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  • Audio player in flex

    - by prasanth
    Hi All, I want to play Audio in flex application.I am using SWFLoader for it.Is there any other way to load audio into Flex applications. Thanks & Regards, Prasanth Babu

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  • Channel Audio from .mp4?

    - by Kyle
    Hi there. I am at a loss here after searching around with no results. I am attempting to channel the audio specifically from an .mp4 for use in a driver. I am aware that there are programs which extract the audio from .mp4's, but I am looking for another approach without using external applications such as those.. is there any direction that someone can point me towards to solve this problem? Thanks, -K

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  • How to intercept and apply effects to Firefox audio/sound output

    - by Tom
    Hi I want to build a Firefox extension that will allow me to directly manipulate the audio output, applying live filters and effects, from (for example) a streaming video site. Im struggling to find any good resources to help me. I think the effects bit will be ok but I need to find a way of intercepting the audio stream output. Does anyone know if this is possible? Thanks, Tom

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  • Control over who can use audio output channel in XP

    - by Phil
    I have a need to turn off other audio sources when I plan to use the Text to Speech API. The other audio may be in another process. I have looked at the mixer control, but I really only have control of the output there. Is there another place in XP that I can control the output so only my app is able to be heard?

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  • How to use the Bass library to convert audio to another format (using C#)

    - by Siemsen
    Good day everyone and happy holidays. I'm trying to create a "simple" program that allows me to create a list of video files (Youtube video's to be precise) that are then run through by my program and converted any given format to pure MP3 audio. For this purpose I'm trying to use the BASS.dll and well it isn't going so well. I was wondering if anyone has used BASS to convert from one audio format to another? Or is there another library better suited for this?

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  • Audio Streaming API's: Wifi vs what?

    - by Moshe
    I've noticed certain radio apps, that some stations required wifi and others did not. What were those other stations possibly using? Are there other methods of streaming audio on iOS? Apparently, I was not clear in my question before. I'm asking in terms of API's. Is there an API to interact directly with say, FM radio, on iOS? Is wifi the only way of streaming audio?

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  • Low delay audio on Android via NDK

    - by hkhauke
    Hi, it seems that this question has been asked before, I just would like to know whether there is an update in Android. I plan to write an audio application involving low delay audio I/O (appr. < 10 ms). It seems not to be possible based on the methods proposed by the SDK, hence is there - in the meantime - a way to achieve this goal using the NDK? Best regards, HK

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  • CMUS Error: opening audio device: No such device

    - by clamp
    I cant seem to play any audio with CMUS because it always gives the above error the output of lspci -v | grep -A7 -i "audio" gives 00:1b.0 Audio device: Intel Corporation NM10/ICH7 Family High Definition Audio Controller (rev 02) Subsystem: ASRock Incorporation Device c892 Flags: bus master, fast devsel, latency 0, IRQ 49 Memory at dff00000 (64-bit, non-prefetchable) [size=16K] Capabilities: Kernel driver in use: snd_hda_intel 00:1c.0 PCI bridge: Intel Corporation NM10/ICH7 Family PCI Express Port 1 (rev 02) (prog-if 00 [Normal decode]) what could be the problem?

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  • How can I play a DVD-A (DVD-Audio) disc?

    - by Marek Grzenkowicz
    I was able to play such a disc using VLC, but I am wondering if any music player supports the DVD-A format. UPDATE: I checked Rhythmbox and Banshee - I found no option like Play Disc or Open Disc, so I have no idea how I could even try to start playing a DVD-A disc. This has also nothing to do with missing plug-ins or codecs - the packages gstreamer0.10-plugins-ugly, gstreamer0.10-plugins-bad and ubuntu-restricted-extras were installed on my machine before. I guess I am stuck with VLC (Totem shows a DVD menu, but then hangs). However, I am missing the regular music player experience - ability to change order of tracks, adding tracks to a playlist, listening only to selected tracks, etc. I found Idea #22415: Please add full support for DVD-Audio and LPCM at Ubuntu Brainstorm.

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  • ReaderWriterLockSlim and Pulse/Wait

    - by Jono
    Is there an equivalent of Monitor.Pulse and Monitor.Wait that I can use in conjunction with a ReaderWriterLockSlim? I have a class where I've encapsulated multi-threaded access to an underlying queue. To enqueue something, I acquire a lock that protects the underlying queue (and a couple of other objects) then add the item and Monitor.Pulse the locked object to signal that something was added to the queue. public void Enqueue(ITask task) { lock (mutex) { underlying.Enqueue(task); Monitor.Pulse(mutex); } } On the other end of the queue, I have a single background thread that continuously processes messages as they arrive on the queue. It uses Monitor.Wait when there are no items in the queue, to avoid unnecessary polling. (I consider this to be good design, but any flames (within reason) are welcome if they help me learn otherwise.) private void DequeueForProcessing(object state) { while (true) { ITask task; lock (mutex) { while (underlying.Count == 0) { Monitor.Wait(mutex); } task = underlying.Dequeue(); } Process(task); } } As more operations are added to this class (requiring read-only access to the lock protected underlying), someone suggested using ReaderWriterLockSlim. I've never used the class before, and assuming it can offer some performance benefit, I'm not against it, but only if I can keep the Pulse/Wait design.

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  • Browser Based Streaming Video/Audio (not progressive download)

    - by Josh
    Hello, I am trying to understand conceptually the best way to deliver real streaming audio and video content. I would want it to be consumed with a web browser, utilizing the least amount of proprietary technology. I wouldn't be serving static files and using progressive download, this would be real audio streams being captured live. How does one broadcast a stream that will be reasonably in sync with the source? What kind of protocol is suitable? Edit: In research I've found that there are a few protocols: RTSP, HTTP Streaming, RTMP, and RTP. HTTP streaming is somewhat unsuitable if you are streaming a live performance/communication of some kind because it relies on TCP (as its HTTP based) and you don't lose packets. In a low bandwidth situation, the client can get significantly behind in playback. ref RTMP is a proprietary technology, requiring flash media server. Crap on that. The reason I looked at flash is because they are extremely flexible as far as user experience goes. SoundManager2 provides an excellent javascript interface for playing media with flash. This is what I would look for in a client application. RTSP/RTP is what Microsoft switched to using, deprecating their MMS protocol. RTSP is the control protocol. Its similar to HTTP with a few distinct difference -- server can also talk to the client, and there are additional commands, like PAUSE. Its also a stateful protocol, which is maintained with a session id. RTP is the protocol for delivering the payload (encoded audio or video). There are a few open sourced projects, one of them being supported by apple here. It seems like this might do what I want it to, and it looks like quite a few players support it. It sounds like it would be suitable for a "live" broadcast from this page here. Thanks, Josh

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  • Mix Audio tracks with offset in SOX

    - by Laramie
    From ASP.Net, I am using FFMPEG to convert flv files on a Flash Media Server to wavs that I need to mix into a single MP3 file. I originally attempted this entirely with FFMPEG but eventually gave up on the mixing step because I don't believe it it possible to combine audio only tracks into a single result file. I would love to be wrong. I am now using FFMPEG to access the FLV files and extract the audio track to wav so that SOX can mix them. The problem is that I must offset one of the audio tracks by a few seconds so that they are synchronized. Each file is one half of a conversation between a student and a teacher. For example teacher.wav might need to begin 3.3 seconds after student.wav. I can only figure out how to mix the files with SOX where both tracks begin at the same time. My best attempt at this point is: ffmpeg -y -i rtmp://server/appName/instance/student.flv -ac 1 student.wav ffmpeg -y -i rtmp://server/appName/instance/teacher.flv -ac 1 teacher.wav sox -m student.wav teacher.wav combined.mp3 splice 3.3 These tools (FFMEG/SoX) were chosen based on my best research, but are not required. Any working solution would allow an ASP.Net service to input the two FMS flvs and create a combined MP3 using open-source or free tools.

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  • Technology and language for a stable Digital Audio Workstation development

    - by Kill KRT
    Hi, I'm designing a cross platform (Windows/Linux/OS X) application, something like a digital audio workstation. I'd like to create a software where users have a fully featured sequencer (multiple tracks with automation) and where it is possible to create instruments using a visual language (as Pure Data/Max MSP). Ehm... I know that I've already posted a question about a related issue... But in order to decide which technology I should use, I think I'd better to make more investigation. I'm a quite experted user of audio trackers (Renoise, Protracker,...) and sequencers (FL Studio, Cubase 5), but I didn't ever try to develop even a basic audio tracker. I know just the basic theory of mixing sound and know how basically a DSP works. My questions are: Where I can find a good tutorial/guide/book about this issue? Do you think using C# (with NAudio) could dramatically reduce performance? I know C++ would be the best choice, but I find C# so elegant and easy to build and port, while C++ is so powerful and fast, but there are too #define and bad things for my taste! ;-) Thank you.

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  • waveInProc / Windows audio question...

    - by BTR
    I'm using the Windows API to get audio input. I've followed all the steps on MSDN and managed to record audio to a WAV file. No problem. I'm using multiple buffers and all that. I'd like to do more with the buffers than simply write to a file, so now I've got a callback set up. It works great and I'm getting the data, but I'm not sure what to do with it once I have it. Here's my callback... everything here works: // Media API callback void CALLBACK AudioRecorder::waveInProc(HWAVEIN hWaveIn, UINT uMsg, DWORD dwInstance, DWORD dwParam1, DWORD dwParam2) { // Data received if (uMsg == WIM_DATA) { // Get wav header LPWAVEHDR mBuffer = (WAVEHDR *)dwParam1; // Now what? for (unsigned i = 0; i != mBuffer->dwBytesRecorded; ++i) { // I can see the char, how do get them into my file and audio buffers? cout << mBuffer->lpData[i] << "\n"; } // Re-use buffer mResultHnd = waveInAddBuffer(hWaveIn, mBuffer, sizeof(mInputBuffer[0])); // mInputBuffer is a const WAVEHDR * } } // waveInOpen cannot use an instance method as its callback, // so we create a static method which calls the instance version void CALLBACK AudioRecorder::staticWaveInProc(HWAVEIN hWaveIn, UINT uMsg, DWORD_PTR dwInstance, DWORD_PTR dwParam1, DWORD_PTR dwParam2) { // Call instance version of method reinterpret_cast<AudioRecorder *>(dwParam1)->waveInProc(hWaveIn, uMsg, dwInstance, dwParam1, dwParam2); } Like I said, it works great, but I'm trying to do the following: Convert the data to short and copy into an array Convert the data to float and copy into an array Copy the data to a larger char array which I'll write into a WAV Relay the data to an arbitrary output device I've worked with FMOD a lot and I'm familiar with interleaving and all that. But FMOD dishes everything out as floats. In this case, I'm going the other way. I guess I'm basically just looking for resources on how to go from LPSTR to short, float, and unsigned char. Thanks much in advance!

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  • Audio Streaming Latency

    - by killianmcc
    I'm writing a UDP local area network video chat system and have got the video and audio streams working. However I'm experiencing a little latency (about half a second) in the audio and was wondering what codecs would provide the least latency. I'm using NAudio (http://naudio.codeplex.com/) which provides me access to the following codecs for streaming; Speex Narrow Band (VBR) Speex Wide Band (16kHz)(VBR) Speex Ultra Wide Band (32kHz)(VBR) DSP Group TrueSpeech (8.5kbps) GSM 6.10 (13kbps) Microsoft ADPCM (32.8kbps) G.711 a-law (64kbps) G.722 16kHz (64kbps) G.711 mu-law (64kbps) PCM 8kHz 16 bit uncompressed (128kbps) I've tried them out and I'm not noticing much difference. Is there any others that I should download and try to reduce latency? I'm only going to be sending voice over the connection but I'm not really worried about quality or background noises too much. UPDATE I'm sending the audio in blocks like so; waveIn = new WaveIn(); waveIn.BufferMilliseconds = 50; waveIn.DeviceNumber = inputDeviceNumber; waveIn.WaveFormat = codec.RecordFormat; waveIn.DataAvailable += waveIn_DataAvailable; void waveIn_DataAvailable(object sender, WaveInEventArgs e) { if (connected) { byte[] encoded = codec.Encode(e.Buffer, 0, e.BytesRecorded); udpSender.Send(encoded, encoded.Length); } }

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  • How To Rip an Audio CD to FLAC with Foobar2000

    - by Mysticgeek
    Foobar2000 is a great audio player that is fully customizable, is light on system resources, and contains a lot of tools and features. Today we show you how to use it to rip an audio CD to FLAC format. Note: For this tutorial we’re going to assume this is the first time you’re ripping a disc with Foobar2000. We’re running it on Windows 7 Ultimate 64-bit. Install Foobar2000 and FLAC First download and install Foobar2000 (link below). The main thing you’ll want to make sure to enable during the install process is Audio CD Support… And the freedb Tagger which are located under Optional Features, then continue through the rest of the install wizard. Next you need to install the latest version of the FLAC codec (link below) following the defaults. Rip Audio CD To rip a CD, place it in your CDROM drive, launch Foobar2000 and click File \ Open Audio CD. Select the appropriate CD drive and click the Rip button. Next you’ll want to lookup the disc information with freedb…or you can manually enter in the track data if it’s a custom disc. Select the proper tag information in the freedb tagger window, then click Update files. The data will be entered in, make sure the radio button next to Go to the Converter Setup dialog is selected, and click the Rip button. In the Converter Setup screen, here you can select the output format, where in our case we’re selecting FLAC. In this window you can choose several other options like the output path, merging the tracks into one or individual files…etc. When you have those settings completed click OK. Next you’ll need to find flac.exe which is located wherever you installed it. On our 64-bit Windows 7 system the default path is C:\Program Files (x86)\FLAC Now wait while your CD is ripped and converted to FLAC. You’ll get a Converter Status Report…after you’ve checked it over you can close out of it. If you set the option to show the output files after conversion you can take a look, make sure all tracks were converted, and play them right away if you want. You can play the tracks in Foobar2000 or any player that supports FLAC. If you want to use WMC or WMP see our article on how to play FLAC files in Windows 7 Media Center or Player. That’s all there is to it! If you’re a fan of Foobar2000 and enjoy your music converted to FLAC format, Foobar2000 does the job quite well. There are a lot of customizations and tools you can use in Foobar2000 that we’ll be taking a look at in future articles. For more information check out our look at this fully customizable music player. Foobar2000 run on XP, Vista, and Windows 7 Links Download Foobar2000 Download FLAC Similar Articles Productive Geek Tips Using Ubuntu: What Package Did This File Come From?Easily Change Audio File Formats with XRECODEFoobar2000 is a Fully Customizable Music PlayerConvert Virtually Any Audio Format with XRECODE IIExtract Audio from a Video File with Pazera Free Audio Extractor TouchFreeze Alternative in AutoHotkey The Icy Undertow Desktop Windows Home Server – Backup to LAN The Clear & Clean Desktop Use This Bookmarklet to Easily Get Albums Use AutoHotkey to Assign a Hotkey to a Specific Window Latest Software Reviews Tinyhacker Random Tips DVDFab 6 Revo Uninstaller Pro Registry Mechanic 9 for Windows PC Tools Internet Security Suite 2010 Download Free MP3s from Amazon Awe inspiring, inter-galactic theme (Win 7) Case Study – How to Optimize Popular Wordpress Sites Restore Hidden Updates in Windows 7 & Vista Iceland an Insurance Job? Find Downloads and Add-ins for Outlook

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  • How to make pulseaudio and ubuntu detect the same audio device as alsa driver

    - by Kiwy
    I use Ubuntu 14.04 x64 and I use gnome-shell on my laptop. I have a Bose companion 5 (which is basically a USB sound system) and a HDMI port, both does work perfectly when I just boot with the cable plugin. However, when my laptop go to sleep or get unplugged from those two outputs, if I plug back the device, I end up without any hardware detection (only the built-in speakers) from pulse and gnome-shell sound output selector while if I use alsamixer, the device look up and ready. gstreamer-properties allow me to select and test effectively any device but while alsa recognize any device on the run, pulse is not capable of handling things correctly, my question is then: How can I make pulse detect and use the same hardware as alsa, or how to remove completely and gracefully pulseaudio (meaning volume applet running in gnome shell) I don't mind if the project implies to recompile half gnome shell if it implies those audio outputs work all the time. Pulse does not list my soundcard when I use command pactl list cards while the module plug&play for sound card is loaded in pactl list modules. I really don't know what to do, the behavior seems pretty random.

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