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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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  • 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 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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  • 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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  • 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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  • 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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  • Blackberry Player, custom data source

    - by Alex
    Hello I must create a custom media player within the application with support for mp3 and wav files. I read in the documentation i cant seek or get the media file duration without a custom datasoruce. I checked the demo in the JDE 4.6 but i have still problems... I cant get the duration, it return much more then the expected so i`m sure i screwed up something while i modified the code to read the mp3 file locally from the filesystem. Somebody can help me what i did wrong ? (I can hear the mp3, so the player plays it correctly from start to end) I must support OSs = 4.6. Thank You Here is my modified datasource LimitedRateStreaminSource.java * Copyright © 1998-2009 Research In Motion Ltd. Note: For the sake of simplicity, this sample application may not leverage resource bundles and resource strings. However, it is STRONGLY recommended that application developers make use of the localization features available within the BlackBerry development platform to ensure a seamless application experience across a variety of languages and geographies. For more information on localizing your application, please refer to the BlackBerry Java Development Environment Development Guide associated with this release. */ package com.halcyon.tawkwidget.model; import java.io.IOException; import java.io.InputStream; import java.io.OutputStream; import javax.microedition.io.Connector; import javax.microedition.io.file.FileConnection; import javax.microedition.media.Control; import javax.microedition.media.protocol.ContentDescriptor; import javax.microedition.media.protocol.DataSource; import javax.microedition.media.protocol.SourceStream; import net.rim.device.api.io.SharedInputStream; /** * The data source used by the BufferedPlayback's media player. / public final class LimitedRateStreamingSource extends DataSource { /* The max size to be read from the stream at one time. */ private static final int READ_CHUNK = 512; // bytes /** A reference to the field which displays the load status. */ //private TextField _loadStatusField; /** A reference to the field which displays the player status. */ //private TextField _playStatusField; /** * The minimum number of bytes that must be buffered before the media file * will begin playing. */ private int _startBuffer = 200000; /** The maximum size (in bytes) of a single read. */ private int _readLimit = 32000; /** * The minimum forward byte buffer which must be maintained in order for * the video to keep playing. If the forward buffer falls below this * number, the playback will pause until the buffer increases. */ private int _pauseBytes = 64000; /** * The minimum forward byte buffer required to resume * playback after a pause. */ private int _resumeBytes = 128000; /** The stream connection over which media content is passed. */ //private ContentConnection _contentConnection; private FileConnection _fileConnection; /** An input stream shared between several readers. */ private SharedInputStream _readAhead; /** A stream to the buffered resource. */ private LimitedRateSourceStream _feedToPlayer; /** The MIME type of the remote media file. */ private String _forcedContentType; /** A counter for the total number of buffered bytes */ private volatile int _totalRead; /** A flag used to tell the connection thread to stop */ private volatile boolean _stop; /** * A flag used to indicate that the initial buffering is complete. In * other words, that the current buffer is larger than the defined start * buffer size. */ private volatile boolean _bufferingComplete; /** A flag used to indicate that the remote file download is complete. */ private volatile boolean _downloadComplete; /** The thread which retrieves the remote media file. */ private ConnectionThread _loaderThread; /** The local save file into which the remote file is written. */ private FileConnection _saveFile; /** A stream for the local save file. */ private OutputStream _saveStream; /** * Constructor. * @param locator The locator that describes the DataSource. */ public LimitedRateStreamingSource(String locator) { super(locator); } /** * Open a connection to the locator. * @throws IOException */ public void connect() throws IOException { //Open the connection to the remote file. _fileConnection = (FileConnection)Connector.open(getLocator(), Connector.READ); //Cache a reference to the locator. String locator = getLocator(); //Report status. System.out.println("Loading: " + locator); //System.out.println("Size: " + _contentConnection.getLength()); System.out.println("Size: " + _fileConnection.totalSize()); //The name of the remote file begins after the last forward slash. int filenameStart = locator.lastIndexOf('/'); //The file name ends at the first instance of a semicolon. int paramStart = locator.indexOf(';'); //If there is no semicolon, the file name ends at the end of the line. if (paramStart < 0) { paramStart = locator.length(); } //Extract the file name. String filename = locator.substring(filenameStart, paramStart); System.out.println("Filename: " + filename); //Open a local save file with the same name as the remote file. _saveFile = (FileConnection) Connector.open("file:///SDCard/blackberry/music" + filename, Connector.READ_WRITE); //If the file doesn't already exist, create it. if (!_saveFile.exists()) { _saveFile.create(); } System.out.println("---------- 1"); //Open the file for writing. _saveFile.setReadable(true); //Open a shared input stream to the local save file to //allow many simultaneous readers. SharedInputStream fileStream = SharedInputStream.getSharedInputStream(_saveFile.openInputStream()); //Begin reading at the beginning of the file. fileStream.setCurrentPosition(0); System.out.println("---------- 2"); //If the local file is smaller than the remote file... if (_saveFile.fileSize() < _fileConnection.totalSize()) { System.out.println("---------- 3"); //Did not get the entire file, set the system to try again. _saveFile.setWritable(true); System.out.println("---------- 4"); //A non-null save stream is used as a flag later to indicate that //the file download was incomplete. _saveStream = _saveFile.openOutputStream(); System.out.println("---------- 5"); //Use a new shared input stream for buffered reading. _readAhead = SharedInputStream.getSharedInputStream(_fileConnection.openInputStream()); System.out.println("---------- 6"); } else { //The download is complete. System.out.println("---------- 7"); _downloadComplete = true; //We can use the initial input stream to read the buffered media. _readAhead = fileStream; System.out.println("---------- 8"); //We can close the remote connection. _fileConnection.close(); System.out.println("---------- 9"); } if (_forcedContentType != null) { //Use the user-defined content type if it is set. System.out.println("---------- 10"); _feedToPlayer = new LimitedRateSourceStream(_readAhead, _forcedContentType); System.out.println("---------- 11"); } else { System.out.println("---------- 12"); //Otherwise, use the MIME types of the remote file. // _feedToPlayer = new LimitedRateSourceStream(_readAhead, _fileConnection)); } System.out.println("---------- 13"); } /** * Destroy and close all existing connections. */ public void disconnect() { try { if (_saveStream != null) { //Destroy the stream to the local save file. _saveStream.close(); _saveStream = null; } //Close the local save file. _saveFile.close(); if (_readAhead != null) { //Close the reader stream. _readAhead.close(); _readAhead = null; } //Close the remote file connection. _fileConnection.close(); //Close the stream to the player. _feedToPlayer.close(); } catch (Exception e) { System.err.println(e.getMessage()); } } /** * Returns the content type of the remote file. * @return The content type of the remote file. */ public String getContentType() { return _feedToPlayer.getContentDescriptor().getContentType(); } /** * Returns a stream to the buffered resource. * @return A stream to the buffered resource. */ public SourceStream[] getStreams() { return new SourceStream[] { _feedToPlayer }; } /** * Starts the connection thread used to download the remote file. */ public void start() throws IOException { //If the save stream is null, we have already completely downloaded //the file. if (_saveStream != null) { //Open the connection thread to finish downloading the file. _loaderThread = new ConnectionThread(); _loaderThread.start(); } } /** * Stop the connection thread. */ public void stop() throws IOException { //Set the boolean flag to stop the thread. _stop = true; } /** * @see javax.microedition.media.Controllable#getControl(String) */ public Control getControl(String controlType) { // No implemented Controls. return null; } /** * @see javax.microedition.media.Controllable#getControls() */ public Control[] getControls() { // No implemented Controls. return null; } /** * Force the lower level stream to a given content type. Must be called * before the connect function in order to work. * @param contentType The content type to use. */ public void setContentType(String contentType) { _forcedContentType = contentType; } /** * A stream to the buffered media resource. */ private final class LimitedRateSourceStream implements SourceStream { /** A stream to the local copy of the remote resource. */ private SharedInputStream _baseSharedStream; /** Describes the content type of the media file. */ private ContentDescriptor _contentDescriptor; /** * Constructor. Creates a LimitedRateSourceStream from * the given InputStream. * @param inputStream The input stream used to create a new reader. * @param contentType The content type of the remote file. */ LimitedRateSourceStream(InputStream inputStream, String contentType) { System.out.println("[LimitedRateSoruceStream]---------- 1"); _baseSharedStream = SharedInputStream.getSharedInputStream(inputStream); System.out.println("[LimitedRateSoruceStream]---------- 2"); _contentDescriptor = new ContentDescriptor(contentType); System.out.println("[LimitedRateSoruceStream]---------- 3"); } /** * Returns the content descriptor for this stream. * @return The content descriptor for this stream. */ public ContentDescriptor getContentDescriptor() { return _contentDescriptor; } /** * Returns the length provided by the connection. * @return long The length provided by the connection. */ public long getContentLength() { return _fileConnection.totalSize(); } /** * Returns the seek type of the stream. */ public int getSeekType() { return RANDOM_ACCESSIBLE; //return SEEKABLE_TO_START; } /** * Returns the maximum size (in bytes) of a single read. */ public int getTransferSize() { return _readLimit; } /** * Writes bytes from the buffer into a byte array for playback. * @param bytes The buffer into which the data is read. * @param off The start offset in array b at which the data is written. * @param len The maximum number of bytes to read. * @return the total number of bytes read into the buffer, or -1 if * there is no more data because the end of the stream has been reached. * @throws IOException */ public int read(byte[] bytes, int off, int len) throws IOException { System.out.println("[LimitedRateSoruceStream]---------- 5"); System.out.println("Read Request for: " + len + " bytes"); //Limit bytes read to our readLimit. int readLength = len; System.out.println("[LimitedRateSoruceStream]---------- 6"); if (readLength > getReadLimit()) { readLength = getReadLimit(); } //The number of available byes in the buffer. int available; //A boolean flag indicating that the thread should pause //until the buffer has increased sufficiently. boolean paused = false; System.out.println("[LimitedRateSoruceStream]---------- 7"); for (;;) { available = _baseSharedStream.available(); System.out.println("[LimitedRateSoruceStream]---------- 8"); if (_downloadComplete) { //Ignore all restrictions if downloading is complete. System.out.println("Complete, Reading: " + len + " - Available: " + available); return _baseSharedStream.read(bytes, off, len); } else if(_bufferingComplete) { if (paused && available > getResumeBytes()) { //If the video is paused due to buffering, but the //number of available byes is sufficiently high, //resume playback of the media. System.out.println("Resuming - Available: " + available); paused = false; return _baseSharedStream.read(bytes, off, readLength); } else if(!paused && (available > getPauseBytes() || available > readLength)) { //We have enough information for this media playback. if (available < getPauseBytes()) { //If the buffer is now insufficient, set the //pause flag. paused = true; } System.out.println("Reading: " + readLength + " - Available: " + available); return _baseSharedStream.read(bytes, off, readLength); } else if(!paused) { //Set pause until loaded enough to resume. paused = true; } } else { //We are not ready to start yet, try sleeping to allow the //buffer to increase. try { Thread.sleep(500); } catch (Exception e) { System.err.println(e.getMessage()); } } } } /** * @see javax.microedition.media.protocol.SourceStream#seek(long) */ public long seek(long where) throws IOException { _baseSharedStream.setCurrentPosition((int) where); return _baseSharedStream.getCurrentPosition(); } /** * @see javax.microedition.media.protocol.SourceStream#tell() */ public long tell() { return _baseSharedStream.getCurrentPosition(); } /** * Close the stream. * @throws IOException */ void close() throws IOException { _baseSharedStream.close(); } /** * @see javax.microedition.media.Controllable#getControl(String) */ public Control getControl(String controlType) { // No implemented controls. return null; } /** * @see javax.microedition.media.Controllable#getControls() */ public Control[] getControls() { // No implemented controls. return null; } } /** * A thread which downloads the remote file and writes it to the local file. */ private final class ConnectionThread extends Thread { /** * Download the remote media file, then write it to the local * file. * @see java.lang.Thread#run() */ public void run() { try { byte[] data = new byte[READ_CHUNK]; int len = 0; //Until we reach the end of the file. while (-1 != (len = _readAhead.read(data))) { _totalRead += len; if (!_bufferingComplete && _totalRead > getStartBuffer()) { //We have enough of a buffer to begin playback. _bufferingComplete = true; System.out.println("Initial Buffering Complete"); } if (_stop) { //Stop reading. return; } } System.out.println("Downloading Complete"); System.out.println("Total Read: " + _totalRead); //If the downloaded data is not the same size //as the remote file, something is wrong. if (_totalRead != _fileConnection.totalSize()) { System.err.println("* Unable to Download entire file *"); } _downloadComplete = true; _readAhead.setCurrentPosition(0); //Write downloaded data to the local file. while (-1 != (len = _readAhead.read(data))) { _saveStream.write(data); } } catch (Exception e) { System.err.println(e.toString()); } } } /** * Gets the minimum forward byte buffer which must be maintained in * order for the video to keep playing. * @return The pause byte buffer. */ int getPauseBytes() { return _pauseBytes; } /** * Sets the minimum forward buffer which must be maintained in order * for the video to keep playing. * @param pauseBytes The new pause byte buffer. */ void setPauseBytes(int pauseBytes) { _pauseBytes = pauseBytes; } /** * Gets the maximum size (in bytes) of a single read. * @return The maximum size (in bytes) of a single read. */ int getReadLimit() { return _readLimit; } /** * Sets the maximum size (in bytes) of a single read. * @param readLimit The new maximum size (in bytes) of a single read. */ void setReadLimit(int readLimit) { _readLimit = readLimit; } /** * Gets the minimum forward byte buffer required to resume * playback after a pause. * @return The resume byte buffer. */ int getResumeBytes() { return _resumeBytes; } /** * Sets the minimum forward byte buffer required to resume * playback after a pause. * @param resumeBytes The new resume byte buffer. */ void setResumeBytes(int resumeBytes) { _resumeBytes = resumeBytes; } /** * Gets the minimum number of bytes that must be buffered before the * media file will begin playing. * @return The start byte buffer. */ int getStartBuffer() { return _startBuffer; } /** * Sets the minimum number of bytes that must be buffered before the * media file will begin playing. * @param startBuffer The new start byte buffer. */ void setStartBuffer(int startBuffer) { _startBuffer = startBuffer; } } And in this way i use it: LimitedRateStreamingSource source = new LimitedRateStreamingSource("file:///SDCard/music3.mp3"); source.setContentType("audio/mpeg"); mediaPlayer = javax.microedition.media.Manager.createPlayer(source); mediaPlayer.addPlayerListener(this); mediaPlayer.realize(); mediaPlayer.prefetch(); After start i use mediaPlayer.getDuration it returns lets say around 24:22 (the inbuild media player in the blackberry say the file length is 4:05) I tried to get the duration in the listener and there unfortunatly returned around 64 minutes, so im sure something is not good inside the datasoruce....

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  • Why isn't this driver install working (sudo code)?

    - by Nick
    I have a soundcard that I'd like to use and I've been trying to install it and being a new Ubuntu user, I get about half way through this in the Terminal and it stops cooperating with me... See the link (soundcard hyperlink) but basically what I have here: I do the following and it works: sudo apt-get install subversion svn co https://line6linux.svn.sourceforge.net/svnroot/line6linux Change to the directory cd line6linux/driver/trunk Time to build from the source but first make sure you have the latest build and headers sudo apt-get install build-essential sudo apt-get install linux-headers Then after this point it says must specify file to install. Not sure how to do this or what it means. Then, running make gives the following output: ./set_revision.sh ./set_revision.sh: 9: test: https://line6linux.svn.sourceforge.net/svnroot/line6linux/driver/trunk: unexpected operator make -C /lib/modules/3.2.0-29-generic-pae/build CONFIG_LINE6_USB=m SUBDIRS=/home/nick/line6linux/driver/trunk modules make[1]: Entering directory /usr/src/linux-headers-3.2.0-29-generic-pae' CC [M] /home/nick/line6linux/driver/trunk/audio.o /home/nick/line6linux/driver/trunk/audio.c: In function ‘line6_init_audio’: /home/nick/line6linux/driver/trunk/audio.c:30:57: error: ‘THIS_MODULE’ undeclared (first use in this function) /home/nick/line6linux/driver/trunk/audio.c:30:57: note: each undeclared identifier is reported only once for each function it appears in make[2]: * [/home/nick/line6linux/driver/trunk/audio.o] Error 1 make[1]: * [module/home/nick/line6linux/driver/trunk] Error 2 make[1]: Leaving directory/usr/src/linux-headers-3.2.0-29-generic-pae' make: * [default] Error 2 This is in Ubuntu 12.04.1 LTS Another thing, semi related. Cut, copy, paste? Seems like it's different from program to program. I was in the terminal and hit Ctrl-C and then Ctrl-Shift-V in Firefox and it won't paste. But in terminal it will paste. I'm confused. Here is what it's giving me after I hit "Make": nick@NickUbuntu:~/line6linux/driver/trunk$ make ./set_revision.sh ./set_revision.sh: 9: test: https://line6linux.svn.sourceforge.net/svnroot/line6linux/driver/trunk: unexpected operator make -C /lib/modules/3.2.0-29-generic-pae/build CONFIG_LINE6_USB=m SUBDIRS=/home/nick/line6linux/driver/trunk modules make[1]: Entering directory /usr/src/linux-headers-3.2.0-29-generic-pae' CC [M] /home/nick/line6linux/driver/trunk/audio.o /home/nick/line6linux/driver/trunk/audio.c: In function ‘line6_init_audio’: /home/nick/line6linux/driver/trunk/audio.c:30:57: error: ‘THIS_MODULE’ undeclared (first use in this function) /home/nick/line6linux/driver/trunk/audio.c:30:57: note: each undeclared identifier is reported only once for each function it appears in make[2]: *** [/home/nick/line6linux/driver/trunk/audio.o] Error 1 make[1]: *** [_module_/home/nick/line6linux/driver/trunk] Error 2 make[1]: Leaving directory/usr/src/linux-headers-3.2.0-29-generic-pae' make: * [default] Error 2 Looks like these folks also had similar problems: http://ubuntuforums.org/showthread.php?t=1163608&page=3

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  • alsa doesn't won't in vlc

    - by freebird
    Alsa Audio Output works fine from terminal aplay /usr/share/sounds/alsa/Noise.wav . But i got to change from default to Alsa Audio Output in vlc . Found in Tools Perfernces Audio Outputs The issue lie when i change it to Alsa i Loose all sound. When i leave it defualt i get a annoying Audio delay of like 200ms or 500ms. from what i have found you have to use Alsa Audio Outpu to fix that issue.

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  • Automatically change Sound Input Output device

    - by Senthil Kumaran
    I have to plugin my USB Audio adapter ( 4300054 Gigawire USB Audio Adapter) for audio input because has a combo-input-output port for voice. After I do this, I have go open Sound Settings and manually select the USB Audio adapter for Input and Output, if I do not, the system default remains selected. Is there anyway, I can make Ubuntu to automatically select the USB Audio Adapter as the default as soon as I plug-in?

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  • Sound not working on an Intel 5 Series/3400

    - by phoenix7
    lspci gives me these two devices: $ lspci | grep Audio 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 05) 02:00.1 Audio device: ATI Technologies Inc RV710/730 There are two devices listed in System Settings|Sound|Output: RV710/730 Digital Stereo (HDMI) Internal Audio Analog Stereo And finally, the are not muted! Also, when I run an application that accesses the sound card, I can see it in the Applications tab. Any ideas?

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  • hdmi audio works only with aplay -D alsa test wavs; open source radeon drivers; kernel 3.5 vgaswitcheroo

    - by user108754
    I've trolled the internets to make hdmi work on my system Ubuntu 12.04 software center kernel 3.5 uname: Linux ubuntu 3.5.0-18-generic #29~precise1-Ubuntu SMP...x86_64 x86_64 x86_64 GNU/Linux open source radeon drivers vgaswitcheroo (hybrid intel/radeon gpu): I boot with intel, not radeon, running. (and recall that with kernel 3.5, vgaswitcheroo now gives info on a third item, "DIS-Audio"; it indicates pwr on my system) ( /etc/rc.local: chown user:user /sys/kernel/debug/ # change "username" with your user name echo OFF /sys/kernel/debug/vgaswitcheroo/switch ) grub indeed now has "radeon.audio=1" for testing audio, I did aplay -l which gave me the card and device, which made me try aplay -D plughw:1,3 /usr/share/sounds/alsa/Front_Center.wav and lo! I get crystal clear sound on my hdtv. If I play an mp3 file as the argument to that command, I get noise as, I guess, aplay interprets the mp3 code as a wav. If I play a .wav that is not in the /usr/share/sounds/alsa/ directory, I get nothing. Internet flash video in browser plays no sound over hdmi. Both system sounds control and pavucontrol have hdmi cedar selected. Alas, I can not get sound for any gui test (left, right). Why would only aplay, and only when directed with "-D plughw", yield sound over hdmi? I've also tried only using one sound program at a time, if it was a limitation of alsa, so I tried aplay with web browser and even the sound control gui closed. I tried each of the last two, running alone. No improvement. alsamixer only shows hda intel and I think it's only the intel audio, not the hdmi.

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  • How can I find the song position of a song being played with XACT?

    - by DJ SymBiotiX
    So I'm making a game in XNA and I need to use XACT for my songs (rather than media player). I need to use XACT because each song will have multiple layers that combine when played at the same time (bass, lead, drums) etc. I cant use the media player because the media player can only play one song at a time. Anyways, so lets say I have a song playing with XACT in my project with the following code public SongController() { audioEngine = new AudioEngine(@"Content\Song1\Song1.xgs"); waveBank = new WaveBank(audioEngine, @"Content\Song1\Layers.xwb"); soundBank = new SoundBank(audioEngine, @"Content\Song1\SongLayers.xsb"); songTime = new PlayTime(); Vox = soundBank.GetCue("Vox"); BG = soundBank.GetCue("BG"); Bass = soundBank.GetCue("Bass"); Lead = soundBank.GetCue("Lead"); Other = soundBank.GetCue("Other"); Vox.SetVariable("CueVolume", 100.0f); BG.SetVariable("CueVolume", 100.0f); Bass.SetVariable("CueVolume", 100.0f); Lead.SetVariable("CueVolume", 100.0f); Other.SetVariable("CueVolume", 100.0f); _bassVol = 100.0f; _voxVol = 100.0f; _leadVol = 100.0f; _otherVol = 100.0f; Vox.Play(); BG.Play(); Bass.Play(); Lead.Play(); Other.Play(); } So when I look at the variables in Vox, or BG (they are Cue's btw) I cant seem to find any play position in them. So I guess the question is: Is there a variable I can query to find that data, or do I need to make my own class that starts counting up from the time I start the song? Thanks

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  • kAudioSessionProperty_CurrentHardwareSampleRate input/output

    - by iter
    kAudioSessionProperty_CurrentHardwareSampleRate seems to describe the input sampling rate. I wonder if there is a way to determine the available output sampling rate on an iPhone / iPad (iPhone supports 44.1K; iPad, 48K). http://developer.apple.com/iphone/library/documentation/AudioToolbox/Reference/AudioSessionServicesReference/Reference/reference.html#//apple_ref/doc/c_ref/kAudioSessionProperty_CurrentHardwareSampleRate

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