public void startRecording(View view) throws IOException{ startButton.setEnabled(false); stopButton.setEnabled(true); File sampleDir = Environment.getExternalStorageDirectory(); try{ audioFile = File.createTempFile("sound", ".3gp", sampleDir); }catch(IOException e){ Toast.makeText(getApplicationContext(), "SD Card Access Error", Toast.LENGTH_LONG).show(); Log.e(TAG, "Sdcard access error"); return; } recorder = new MediaRecorder(); recorder.setAudioSource(MediaRecorder.AudioSource.MIC); recorder.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP); recorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB); recorder.setAudioEncodingBitRate(16); recorder.setAudioSamplingRate(44100); recorder.setOutputFile(audioFile.getAbsolutePath()); recorder.prepare(); recorder.start(); } public void stopRecording(View view){ startButton.setEnabled(true); stopButton.setEnabled(false); recorder.stop(); recorder.release(); addRecordingToMediaLibrary(); } protected void addRecordingToMediaLibrary(){ ContentValues values = new ContentValues(4); long current = System.currentTimeMillis(); values.put(MediaStore.Audio.Media.TITLE, "audio" + audioFile.getName()); values.put(MediaStore.Audio.Media.DATE_ADDED, (int)(current/1000)); values.put(MediaStore.Audio.Media.MIME_TYPE, "audio/3gpp"); values.put(MediaStore.Audio.Media.DATA, audioFile.getAbsolutePath()); ContentResolver contentResolver = getContentResolver(); Uri base = MediaStore.Audio.Media.EXTERNAL_CONTENT_URI; Uri newUri = contentResolver.insert(base, values); sendBroadcast(new Intent(Intent.ACTION_MEDIA_SCANNER_SCAN_FILE, newUri)); Toast.makeText(this, "Added File" + newUri, Toast.LENGTH_LONG).show(); } } And here is the xml layout. <?xml version="1.0" encoding="utf-8"?> <RelativeLayout xmlns:android="http://schemas.android.com/apk/res/android" android:id="@+id/RelativeLayout1" android:layout_width="fill_parent" android:layout_height="fill_parent" android:orientation="vertical" > <Button android:id="@+id/start" android:layout_width="wrap_content" android:layout_height="wrap_content" android:layout_alignParentTop="true" android:layout_centerHorizontal="true" android:layout_marginTop="146dp" android:onClick="startRecording" android:text="Start Recording" /> <Button android:id="@+id/stop" android:layout_width="wrap_content" android:layout_height="wrap_content" android:layout_alignLeft="@+id/start" android:layout_below="@+id/start" android:layout_marginTop="41dp" android:enabled="false" android:onClick="stopRecording" android:text="Stop Recording" /> </RelativeLayout> And I added permission to AndroidManifest file. <?xml version="1.0" encoding="utf-8"?> <manifest xmlns:android="http://schemas.android.com/apk/res/android" package="in.isuru.audiorecorder" android:versionCode="1" android:versionName="1.0" > <uses-sdk android:minSdkVersion="8" /> <application android:icon="@drawable/ic_launcher" android:label="@string/app_name" > <activity android:name=".AudioRecorderActivity" android:label="@string/app_name" > <intent-filter> <action android:name="android.intent.action.MAIN" /> <category android:name="android.intent.category.LAUNCHER" /> </intent-filter> </activity> </application> <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/> <uses-permission android:name="android.permission.RECORD_AUDIO" /> </manifest> I need to record high quality audio. Thanks!

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  • How to fix these compiler errors?

    - by Sandra Schlichting
    I have this source code from 2001 that I would like to compile. It gives this: $ make g++ -O99 -Wall -DLINUX -pedantic -c -o audio.o audio.cpp In file included from audio.cpp:7: audio.h:14: error: use of enum ‘mad_flow’ without previous declaration audio.h:15: error: use of enum ‘mad_flow’ without previous declaration audio.h:17: error: use of enum ‘mad_flow’ without previous declaration audio.cpp: In function ‘mad_flow audio::input(void*, mad_stream*)’: audio.cpp:19: error: new declaration ‘mad_flow audio::input(void*, mad_stream*)’ audio.h:14: error: ambiguates old declaration ‘int audio::input(void*, mad_stream*)’ audio.h:11: error: ‘size_t audio::stream::BufferPos’ is private audio.cpp:23: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:23: error: within this context audio.h:10: error: ‘char* audio::stream::Buffer’ is private audio.cpp:26: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:26: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferPos’ is private audio.cpp:27: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:27: error: within this context audio.cpp: In function ‘mad_flow audio::output(void*, const mad_header*, mad_pcm*)’: audio.cpp:49: error: new declaration ‘mad_flow audio::output(void*, const mad_header*, mad_pcm*)’ audio.h:15: error: ambiguates old declaration ‘int audio::output(void*, const mad_header*, mad_pcm*)’ audio.cpp: In function ‘mad_flow audio::error(void*, mad_stream*, mad_frame*)’: audio.cpp:83: error: new declaration ‘mad_flow audio::error(void*, mad_stream*, mad_frame*)’ audio.h:17: error: ambiguates old declaration ‘int audio::error(void*, mad_stream*, mad_frame*)’ audio.cpp: In constructor ‘audio::stream::stream(const char*)’: audio.cpp:119: error: ‘input’ was not declared in this scope audio.cpp:122: error: ‘output’ was not declared in this scope audio.cpp:123: error: ‘error’ was not declared in this scope make: *** [audio.o] Error 1 audio.h contains #include <stdlib.h> #include "mad.h" namespace audio { class stream { private: char* Buffer; size_t BufferSize, BufferPos; struct mad_decoder Decoder; friend enum mad_flow input(void* Data, struct mad_stream* MadStream); friend enum mad_flow output(void* Data, const struct mad_header* Header, struct mad_pcm* PCM); friend enum mad_flow error(void* Data, struct mad_stream* MadStream, struct mad_frame* Frame); public: stream(const char* FileName); ~stream(); void play(); }; } I have tried to just insert enum mad_flow {}; but that just gave a new problem. Can anyone see how to fix this?

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  • Is there an audio recording application/tool that has Tivo-like functionality?

    - by Bob
    I do a lot of live speech recording that requires me to quickly jump back and then transcribe a particular piece of the audio, then go back to recording again, while still maintaining the full audio file. So Far I've done this by splitting the audio and running one line to a recorder (for the whole audio), and one to my computer. Then I use something like Audacity to record, and then stop/go back whenever I hear something worth transcribing. This requires me to stop the recording, then start it up again and I end up missing chunks of the speech I'm listening to. Is there a tool that would let me rewind, then listen again and continue listening at a buffered distance from the audio recording, the way Tivo does with television shows?

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  • HTML Audio performance

    - by user1888309
    I'm working on HTML drum machine, and I`ve met some performance issues, rhythm start to break if BPM is higher than 110 but I'm expecting to make it work on BPM over 180. I guess that it can be related with format or codec of audio files, however it also maybe that my code is not very optimised (as I can see from JS CPU profiling it's not). So I'm expecting you guys give me some code review or some hints on optimisation. Although all similar projects I've found on internet didn't work good and maybe it's just restrictions of Audio API. By the way, it's very raw and sounds works only on Chrome under Mac OS, so any advise on audio encoding for web also would be great Project on Github pages Screenshot of Groove which breaks UPDATE Ok, I've found that I was encoding audio files incorrectly, after fixing that rhythm stopped breaking, and also it started working in Mozilla. But still there are issues on windows OS.

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