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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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  • Our Flash Streaming Player Occasionally Stutters like a Skipping CD after a Period of Time

    - by Jonathan Fritz
    We offer a streaming player for a number of our clients, who are responsible for their providing us with their own audio streams. We have written a very simple flash player that can play all of the streams that we support (icecast/shoutcast/live365/mp3 over http/etc). Unfortunately, we have found that when listening, our player sometimes begins to stutter (like a skipping cd), sometimes after only 10 minutes, and sometimes after an hour of listening. We have noticed this behaviour in firefox on both linux and windows. Does anybody know anything about this problem? We know that flash isn't ideal for infinite streams of audio, but it's about all that we can find that's on every platform out there. If anybody can suggest a solution to our problem, I'll be your friend forever. Here is a link to the live player: http://cr-jf.jfritz.02.dev.wecreate.com/streaming/player_v5/ Note that you'll need to test in a browser that isn't IE, because we use WMP in IE, and that the JavaScript on the page will cause the player to unload and re-load once an hour because of memory issues. Because I can only put one hyperlink in a post, I'll add a link to the player source code as a comment. Thanks all!

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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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  • Set default system audio output port (for all accounts)

    - by Ludwik Trammer
    The default output audio port Ubuntu doesn't work on my system. It should be "Analog Mono Output/Amplifier", instead of "Analog Output/Amplifier". I can easily change that in sound preferences, just by choosing the right port in the "Output" tab. The problem is this would only apply to a single account, and I would like to change it system-wide, so it applies to all accounts on the system (I have more than 100 users...). I'm after 2 hours of Googling, so any help would be appreciated.

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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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  • Save a view in Windows Media Player

    - by Charles Roper
    I like to view my library in various ways in WMP. For example, I usually search for Podcast and order the result by date added. This gives me a list of my podcasts by date order, newest to oldest. Is there a way of saving this view so that I don't have recreate it each time I open WMP? If it's not possible to do this, can anyone suggest an app that does do it, and that handles syncing as well as WMP.

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  • Stream Music and Video Over the Internet with Windows Media Player 12

    - by DigitalGeekery
    A new feature in Windows Media Player 12, which is included with Windows 7, is being able to stream media over the web to other Windows 7 computers.  Today we will take a look at how to set it up and what you need to begin. Note: You will need to perform this process on each computer that you want to use. What You’ll Need Two computers running Windows 7 Home Premium, Professional, or Ultimate. The host, or home computer that you will be streaming the media from, cannot be on a public network or part of domain. Windows Live ID UPnP or Port Forwarding enabled on your home router Media files added to your Windows Media Player library Windows Live ID Sign up online for a Windows Live ID if you do not already have one. See the link below for a link to Windows Live.   Configuring the Windows 7 Computers Open Windows Media Player and go to the library section. Click on Stream and then “Allow Internet access to home media.”   The Internet Home Media Access pop up window will prompt you to link your Windows Live ID to a user account. Click “Link an online ID.” If you haven’t already installed the Windows Live ID Sign-In Assistant, you will be taken to Microsoft’s website and prompted to download it. Once you have completed the Windows Live download assistant install, you will see Windows Live ID online provider appear in the “Link Online IDs” window. Click on “Link Online ID.” Next, you’ll be prompted for a Windows Live ID and password. Enter your Windows Live ID and password and click “Sign In.” A pop up window will notify you that you have successfully allowed Internet access to home media. Now, you will have to repeat the exact same configuration on the 2nd Windows 7 computer. Once you have completed the same configuration on your 2nd computer, you might also need to configure your home router for port forwarding. If your router supports UPnP, you may not need to manually forward any ports on your router. So, this would be a good time to test your connection. Go to a nearby hotspot, or perhaps a neighbor’s house, and test to see if you can stream your media. If not, you’ll need to manually forward the ports. You can always choose to forward the ports anyway, just in case. Note: We tested on a Linksys WRT54GL router, which supports UPnP, and found we still needed to manually forward the ports. Finding the ports to forward on the router Open Windows Media Player and make sure you are in Library view. Click on “Stream” on the top menu, and select “Allow Internet access to home media.”   On the “Internet Home Media Access” window, click on “Diagnose connections.” The “Internet Streaming Diagnostic Tool” will pop up. Click on “Port forwarding information” near the bottom.   On the “Port Forwarding Information” window you will find both the Internal and External Port numbers you will need to forward on your router. The Internal port number should always be 10245. The external number will be different depending on your computer. Microsoft also recommends forwarding port 443. Configuring the Router Next, you’ll need to configure Port Forwarding on your home router. We will show you the steps for a Linksys WRT54GL router, however, the steps for port forwarding will vary from router to router. On the Linksys configuration page, click on the Administration Tab along the top, click the “Applications & Gaming Tab, and then the “Port Range Forward” tab below it. Under “Application,” type in a name. It can be any name you choose. In both the “Start” and “End” boxes, type the port number. Enter the IP address of your home computer in the IP address column. Click the check box under “Enable.” Do this for both the internal and external port numbers and port 443. When finished, click the “Save Settings” button. Note: It’s highly recommended that you configure your home computer with a static IP address When you’re ready to play your media over the Internet, open up Windows Media Player and look for your host computer and username listed under “Other Libraries.” Click on it expand the list to see your media libraries. Choose a library and a file to play. Now you can enjoy your streaming media over the Internet. Conclusion We found media streaming over the Internet to work fairly well. However, we did see a loss of quality with streaming video. Also, Recorded TV .wtv and dvr-ms files did not play at all. Check out our previous article to see how to stream media share and stream media between Windows 7 computers on your home network. Similar Articles Productive Geek Tips Enable Media Streaming in Windows Home Server to Windows Media PlayerFixing When Windows Media Player Library Won’t Let You Add FilesShare Digital Media With Other Computers on a Home Network with Windows 7Share and Stream Digital Media Between Windows 7 Machines On Your Home NetworkLearning Windows 7: Manage Your Music with Windows Media Player 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 Revo Uninstaller Pro Registry Mechanic 9 for Windows PC Tools Internet Security Suite 2010 PCmover Professional Stormpulse provides slick, real time weather data Geek Parents – Did you try Parental Controls in Windows 7? 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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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  • Improve Playback Using Enhancements in Windows Media Player 12

    - by DigitalGeekery
    Are you looking for ways to improve the playback of your media in Windows Media Player 12? We’ll show you how to do that by using the enhancements in WMP 12. If you are in Library mode, you’ll need to click the icon at the lower right to switch to Now Playing mode. Right-click anywhere in Media Player while in Now Playing mode, select Enhancements, and select any of the available options.   You can switch between the individual enhancements by clicking the right and left buttons at the top left.   Crossfading and Auto Volume Leveling The Auto Volume Leveling setting is just a simple toggle on and off. If your MP3 or WMA files have volume leveling information values.   You can automatically add volume leveling information values to all files you add to your library by switching to Library view, going to Tools > Options, and selecting Add volume leveling information values for new files on the Library tab. Click OK when finished.   Crossfading will gradually decrease the volume of the song that is ending (fade out) and increase volume of the song that is beginning. Click Turn on Crossfading and then click and drag the slider left or right change the amount of overlap between tracks. Graphic Equalizer The graphic equalizer is toggled on and off by clicking Turn on / Turn off at the top left. You can select pre-defined equalizer settings by music genre by clicking the Default list. The radio buttons on the left allow you to move the sliders individually, in a loose group or a tight group. You can always return to the default settings by clicking Reset. Play Speed Settings Choose a pre-defined settings by clicking Slow, Normal, or Fast. Uncheck the Snap slider to common speeds the move the slider right and left to your desired speed. If nothing else, these settings provide a little fun and amusement. Quiet Mode Quiet mode will level out any sharp volume highs and lows within a single track. Simply toggle the setting on or off and select whether you prefer Medium difference or Little difference by selecting one of the radio buttons. SRS WOW effects SRS WOW effects enhance low-frequency and stereo sound performance. Click Turn on to enable the TruBass and WOW Effect sliders. You can also optimize for your speaker type. Click to switch between Regular, Large, and Headphones. Video Settings Video Settings allow you to adjust the Hue, Brightness, Saturation, and Contrast.   You can also adjust the zoom settings by clicking Select video zoom settings.   Dolby Digital Settings Choose between Normal, Night, and Theater settings to adjust the audio for Dolby Digital content. This setting will only effect media with Dolby Digital sound. Looking for more ways to improve your media experience in WMP 12? Check out how to update metadata and cover art and how to share media with other Windows 7 computers on your home network. Similar Articles Productive Geek Tips Fixing When Windows Media Player Library Won’t Let You Add FilesInstall and Use the VLC Media Player on Ubuntu LinuxHow To Rip a Music CD in Windows 7 Media CenterStream Media from Windows 7 to XP with VLC Media PlayerInstalling Windows Media Player Plugin for Firefox 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 Acronis Online Backup DVDFab 6 Revo Uninstaller Pro Registry Mechanic 9 for Windows Check these Awesome Chrome Add-ons iFixit Offers Gadget Repair Manuals Online Vista style sidebar for Windows 7 Create Nice Charts With These Web Based Tools Track Daily Goals With 42Goals Video Toolbox is a Superb Online Video Editor

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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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  • yahoo media player not working.

    - by luca590
    I have a yahoo media player embedded in my webpage. I am currently using Ruby on Rails to create/edit my web page. When i click the play button next to a track the YMP waits a while and then goes to the next track without playing the first one. I then get a warning on my second (last) track that its file could not be found. Does anyone has a better recommendation for an audio player or a way to fix this one?

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  • yahoo media player not working

    - by luca590
    I have a yahoo media player embedded in my webpage. I am currently using Ruby on Rails to create/edit my web page. When i click the play button next to a track the YMP waits a while and then goes to the next track without playing the first one. I then get a warning on my second (last) track that its file could not be found. Does anyone has a better recommendation for an audio player or a way to fix this one?

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  • Yahoo media player not working with Ruby on rails

    - by luca590
    I have a yahoo media player embedded in my webpage. I am currently using Ruby on Rails to create/edit my web page. When i click the play button next to a track the YMP waits a while and then goes to the next track without playing the first one. I then get a warning on my second (last) track that its file could not be found. Does anyone has a better recommendation for an audio player or a way to fix this one?

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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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  • App / protocol to tune into live audio and video based on schedule or subscription

    - by Richard
    Many of us have embraced the podcasting revolution enabled by rss feeds and podcatchers. Alot of sites now broadcast live streams of what is eventually edited into a podcast. In most cases listening to the live stream gets you the info several days sooner then the podcast. So I was wondering if anybody knows of a notification protocol / app that allows me to auto tune into certain streams when they go live, or based on a schedule. I imagine twitter could be used for the notification but It'd be better not to be tied to a proprietary service. Example podcasts / live streams noagenda.squarespace.com jupiterbroadcasting.com twit.tv

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  • jQuery Audio Player

    - by tony noriega
    I was given 2 MP3 files, one that is 4.5Mb and one that is 5.6Mb. I was instructed to have them play on a website i am managing. I have found a nice, clean looking CSS based jQuery audio player. My question is, is this the right solution for files that big? I am not sure if the player preloads the file, or streams it ? (if that is the correct terminology) i dont deal much with audio players and such... this player is from happyworm.com/jquery/jplayer/latest/demo-01.htm is there another approach i shoudl take to get this to play properly? I dont want it to have to buffer, and the visitor to wait, or slow page loading...etc..etc.. i want it to play clean and not affect the visitors session to the site. thanks

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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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