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  • aplay -l says no soundcards found; alsaconf says no supported cords; yet /proc/asound contains cards

    - by nimasmi
    I am trying to get HDMI output using a Gainward Nvidia 210 512 MB on Ubuntu 10.04 Lucid Lynx. I have upgraded alsa-driver, alsa-lib and alsa-utils to 1.0.24 by building from source, thanks to this blog post. Some relevant output... user@box:~$ lspci | grep Audio 00:05.0 Audio device: nVidia Corporation MCP61 High Definition Audio (rev a2) 01:09.0 Multimedia video controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder (rev 05) 01:09.2 Multimedia controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder [MPEG Port] (rev 05) 01:09.4 Multimedia controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder [IR Port] (rev 05) 02:00.1 Audio device: nVidia Corporation High Definition Audio Controller (rev a1) user@box:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.24. Compiled on Sep 15 2012 for kernel 2.6.32-42-generic (SMP). user@box:~$ ls /proc/asound` card0 cards hwdep NVidia oss seq version card1 devices modules NVidia_1 pcm timers user@box:~$ aplay -l aplay: device_list:240: no soundcards found... user@box:~$ sudo /sbin/alsa-utils start * Setting up ALSA... * warning: 'alsactl restore' failed with error message 'alsactl: set_control:1403: Cannot write control '2:0:0:IEC958 Playback Default:0' : Operation not permitted'... amixer: Invalid command! ...done. Any help appreciated. PS my video card is connected only through the PCI-E slot. I assume there is no extra audio connection required.

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  • NAudio demos not working anymore

    - by Kurru
    I just tried to run the NAudio demos and I'm getting a weird error: System.BadImageFormatException: Could not load file or a ssembly 'NAudio, Version=1.3.8.0, Culture=neutral, PublicKeyToken=null' or one o f its dependencies. An attempt was made to load a program with an incorrect form at. File name: 'NAudio, Version=1.3.8.0, Culture=neutral, PublicKeyToken=null' at NAudioWpfDemo.AudioGraph..ctor() at NAudioWpfDemo.ControlPanelViewModel..ctor(IWaveFormRenderer waveFormRender er, SpectrumAnalyser analyzer) in C:\Users\Admin\Downloads\NAudio-1.3\NAudio-1-3 \Source Code\NAudioWpfDemo\ControlPanelViewModel.cs:line 23 at NAudioWpfDemo.MainWindow..ctor() in C:\Users\Admin\Downloads\NAudio-1.3\NA udio-1-3\Source Code\NAudioWpfDemo\MainWindow.xaml.cs:line 15 WRN: Assembly binding logging is turned OFF. To enable assembly bind failure logging, set the registry value [HKLM\Software\M icrosoft\Fusion!EnableLog] (DWORD) to 1. Note: There is some performance penalty associated with assembly bind failure lo gging. To turn this feature off, remove the registry value [HKLM\Software\Microsoft\Fus ion!EnableLog]. Since the last time I used NAudio demos I have changed from 32bit Windows XP to 64bit Windows 7. Would this cause this issue? Its very annoying as I was about to try my hand at audio in C# again

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  • Silverlight MediaElement Position Property Weirdness

    - by BarrettJ
    I have a MediaElement that is reporting its position incorrectly and weirdly, but consistently. It seems like when it gets to the last second of the audio (and it's always the last second, regardless if the sound is two seconds or 10), it doesn't update it's position until it finishes. Example output: Playback Progress: 0/3.99 - 0 Playback Progress: 0.01/3.99 - 0 Playback Progress: 0.03/3.99 - 0 Playback Progress: 0.06/3.99 - 1 Playback Progress: 0.07/3.99 - 1 Playback Progress: 0.08/3.99 - 2 Playback Progress: 0.11/3.99 - 2 Playback Progress: 0.14/3.99 - 3 Playback Progress: 0.19/3.99 - 4 Playback Progress: 0.23/3.99 - 5 Playback Progress: 0.25/3.99 - 6 Playback Progress: 0.28/3.99 - 7 Playback Progress: 0.3/3.99 - 7 Playback [SNIP] Playback Progress: 2.8/3.99 - 70 Playback Progress: 2.83/3.99 - 70 Playback Progress: 2.88/3.99 - 72 Playback Progress: 2.9/3.99 - 72 Playback Progress: 2.91/3.99 - 72 Playback Progress: 2.92/3.99 - 73 Playback Progress: 2.99/3.99 - 74 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3.99/3.99 - 100 That is the result of: WriteLine("Playback Progress: " + Position + "/" + LengthInSeconds + " - " + (int)((Position / LengthInSeconds) * 100)); public double Position { get { return my_media_element != null ? my_media_element.Position.TotalSeconds : 0; } } public double LengthInSeconds { get { return my_media_element != null ? my_media_element.NaturalDuration.TimeSpan.TotalSeconds : 0; } } Anyone have any ideas why this is occurring?

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  • How to get a volume measurement of iPhone recording in dB, with a limit of at least 120dB

    - by Cyber
    Hello, I am trying to make a simple volume meter for the iPhone. I want the volume displayed in dB. When using this turorial, I am only getting measurements up to 78 dB. I've read that that is because the dBFS spectrum for 16 bit audio recordings is only 96 dB. I tried modifying this piece of code in the init funcyion: dataFormat.mSampleRate = 44100.0f; dataFormat.mFormatID = kAudioFormatLinearPCM; dataFormat.mFramesPerPacket = 1; dataFormat.mChannelsPerFrame = 1; dataFormat.mBytesPerFrame = 2; dataFormat.mBytesPerPacket = 2; dataFormat.mBitsPerChannel = 16; dataFormat.mReserved = 0; I changed the value of mBitsPerChannel, hoping to increase the bit value of the recording. dataFormat.mBitsPerChannel = 32; With that variable set to 32, the "mAveragePower" function returns only 0. So, how can i measure more decibels? All my code is practically the same as in the tutorial i posted above. Thanks in advance, Thomas

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  • Can't play wav file from Javascript in Firefox for Mac

    - by Mike Royle
    I have the following html file that plays a wav file when the user hovers over the 'Play' anchor tag. It works perfectly on IE, Chrome, Firefox, Opera, Safari on both Windows and Mac - except for Firefox on the Mac which does not play the file. <html> <head> <title></title> <script> function PlayAudio() { var s = document.getElementById("soundFile"); s.Play(); } </script> </head> <body> <embed src="MySound.wav" enablejavascript="true" type="audio/wav" autostart="false" width="0" height="0" id="soundFile" /> <a href="#" onmouseover="PlayAudio()">Play</a> </body> </html> If the autostart attribute of the embed tag is set to true then the wav file plays as expected in Firefox for Mac, but not on the mouseover of the anchor tag. Any ideas?

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  • Debug NAudio MP3 reading difference?

    - by Conrad Albrecht
    My code using NAudio to read one particular MP3 gets different results than several other commercial apps. Specifically: My NAudio-based code finds ~1.4 sec of silence at the beginning of this MP3 before "audible audio" (a drum pickup) starts, whereas other apps (Windows Media Player, RealPlayer, WavePad) show ~2.5 sec of silence before that same drum pickup. The particular MP3 is "Like A Rolling Stone" downloaded from Amazon.com. Tested several other MP3s and none show any similar difference between my code and other apps. Most MP3s don't start with such a long silence so I suspect that's the source of the difference. Debugging problems: I can't actually find a way to even prove that the other apps are right and NAudio/me is wrong, i.e. to compare block-by-block my code's results to a "known good reference implementation"; therefore I can't even precisely define the "error" I need to debug. Since my code reads thousands of samples during those 1.4 sec with no obvious errors, I can't think how to narrow down where/when in the input stream to look for a bug. The heart of the NAudio code is a P/Invoke call to acmStreamConvert(), which is a Windows "black box" call which I can't think how to error-check. Can anyone think of any tricks/techniques to debug this?

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  • How can I implement a volume meter for a song currently playing? (iPhone OS 3.1.3)

    - by Adam
    Hi i'm very new to core audio and I just would like some help in coding up a little volume meter for whatever's being outputted through headphones or built-in speaker. Like a dB meter. I have the following code, and have been trying to go through the apple source project "SpeakHere", but it's a nightmare trying to go through all that, without knowing how it works first... Could anyone shed some light? Here's the code I have so far... (void)displayWaveForm { while (musicIsPlaying == YES { NSLog(@"%f",sizeof(AudioQueueLevelMeterState)); } } (IBAction)playMusic { if (musicIsPlaying == NO) { NSURL *url = [NSURL fileURLWithPath:[NSString stringWithFormat:@"%@/track7.wav",[[NSBundle mainBundle] resourcePath]]]; NSError *error; music = [[AVAudioPlayer alloc] initWithContentsOfURL:url error:&error]; music.numberOfLoops = -1; music.volume = 0.5; [music play]; musicIsPlaying = YES; [self displayWaveForm]; } else { [music pause]; musicIsPlaying = NO; } }

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  • How would i down-sample a .wav file then reconstruct it using nyquist? - in MATLAB

    - by Andrew
    This is all done in MATLAB 2010 My objective is to show the results of: undersampling, nyquist rate/ oversampling First i need to downsample the .wav file to get an incomplete/ or impartial data stream that i can then reconstuct. Heres the flow chart of what im going to be doing So the flow is analog signal - sampling analog filter - ADC - resample down - resample up - DAC - reconstruction analog filter what needs to be achieved: F= Frequency F(Hz=1/s) E.x. 100Hz = 1000 (Cyc/sec) F(s)= 1/(2f) Example problem: 1000 hz = Highest frequency 1/2(1000hz) = 1/2000 = 5x10(-3) sec/cyc or a sampling rate of 5ms This is my first signal processing project using matlab. what i have so far. % Fs = frequency sampled (44100hz or the sampling frequency of a cd) [test,fs]=wavread('test.wav'); % loads the .wav file left=test(:,1); % Plot of the .wav signal time vs. strength time=(1/44100)*length(left); t=linspace(0,time,length(left)); plot(t,left) xlabel('time (sec)'); ylabel('relative signal strength') **%this is were i would need to sample it at the different frequecys (both above and below and at) nyquist frequency.*I think.*** soundsc(left,fs) % shows the resaultant audio file , which is the same as original ( only at or above nyquist frequency however) Can anyone tell me how to make it better, and how to do the sampling at verious frequencies?

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  • OpenAL not playing on Max OS X 10.6

    - by Grimless
    I've been working on getting a basic audio engine running on my Mac using OpenAL. It seems relatively straightforward after working with OpenGL for a while. However, despite the fact that I believe I have everything in place, my sound will not play. Here is the order of things I am doing: //Creating a new device ALCdevice* device = alcOpenDevice(NULL); //Create a new context with the device ALCcontext* context = alcCreateContext(device, NULL); //Make that context current alcMakeContextCurrent(context); //Do lots of loading stuff to bring in an AIFF... voodooAIFF = myAIFFLoader("name"); //Then use that data ALuint buf; alGenBuffers(1, &buf); //Check for errors, but none happen... //Bind buffer data. alBufferData(buf, voodooAIFF.format, voodooAIFF.data, voodooAIFF.sizeInBytes, voodooAIFF.frequency); //Check for errors, none here either... //Create Source ALuint src; alGenSources(1, &src); //Error check again, no errors. //Bind source to buffer alSourcei(src, AL_BUFFER, buf); //Set reference distance alSourcei(sourceID, AL_REFERENCE_DISTANCE, 1); //Set source attributes including gain and pitch to 1 (direction set to 0,0,0) //Check for errors, nothing... //Set up listener attributes. //Check for errors, no errors. //Begin playing. alSourcePlay(src); Observe silence... Any insight, what steps am I missing here?

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  • Difficulty porting raw PCM output code from Java to Android AudioTrack API.

    - by IndigoParadox
    I'm attempting to port an application that plays chiptunes (NSF, SPC, etc) music files from Java SE to Android. The Android API seems to lack the javax multimedia classes that this application uses to output raw PCM audio. The closest analog I've found in the API is AudioTrack and so I've been wrestling with that. However, when I try to run one of my sample music files through my port-in-progress, all I get back is static. My suspicion is that it's the AudioTrack I've setup which is at fault. I've tried various different constructors but it all just outputs static in the end. The DataLine setup in the original code is something like: AudioFormat audioFormat = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED, 44100, 16, 2, 4, 44100, true ); DataLine.Info lineInfo = new DataLine.Info( SourceDataLine.class, audioFormat ); DataLine line = (SourceDataLine)AudioSystem.getLine( lineInfo ); The constructor I'm using right now is: AudioTrack = new AudioTrack( AudioManager.STREAM_MUSIC, 44100, AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT, AudioTrack.getMinBufferSize( 44100, AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT ), AudioTrack.MODE_STREAM ); I've replaced constants and variables in those so they make sense as concisely as possible, but my basic question is if there are any obvious problems in the assumptions I made when going from one format to the other.

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  • Is there a best practice for concatenating MP3 Files, adjusting sample rates to match, while preserving original files?

    - by Scott
    Hello overflow community! Does anyone know if there is a "best practice" to concatenate mp3 files to create new files, while preserving the original files? I am working on a CentOS Linux machine, in command line. I will eventually call the command line from a PHP script. I have been doing research and I have come up with a process that I think could work. It combines general advice from different forums, blogs, and sources like this one. So here I go: Create a temporary folder Loop through files to create a new, converted copy, of file into a "raw" format (which one, I don't know. I didn't know "raw" files existed before too long ago. I could use some suggestions on this) Store the path to the temporary files, in the temporary folder, and then loop through the files to concatenate them and then put the new merged file the final "processed directory" Delete the contents of the temporary file with the temporary raw files inside. Convert the final file from "raw" to mp3 and enjoy the finished result I'm thinking that this course of action might be best because I can't necessarily control the quality of the original "source" mp3s. The only other option I could think of would be to create a script that would perform a similar process upon files being added to the system leaving only the files with the "proper" format and removing the original "erroneous" file. Hopefully you can see that I have put some thought into this and that I'm trying to leverage the collective knowledge of this community to choose the best direction. Perhaps there is a better path that I could take? By concatenate, I mean to join together in sequence to create a new audio file from the "concatenated files."

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  • Problems with video conversions through the web (local host)

    - by ron-d
    Hello, I get the following errors when I attempt video format conversions called from the local host: “An invalid media type was specified” for M4V to WMV conversions. “One or more arguments are invalid” for MP4 to WMV conversions. Here are the details of the problems: I’ve written a dll in C# that accepts videos in the formats AVI, WMV, M4V and MP4 and performs the following actions: Creates a copy of the input video in WMV format . Creates a WAV file of the input video audio portion. Creates a JPG image from a frame of the input video. I attached the dll to an ASP.NET web project that performs the dll actions. When tested through the developer studio, the actions are performed as intended for all formats. When I place the web project in place to be read when the local host is called through the web browser, the following behavior takes place: WMV format: All actions performed as intended. AVI format: Creates WMV file – OK Creates JPG image – OK Creates empty WAV file – problem. M4V format: Creates empty WAV file – problem. Does not create WMV file -problem Does not create JPG file –problem Throws me the error “An invalid media type was specified” MP4 format: Creates empty WAV file – problem. Does not create WMV file -problem Does not create JPG file –problem Throws me the error “One or more arguments are invalid” When I check their security property, all the files have the same permission access parameters (when I check their security property. Can anyone guide me as to how to solve these problems when the web project is called from the local host? Thank you.

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  • To display an album art from media store in android

    - by user1834724
    I'm not able to display album art from media store while listing albums,I'm getting following error Bad request for field slot 0,-1. numRows = 32, numColumns = 7 01-02 02:48:16.789: D/AndroidRuntime(4963): Shutting down VM 01-02 02:48:16.789: W/dalvikvm(4963): threadid=1: thread exiting with uncaught exception (group=0x4001e578) 01-02 02:48:16.804: E/AndroidRuntime(4963): FATAL EXCEPTION: main 01-02 02:48:16.804: E/AndroidRuntime(4963): java.lang.IllegalStateException: get field slot from row 0 col -1 failed Can anyone kindly help with this issue,Thanks in advance public class AlbumbsListActivity extends Activity { private ListAdapter albumListAdapter; private HashMap<Integer, Integer> albumInfo; private HashMap<Integer, Integer> albumListInfo; private HashMap<Integer, String> albumListTitleInfo; private String audioMediaId; private static final String TAG = "AlbumsListActivity"; Boolean showAlbumList = false; Boolean AlbumListTitle = false; ImageView album_art ; public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.albums_list_layout); Cursor cursor; ContentResolver cr = getApplicationContext().getContentResolver(); if (getIntent().hasExtra(Util.ALBUM_ID)) { int albumId = getIntent().getIntExtra(Util.ALBUM_ID, Util.MINUS_ONE); String[] projection = new String[] { Albums._ID, Albums.ALBUM, Albums.ARTIST, Albums.ALBUM_ART, Albums.NUMBER_OF_SONGS }; String selection = null; String[] selectionArgs = null; String sortOrder = Media.ALBUM + " ASC"; cursor = cr.query(Albums.EXTERNAL_CONTENT_URI, projection, selection, selectionArgs, sortOrder); /* final String[] ccols = new String[] { //MediaStore.Audio.Albums., MediaStore.Audio.Albums._ID, MediaStore.Audio.Albums.ALBUM, MediaStore.Audio.Albums.ARTIST, MediaStore.Audio.Albums.ALBUM_ART, MediaStore.Audio.Albums.NUMBER_OF_SONGS }; cursor = cr.query(MediaStore.Audio.Albums.getContentUri( "external"), ccols, null, null, MediaStore.Audio.Albums.DEFAULT_SORT_ORDER);*/ showAlbumList = true; } else { String order = MediaStore.Audio.Albums.ALBUM + " ASC"; String where = MediaStore.Audio.Albums.ALBUM; cursor = managedQuery(Media.EXTERNAL_CONTENT_URI, DbUtil.projection, null, null, order); showAlbumList = false; } albumInfo = new HashMap<Integer, Integer>(); albumListInfo = new HashMap<Integer, Integer>(); ListView listView = (ListView) findViewById(R.id.mylist_album); listView.setFastScrollEnabled(true); listView.setOnItemLongClickListener(new ItemLongClickListener()); listView.setAdapter(new AlbumCursorAdapter(this, cursor, DbUtil.displayFields, DbUtil.displayViews,showAlbumList)); final Uri uri = MediaStore.Audio.Albums.EXTERNAL_CONTENT_URI; final Cursor albumListCursor = cr.query(uri, DbUtil.Albumprojection, null, null, null); } private class AlbumCursorAdapter extends SimpleCursorAdapter implements SectionIndexer{ private final Context context; private final Cursor cursorValues; private Time musicTime; private Boolean isAlbumList; private MusicAlphabetIndexer mIndexer; private int mTitleIdx; public AlbumCursorAdapter(Context context, Cursor cursor, String[] from, int[] to,Boolean isAlbumList) { super(context, 0, cursor, from, to); this.context = context; this.cursorValues = cursor; //musicTime = new Time(); this.isAlbumList = isAlbumList; } String albumName=""; String artistName = ""; String numberofsongs = ""; long albumid; @Override public View getView(int position, View convertView, ViewGroup parent) { View rowView = convertView; if (rowView == null) { LayoutInflater inflater = (LayoutInflater) context .getSystemService(Context.LAYOUT_INFLATER_SERVICE); rowView = inflater .inflate(R.layout.row_album_layout, parent, false); } this.cursorValues.moveToPosition(position); String title = ""; String artistName = ""; String albumName = ""; int count; long albumid = 0; String songDuration = ""; if (isAlbumList) { albumInfo.put( position, Integer.parseInt(this.cursorValues.getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Albums._ID)))); artistName = this.cursorValues .getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Albums.ARTIST)); albumName = this.cursorValues .getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Albums.ALBUM)); albumid=Integer.parseInt(this.cursorValues.getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Albums.ALBUM_ID))); } else { albumInfo.put(position, Integer.parseInt(this.cursorValues .getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Media._ID)))); artistName = this.cursorValues.getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Media.ARTIST)); albumName = this.cursorValues.getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Media.ALBUM)); albumid=Integer.parseInt(this.cursorValues.getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Media.ALBUM_ID))); } //code for Alphabetical Indexer mTitleIdx = cursorValues.getColumnIndex(MediaStore.Audio.Media.ALBUM); mIndexer = new MusicAlphabetIndexer(cursorValues, mTitleIdx, getResources().getString(R.string.fast_scroll_alphabet)); //end TextView metaone = (TextView) rowView.findViewById(R.id.album_name); TextView metatwo = (TextView) rowView.findViewById(R.id.artist_name); ImageView metafour = (ImageView) rowView.findViewById(R.id.album_art); TextView metathree = (TextView) rowView .findViewById(R.id.songs_count); metaone.setText(albumName); metatwo.setText(artistName); (metafour)getAlbumArt(albumid); System.out.println("albumid----------"+albumid); metaThree.setText(DbUtil.makeTimeString(context, secs)); getAlbumArt(albumid); } TextView metaone = (TextView) rowView.findViewById(R.id.album_name); TextView metatwo = (TextView) rowView.findViewById(R.id.artist_name); album_art = (ImageView) rowView.findViewById(R.id.album_art); //TextView metathree = (TextView) rowView.findViewById(R.id.songs_count); metaone.setText(albumName); metatwo.setText(artistName); return rowView; } } String albumArtUri = ""; private void getAlbumArt(long albumid) { Uri uri=ContentUris.withAppendedId(MediaStore.Audio.Albums.EXTERNAL_CONTENT_URI, albumid); System.out.println("hhhhhhhhhhh" + uri); Cursor cursor = getContentResolver().query( ContentUris.withAppendedId( MediaStore.Audio.Albums.EXTERNAL_CONTENT_URI, albumid), new String[] { MediaStore.Audio.AlbumColumns.ALBUM_ART }, null, null, null); if (cursor.moveToFirst()) { albumArtUri = cursor.getString(0); } System.out.println("kkkkkkkkkkkkkkkkkkk :" + albumArtUri); cursor.close(); if(albumArtUri != null){ Options opts = new Options(); opts.inJustDecodeBounds = true; Bitmap albumCoverBitmap = BitmapFactory.decodeFile(albumArtUri, opts); opts.inJustDecodeBounds = false; albumCoverBitmap = BitmapFactory.decodeFile(albumArtUri, opts); if(albumCoverBitmap != null) album_art.setImageBitmap(albumCoverBitmap); }else { // TODO: Options opts = new Options(); Bitmap albumCoverBitmap = BitmapFactory.decodeResource(getApplicationContext().getResources(), R.drawable.albumart_mp_unknown_list, opts); if(albumCoverBitmap != null) album_art.setImageBitmap(albumCoverBitmap); } } } }

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  • how to convert video from one format to another using php

    - by Meena
    hi i want to include the vedio download option in my webpage. I am using ffmpeg, but it seems to work very slow. Is there is any other way to do this or how to spead up the ffmpeg. i am using this code to get the frames from the vedio. to convert the vedio $call="ffmpeg -i ".$_SESSION['video_to_convert']." -vcodec libvpx -r 30 -b ".$quality." -acodec libvorbis -ab 128000 -ar ".$audio." -ac 2 -s ".$size." ".$converted_vids.$name.".".$type." -y 2> log/".$name.".txt"; $convert = (popen("start /b ".$call, "r")); pclose($convert); to get the frame from the vedio exec("ffmpeg -vframes 1 -ss ".$time_in_seconds." -i $converted_vids video_images.jpg -y 2>); but this code does not generate any error its loading continously.

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  • How can I send audio input as chunked HTTP?

    - by Noli
    I am trying to create an interface with an external server, and don't know where to start. I would need to take audio as input to my computer, and send it to the remote server as a chunked HTTP request. The api that i'm trying to connect to is described here p1-5 http://dragonmobile.nuancemobiledeveloper.com/public/Help/HttpInterface/HTTP_Services_for_NDEV_v1.2_Silver_Version.pdf I have never worked with audio programmatically, so don't know what would be the most straighforward way to go about this? Are there solutions that exist out there that already do this? I've come across references to Shoutcast, VLC, Icecast, FFMPeg, Darkice, but I don't know if those are appropriate for what I'm trying to accomplish or not. Would appreciate any guidance, Thanks

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  • Can Google Translate's audio files be used in a game?

    - by ashes999
    For my game, I need text-to-speech. Since it's Android, I decided to settle for MP3s, since the range of words spoken is few. For my prototype, I'm using Google Translate to generate the audio since it has awesome pronounciation across multiple languages. But can I use it in production? What if I sell my game for $1 on the app store? All I can find on SE is that the API may be LGPL, and that the licensing page mentions the API is only available for academic research -- nothing more. My usage is a bit different; I'm actually capturing the audio bits and using those instead. I'm curious to know the license for this; I can't find anything with my Google-fu.

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  • CPU spikes cause audio stuttering in Audacious when browsing? (Lubuntu)

    - by Alucai Vivorvel
    My default audio player is Audacious, browser Google Chrome. I tried Firefox, and while I love it, the CPU load spikes when doing something as simple and small and switching a tab, which causes the audio playing to stutter (as sound is onboard and handled thru the CPU). Chrome doesn't do this as much, but there is the occasional stuttering when browsing, which is ridiculous, as not even Windows Vista does this. So I thought maybe it's something to do with how Lubuntu handles sound, I checked and only ALSA was installed. I tried installing PulseAudio, but, while the music "plays", nothing comes through the speakers. Immediately after switching back to ALSA the music pours out of them. So I was wondering if you had any idea what was going on here. I asked on Ubuntu Forums but apparently my problem is too complex, as it's been over a week since the last reply. Specs are: AMD Athlon 64 3200+ @ 2GHz 2GB Corsair 667MHz DDR2 RAM ATi HD Radeon 3650 (AGP) 512MB 500W Cooler Master PSU 80GB SATA II HDD (Vista is installed on 500GB drive) Biostar K8M800 Motherboard

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  • Problems upgrading VB.Net 2008 project into VS2010

    - by Brett Rigby
    Hi there, I have been upgrading several different VS2008 projects into VS2010 and have found a problem with VB.Net projects when they are converted. Once converted, the .vbproj files have changed from this in VS2008: <PropertyGroup Condition=" '$(Configuration)|$(Platform)' == 'Debug|AnyCPU' "> <DebugSymbols>true</DebugSymbols> <DebugType>full</DebugType> <DefineDebug>true</DefineDebug> <DefineTrace>true</DefineTrace> <OutputPath>bin\Debug\</OutputPath> <DocumentationFile>CustomerManager.xml</DocumentationFile> <WarningsAsErrors>41999,42016,42017,42018,42019,42020,42021,42022,42032,42036</WarningsAsErrors> </PropertyGroup> To this in VS2010: <PropertyGroup Condition=" '$(Configuration)|$(Platform)' == 'Debug|AnyCPU' "> <DebugSymbols>true</DebugSymbols> <DebugType>full</DebugType> <DefineDebug>true</DefineDebug> <DefineTrace>true</DefineTrace> <OutputPath>bin\Debug\</OutputPath> <DocumentationFile>CustomerManager.xml</DocumentationFile> <NoWarn>42353,42354,42355</NoWarn> <WarningsAsErrors>41999,42016,42017,42018,42019,42020,42021,42022,42032,42036</WarningsAsErrors> </PropertyGroup> The main difference, is that in the VS2010 version, the 42353,42354,42355 value has been added; Inside the IDE, this manifests itself as the following setting in the Project Properties | Compile section as: "Function returning intrinsic value type without return value" = None This isn't a problem when building code inside Visual Studio 2010, but when trying to build the code through our continuous integration scripts, it fails with the following errors: [msbuild] vbc : Command line error BC2026: warning number '42353' for the option 'nowarn' is either not configurable or not valid [msbuild] vbc : Command line error BC2026: warning number '42354' for the option 'nowarn' is either not configurable or not valid [msbuild] vbc : Command line error BC2026: warning number '42355' for the option 'nowarn' is either not configurable or not valid I couldn't find anything on Google for these messages, which is strange, as I am trying to find out why this is happening. Any suggestions as to why Visual Studio 2010's conversion wizard is doing this?

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  • How to overlay audio file on .wmv video file using c#?

    - by Vipul jain
    Hello, I want to record video and audio files using C#. After recording of audio + video i want to merge them. There can be only one video file and 10 audio file. I want this ten files to overlay on one video file. I am assure that i want video file in .wmv format. Can you tell me i should record audios in which format so later i can overlay those audio files on .wmv format video file? Also please let me know how to overlay audio file on .wmv video file? Hope i will get prompt reply for this

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  • Can anyone help convert this VB Webservice?

    - by CraigJSte
    I can't figure it out for the life of me.. I'm using SQL DataSet Query to iterate to a Class Object which acts as a Data Model for Flex... Initially I used VB.net but now need to convert to C#.. This conversion is done except for the last section where I create a DataRow arow and then try to add the DataSet Values to the Class (Results Class)... I get an error message.. 'VTResults.Results.Ticker' is inaccessible due to its protection level (this is down at the bottom) using System; using System.Web; using System.Collections; using System.Web.Services; using System.Web.Services.Protocols; using System.Data; using System.Data.SqlClient; using System.Configuration; /// <summary> /// Summary description for VTResults /// </summary> [WebService(Namespace = "http://velocitytrading.net/ResultsVT.aspx")] [WebServiceBinding(ConformsTo = WsiProfiles.BasicProfile1_1)] public class VTResults : System.Web.Services.WebService { public class Results { string Ticker; string BuyDate; decimal Buy; string SellDate; decimal Sell; string Profit; decimal Period; } [WebMethod] public Results[] GetResults() { string conn = ConfigurationManager.ConnectionStrings["LocalSqlServer"].ConnectionString; SqlConnection myconn = new SqlConnection(conn); SqlCommand mycomm = new SqlCommand(); SqlDataAdapter myda = new SqlDataAdapter(); DataSet myds = new DataSet(); mycomm.CommandType = CommandType.StoredProcedure; mycomm.Connection = myconn; mycomm.CommandText = "dbo.Results"; myconn.Open(); myda.SelectCommand = mycomm; myda.Fill(myds); myconn.Close(); myconn.Dispose(); int i = 0; Results[] dts = new Results[myds.Tables[0].Rows.Count]; foreach(DataRow arow in myds.Tables[0].Rows) { dts[i] = new Results(); dts[i].Ticker = arow["Ticker"]; dts[i].BuyDate = arow["BuyDate"]; dts[1].Buy = arow["Buy"]; dts[i].SellDate = arow["SellDate"]; dts[i].Sell = arow["Sell"]; dts[i].Profit = arow["Profit"]; dts[i].Period = arow["Period"]; i+=1; } return dts; } } The VB.NET WEBSERVICE that runs fine which I am trying to convert to C# is here. Imports System.Web Imports System.Web.Services Imports System.Web.Services.Protocols Imports System.Data Imports System.Data.SqlClient <WebService(Namespace:="http://localhost:2597/Results/ResultsVT.aspx")> _ <WebServiceBinding(ConformsTo:=WsiProfiles.BasicProfile1_1)> _ <Global.Microsoft.VisualBasic.CompilerServices.DesignerGenerated()> _ Public Class VTResults Inherits System.Web.Services.WebService Public Class Results Public Ticker As String Public BuyDate As String Public Buy As Decimal Public SellDate As String Public Sell As Decimal Public Profit As String Public Period As Decimal End Class <WebMethod()> _ Public Function GetResults() As Results() Try Dim conn As String = ConfigurationManager.ConnectionStrings("LocalSqlServer").ConnectionString Dim myconn = New SqlConnection(conn) Dim mycomm As New SqlCommand Dim myda As New SqlDataAdapter Dim myds As New DataSet mycomm.CommandType = CommandType.StoredProcedure mycomm.Connection = myconn mycomm.CommandText = "dbo.Results" myconn.Open() myda.SelectCommand = mycomm myda.Fill(myds) myconn.Close() myconn.Dispose() Dim i As Integer i = 0 Dim dts As Results() = New Results(myds.Tables(0).Rows.Count - 1) {} Dim aRow As DataRow For Each aRow In myds.Tables(0).Rows dts(i) = New Results dts(i).Ticker = aRow("Ticker") dts(i).BuyDate = aRow("BuyDate") dts(i).Buy = aRow("Buy") dts(i).SellDate = aRow("SellDate") dts(i).Sell = aRow("Sell") dts(i).Profit = aRow("Profit") dts(i).Period = aRow("Period") i += 1 Next Return dts Catch ex As DataException Throw ex End Try End Function End Class

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  • Beat Detection on iPhone with wav files and openal

    - by Dmacpro
    Using this website i have tried to make a beat detection engine. http://www.gamedev.net/reference/articles/article1952.asp { ALfloat energy = 0; ALfloat aEnergy = 0; ALint beats = 0; bool init = false; ALfloat Ei[42]; ALfloat V = 0; ALfloat C = 0; ALshort *hold; hold = new ALshort[[myDat length]/2]; [myDat getBytes:hold length:[myDat length]]; ALuint uiNumSamples; uiNumSamples = [myDat length]/4; if(alDatal == NULL) alDatal = (ALshort *) malloc(uiNumSamples*2); if(alDatar == NULL) alDatar = (ALshort *) malloc(uiNumSamples*2); for (int i = 0; i < uiNumSamples; i++) { alDatal[i] = hold[i*2]; alDatar[i] = hold[i*2+1]; } energy = 0; for(int start = 0; start<(22050*10); start+=512){ //detect for 10 seconds of data for(int i = start; i<(start+512); i++){ energy+= fabs(alDatal[i]) + fabs(alDatar[i]); } aEnergy = 0; for(int i = 41; i>=0; i--){ if(i ==0){ Ei[0] = energy; } else { Ei[i] = Ei[i-1]; } if(start >= 21504){ aEnergy+=Ei[i]; } } aEnergy = aEnergy/43.f; if (start >= 21504) { for(int i = 0; i<42; i++){ V += (Ei[i]-aEnergy); } V = V/43.f; C = (-0.0025714*V)+1.5142857; init = true; if(energy >(C*aEnergy)) beats++; } } } alDatal and alDatar are (short*) type; myDat is NSdata that holds the actual audio data of a wav file formatted to 22050 khz and 16 bit stereo. This doesn't seem to work correctly. If anyone could help me out that would be amazing. I've been stuck on this for 3 days.

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  • C# Can anyone help convert this VB Webservice?

    - by CraigJSte
    I can't figure it out for the life of me.. I'm using SQL DataSet Query to iterate to a Class Object which acts as a Data Model for Flex... Initially I used VB.net but now need to convert to C#.. This conversion is done except for the last section where I create a DataRow arow and then try to add the DataSet Values to the Class (Results Class)... I get an error message.. 'VTResults.Results.Ticker' is inaccessible due to its protection level (this is down at the bottom) using System; using System.Web; using System.Collections; using System.Web.Services; using System.Web.Services.Protocols; using System.Data; using System.Data.SqlClient; using System.Configuration; /// <summary> /// Summary description for VTResults /// </summary> [WebService(Namespace = "http://velocitytrading.net/ResultsVT.aspx")] [WebServiceBinding(ConformsTo = WsiProfiles.BasicProfile1_1)] public class VTResults : System.Web.Services.WebService { public class Results { string Ticker; string BuyDate; decimal Buy; string SellDate; decimal Sell; string Profit; decimal Period; } [WebMethod] public Results[] GetResults() { string conn = ConfigurationManager.ConnectionStrings["LocalSqlServer"].ConnectionString; SqlConnection myconn = new SqlConnection(conn); SqlCommand mycomm = new SqlCommand(); SqlDataAdapter myda = new SqlDataAdapter(); DataSet myds = new DataSet(); mycomm.CommandType = CommandType.StoredProcedure; mycomm.Connection = myconn; mycomm.CommandText = "dbo.Results"; myconn.Open(); myda.SelectCommand = mycomm; myda.Fill(myds); myconn.Close(); myconn.Dispose(); int i = 0; Results[] dts = new Results[myds.Tables[0].Rows.Count]; foreach(DataRow arow in myds.Tables[0].Rows) { dts[i] = new Results(); dts[i].Ticker = arow["Ticker"]; dts[i].BuyDate = arow["BuyDate"]; dts[1].Buy = arow["Buy"]; dts[i].SellDate = arow["SellDate"]; dts[i].Sell = arow["Sell"]; dts[i].Profit = arow["Profit"]; dts[i].Period = arow["Period"]; i+=1; } return dts; } } The VB.NET WEBSERVICE that runs fine which I am trying to convert to C# is here. Imports System.Web Imports System.Web.Services Imports System.Web.Services.Protocols Imports System.Data Imports System.Data.SqlClient <WebService(Namespace:="http://localhost:2597/Results/ResultsVT.aspx")> _ <WebServiceBinding(ConformsTo:=WsiProfiles.BasicProfile1_1)> _ <Global.Microsoft.VisualBasic.CompilerServices.DesignerGenerated()> _ Public Class VTResults Inherits System.Web.Services.WebService Public Class Results Public Ticker As String Public BuyDate As String Public Buy As Decimal Public SellDate As String Public Sell As Decimal Public Profit As String Public Period As Decimal End Class <WebMethod()> _ Public Function GetResults() As Results() Try Dim conn As String = ConfigurationManager.ConnectionStrings("LocalSqlServer").ConnectionString Dim myconn = New SqlConnection(conn) Dim mycomm As New SqlCommand Dim myda As New SqlDataAdapter Dim myds As New DataSet mycomm.CommandType = CommandType.StoredProcedure mycomm.Connection = myconn mycomm.CommandText = "dbo.Results" myconn.Open() myda.SelectCommand = mycomm myda.Fill(myds) myconn.Close() myconn.Dispose() Dim i As Integer i = 0 Dim dts As Results() = New Results(myds.Tables(0).Rows.Count - 1) {} Dim aRow As DataRow For Each aRow In myds.Tables(0).Rows dts(i) = New Results dts(i).Ticker = aRow("Ticker") dts(i).BuyDate = aRow("BuyDate") dts(i).Buy = aRow("Buy") dts(i).SellDate = aRow("SellDate") dts(i).Sell = aRow("Sell") dts(i).Profit = aRow("Profit") dts(i).Period = aRow("Period") i += 1 Next Return dts Catch ex As DataException Throw ex End Try End Function End Class

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  • FMOD surround sound openframeworks

    - by user1449425
    Ok, I hope I don't mess this up, I have had a look for some answers but can't find anything. I am trying to make a simple sampler in openframeworks using the FMOD sound player in 3D mode. I can make a single instance work fine (recording a new file using libsndfilerecorder and then playing it back and moving it in surround. However I want to have 8 layers of looping audio that I can record and replace one layer at a time in a live show. I get a lot of problems as soon as I have more than 1 layer. The first part of my question relates to the FMOD 3D modes, it is listener relative, so I have to define the position of my listener for every sound (I would prefer to have head relative mode but I cannot make this work at all. Again this works fine when I am using a single player but with multiple players only the last listener I update actually works. The main problem I have is that when I use multiple players I get distortion, and often a mix of other currently playing sounds (even when the microphone cannot hear them) in my new recordings. Is there an incompatability with libsndfilerecorder and FMOD? Here I initialise the players for (int i=0; i<CHANNEL_COUNT; i++) { lvelocity[i].set(1, 1, 1); lup[i].set(0, 1, 0); lforward[i].set(0, 0, 1); lposition[i].set(0, 0, 0); sposition[i].set(3, 3, 2); svelocity[i].set(1, 1, 1); //player[1].initializeFmod(); //player[i].loadSound( "1.wav" ); player[i].setVolume(0.75); player[i].setMultiPlay(true); player[i].play(); setupHold[i]==false; recording[i]=false; channelHasFile[i]=false; settingOsc[i]=false; } When I am recording I unload the file and make sure the positions of the player that is not loaded are not updating. void fmodApp::recordingStart( int recordingId ){ if (recording[recordingId]==false) { setupHold[recordingId]=true; //this stops the position updating cout<<"Start recording Channel " + ofToString(recordingId+1)+" setup hold is true \n"; pt=getDateName() +".wav"; player[recordingId].stop(); player[recordingId].unloadSound(); audioRecorder.setup(pt); audioRecorder.setFormat(SF_FORMAT_WAV | SF_FORMAT_PCM_16); recording[recordingId]=true; //this starts the libSndFIleRecorder } else { cout<<"Channel" + ofToString(recordingId+1)+" is already recording \n"; } } And I stop the recording like this. void fmodApp::recordingEnd( int recordingId ){ if (recording[recordingId]=true) { recording[recordingId]=false; cout<<"Stop recording" + ofToString(recordingId+1)+" \n"; audioRecorder.finalize(); audioRecorder.close(); player[recordingId].loadSound(pt); setupHold[recordingId]=false; channelHasFile[recordingId]=true; cout<< "File recorded channel " + ofToString(recordingId+1) + " file is called " + pt + "\n"; } else { cout << "Sorry track" + ofToString(recordingId+1) + "is not recording"; } } I am careful not to interrupt the updating process but I cannot see where I am going wrong. Many Thanks

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  • Alsa doesn't work in vlc

    - by freebird
    Alsa Audio Output works fine from terminal, e.g. aplay /usr/share/sounds/alsa/Noise.wav. But I got to change from default to Alsa Audio Output in vlc. I found it in Tools Perfernces Audio Outputs. The issue is that when I change it to Alsa, I Loose all sound. When I leave the default I get an annoying Audio delay of about 200ms or 500ms. From what I have found you have to use Alsa Audio Outpu to fix that issue. Updated 6-26-2011 10:28pm To fix the Alsa Audio Output: sudo add-apt-repository ppa:ferramroberto/vlc sudo apt-get update sudo apt-get install vlc mozilla-plugin-vlc then, opened Update Manager, there were 2 updates for vlc there, I installed them and rebooted. Now alsa works fine and audio is in sync with video.

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  • Implicit constructor available for all types derived from Base excepted the current type?

    - by Vincent
    The following code sum up my problem : template<class Parameter> class Base {}; template<class Parameter1, class Parameter2, class Parameter> class Derived1 : public Base<Parameter> { }; template<class Parameter1, class Parameter2, class Parameter> class Derived2 : public Base<Parameter> { public : // Copy constructor Derived2(const Derived2& x); // An EXPLICIT constructor that does a special conversion for a Derived2 // with other template parameters template<class OtherParameter1, class OtherParameter2, class OtherParameter> explicit Derived2( const Derived2<OtherParameter1, OtherParameter2, OtherParameter>& x ); // Now the problem : I want an IMPLICIT constructor that will work for every // type derived from Base EXCEPT // Derived2<OtherParameter1, OtherParameter2, OtherParameter> template<class Type, class = typename std::enable_if</* SOMETHING */>::type> Derived2(const Type& x); }; How to restrict an implicit constructor to all classes derived from the parent class excepted the current class whatever its template parameters, considering that I already have an explicit constructor as in the example code ? EDIT : For the implicit constructor from Base, I can obviously write : template<class OtherParameter> Derived2(const Base<OtherParameter>& x); But in that case, do I have the guaranty that the compiler will not use this constructor as an implicit constructor for Derived2<OtherParameter1, OtherParameter2, OtherParameter> ? EDIT2: Here I have a test : (LWS here : http://liveworkspace.org/code/cd423fb44fb4c97bc3b843732d837abc) #include <iostream> template<typename Type> class Base {}; template<typename Type> class Other : public Base<Type> {}; template<typename Type> class Derived : public Base<Type> { public: Derived() {std::cout<<"empty"<<std::endl;} Derived(const Derived<Type>& x) {std::cout<<"copy"<<std::endl;} template<typename OtherType> explicit Derived(const Derived<OtherType>& x) {std::cout<<"explicit"<<std::endl;} template<typename OtherType> Derived(const Base<OtherType>& x) {std::cout<<"implicit"<<std::endl;} }; int main() { Other<int> other0; Other<double> other1; std::cout<<"1 = "; Derived<int> dint1; // <- empty std::cout<<"2 = "; Derived<int> dint2; // <- empty std::cout<<"3 = "; Derived<double> ddouble; // <- empty std::cout<<"4 = "; Derived<double> ddouble1(ddouble); // <- copy std::cout<<"5 = "; Derived<double> ddouble2(dint1); // <- explicit std::cout<<"6 = "; ddouble = other0; // <- implicit std::cout<<"7 = "; ddouble = other1; // <- implicit std::cout<<"8 = "; ddouble = ddouble2; // <- nothing (normal : default assignment) std::cout<<"\n9 = "; ddouble = Derived<double>(dint1); // <- explicit std::cout<<"10 = "; ddouble = dint2; // <- implicit : WHY ?!?! return 0; } The last line worry me. Is it ok with the C++ standard ? Is it a bug of g++ ?

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