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  • How can I stop Ubuntu from playing audio from 2 interfaces at the same time?

    - by Solignis
    Hi there, I just loaded Ubuntu 10.10 Maverick on my home machine. The machine is running a Core2Duo E6750 on an MSI motherboard with an Nvidia GTX260-OC Graphics card. The problem I am having as stated in the title is for some reason Ubuntu is playing audio through my headphone coming out from the computer and it is also playing the audio at the exact same time through the HDMI connection coming out of the graphics card, it has a plug to allow this. What is going on, I have never seen this before. Most importantly of all can it be fixed so that I can sepertate the 2 interfaces, the one is a standard PC audio IO and the HDMI one is connected through the mobo's internal SPDIF. More information can be provided if required.

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  • Default audio device gives an error on WINDOWS 7 (x64) when triing to run VLC from CMD (VideoLAN, VL

    - by Ole Jak
    I use WINDOWS 7 (x64) (Russian) I want to stream life audio from my default audio capture device (microphone) When I set up VLM settings using visual enviroment instruments - VLM settings it all works fine. But when I export created settings/configuration *.vlm file and try to inport it into VLM it gives me nothing I opened that .vlm there is some text... so now I try to run VLC with default settings like this: vlc -i dshow:// --dshow-adev= :sout=#transcode{acodec=mp3,ab=128,channels=2,samplerate=44100}:std{access=http,mux=raw,dst=127.0.0.1:8084} but it dies giving me errors...=( So what shall I do to do live MP3 streaming from my default audio input device using VLC in non UI mode?

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  • Can I reprogram a microphone input to be used as an audio output? (on XP)

    - by qftme
    I have a five year old Sony Vaio laptop (vgn-fw31m) that has had impact damage to the audio-output mini-jack for about the last year or so. In a recent discussion with my brother, we wondered whether it would be possible to write a program that would enable windows to use the microphone mini-jack input as the audio-output? As I currently use this laptop for work I am not keen to risk pulling it apart in order to replace the components comprising the audio-out. I therefore 'hope' that a programming solution exists. I would really appreciate any advice on this and eagerly await your response. Kind regards, qftme :)

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  • xVelocity engines compared: VertiPaq vs ColumnStore #ssas #vertipaq #xvelocity #sql #tabular

    - by Marco Russo (SQLBI)
    During the last months I and Alberto worked in several projects using Analysis Services Tabular and we had to face real world issues, such as complex queries, large data volume, frequent data updates and so on. Sometime we faced the challenge of comparing Tabular performance with SQL Server. It seemed a non-sense, because even if the same core xVelocity technology is implemented in both products (SQL Server 2012 uses ColumnStore indexes, whereas Analysis Services 2012 uses VertiPaq), we initially assumed that the better optimization for the in-memory engine used by Analysis Services would have been always better than SQL Server. However, we discovered several important things: Processing time might be different and having data on SQL Server could make ColumnStore way faster for processing. Partitioning in SQL Server might be much more effective for query performance than Analysis Services. A single query can scale easily on more processor on SQL Server, whereas in Analysis Services the formula engine is single-threaded and could be a bottleneck for certain queries. In case of a large workload with many concurrent users, storage engine cache in Analysis Services could be a big advantage over SQL Server, especially for scalability As you can see, these considerations are not always obvious and you might be tempted to make other assumptions based on these information. Well, don’t do that. Before anything else, read the whitepaper VertiPaq vs ColumnStore Comparison written by Alberto Ferrari. Then, measure your workload. Finally, make some conclusion. But don’t make too many assumptions. You might be wrong, as we did at the beginning of this journey.

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  • Order by text and then by number

    - by Chaim Chaikin
    I have data like: Audio 1 File 10 Audio 2 Audio 3 File 11 Audio 13 Audio 22 File 20 Test 22 Audio 10 File 1 File 2 I need it order first by the text (i.e. Audio, File, Test) and then by number (1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22 etc.) The problem is that sorting it returns something like this: Audio 1 Audio 10 Audio 13 Audio 2 Audio 22 Audio 3 File 1 File 10 File 11 File 2 File 20 Test 22 While the result I want is: Audio 1 Audio 2 Audio 3 Audio 10 Audio 13 Audio 22 File 1 File 2 File 10 File 11 File 20 Test 22 If they were just numbers (i.e. without the audio, file, test) then I could just sort numerically. However, how can I sort here first by text and then by number.

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  • KDE: How can I select audio output device for mplayer?

    - by grimripper
    I recently installed Kubuntu 13.10 64-bit, and I'm having a problem with selecting audio output device. In Phonon, when I select audio device preference order and press Apply, Amarok and Dragon will immediately switch to the preferred device. VLC and SMplayer are not affected. VLC has its own setting for selecting the output device, but SMplayer remains a problem. It always plays audio on internal audio, and I can't change output to HDMI. How can I select HDMI for SMplayer's audio output device? I don't know if it matters, but when I select HDMI audio in Phonon and click Test, the test sound plays on the internal audio output as well. In the hardware settings tab, the front left and front right test buttons play audio on HDMI. Also, volume up/down buttons affect HDMI volume when SMplayer is focused. This would make sense if I could get SMplayer to play audio over HDMI, but it would be better if the volume keys affected SMplayer's own volume, or the "mplayer2: audio stream" which appears in volume control while mplayer is playing. EDIT: I've recompiled mplayer with alsa support, and can now select the audio output device from SMplayer's settings. Didn't affect the issue with Phonon of course, but it's a suitable workaround.

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  • AVAudioRecorder - Continue recording to file after user stops recording by leaving the application a

    - by Tegeril
    Can this be done? And if not, how far down towards Core Audio do I need to go (what method of recording should I be using instead)? I've noticed the behavior of AVAudioRecorder is to overwrite a file if it finds one at the path provided when you request that it record again, so I know that's not going to work. I'm also curious about file format restriction with this idea. Can you effectively resume an AAC or IMA4 encoding (the length of the files I want to record make WAV and probably even Apple Lossless prohibitive)? Thanks.

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  • How to get audio driver for compaq c700 ?

    - by Leena
    Hi, Initially i have audio driver and its works fine.But some times speaker was clear.So one of my friend installed some audio driver,after that totally disabled the volume. For that reason, i also tried to get audio driver and installed many times.Now i don't know many drivers .inf in my laptop.from device manager i have deleted the audio driver's,below i have attached the screen shot yours kind reference. Please help me to get audio drivers.First, i need to remove the unwanted drivers .inf files from laptop then i have to install the new audio driver. Experts,please suggest me to get audio driver without reinstall the OS. Details: Compaq c700 (i don't know model number) windows xp sp2 p/n : KT188PA#ACJ I appreciate your help.

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  • How to get audio driver for compaq c700 ?

    - by Leena
    Hi, Initially i have audio driver and its works fine.But some times speaker was clear.So one of my friend installed some audio driver,after that totally disabled the volume. For that reason, i also tried to get audio driver and installed many times.Now i don't know many drivers .inf in my laptop.from device manager i have deleted the audio driver's,below i have attached the screen shot yours kind reference. Please help me to get audio drivers.First, i need to remove the unwanted drivers .inf files from laptop then i have to install the new audio driver. Experts,please suggest me to get audio driver without reinstall the OS. Details: Compaq c700 (i don't know model number) windows xp sp2 p/n : KT188PA#ACJ I appreciate your help.

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  • How to get audio driver for compaq c700 ?

    - by Leena
    Hi, Initially i have audio driver and its works fine.But some times speaker was clear.So one of my friend installed some audio driver,after that totally disabled the volume. For that reason, i also tried to get audio driver and installed many times.Now i don't know many drivers .inf in my laptop.from device manager i have deleted the audio driver's,below i have attached the screen shot yours kind reference. Please help me to get audio drivers.First, i need to remove the unwanted drivers .inf files from laptop then i have to install the new audio driver. Experts,please suggest me to get audio driver without reinstall the OS. Details: Compaq c700 (i don't know model number) windows xp sp2 p/n : KT188PA#ACJ I appreciate your help.

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  • Streaming Audio over UDP to Android

    - by Mr. Pig
    Is it possible to have Android (perhaps via MediaPlayer or a different existing class) accept media streams over UDP? I've successfully had MediaPlayer connect to an HTTP stream (as well as static files hosted on an HTTP server) but I'm wondering how one would go about accepting a stream from a UDP source. I've seen this and suppose a solution similar to that (where I download the stream via an independent UDP socket and then move the data to a MemoryBuffer that I then pass to MediaPlayer) is an option but I'm curious if a method already exists in the SDK, and if it does not, what other options do I have? Thanks

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  • Audio recording and playback in Silverlight

    - by Ramesh
    I have a Silverlight 4 application that records user's voice through the mic. Now, as soon as the recording is completed, I need to play the recorded voice back to the user before posting it to the server. Is it at all possible to play it back to the user without getting into format conversions etc? Any ideas are welcome. Thanks!

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  • Android - Getting audio to play through earpiece

    - by Donal Rafferty
    I currently have code that reads a recording in from the devices mic using the AudioRecord class and then playing it back out using the AudioTrack class. My problem is that when I play it out it plays vis the speaker phone. I want it to play out via the ear piece on the device. Here is my code: public class LoopProg extends Activity { boolean isRecording; //currently not used AudioManager am; int count = 0; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.main); am = (AudioManager) getSystemService(Context.AUDIO_SERVICE); am.setMicrophoneMute(true); while(count <= 1000000){ Record record = new Record(); record.run(); count ++; Log.d("COUNT", "Count is : " + count); } } public class Record extends Thread { static final int bufferSize = 200000; final short[] buffer = new short[bufferSize]; short[] readBuffer = new short[bufferSize]; public void run() { isRecording = true; android.os.Process.setThreadPriority (android.os.Process.THREAD_PRIORITY_URGENT_AUDIO); int buffersize = AudioRecord.getMinBufferSize(11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT); AudioRecord arec = new AudioRecord(MediaRecorder.AudioSource.MIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize); AudioTrack atrack = new AudioTrack(AudioManager.STREAM_MUSIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize, AudioTrack.MODE_STREAM); am.setRouting(AudioManager.MODE_NORMAL,1, AudioManager.STREAM_MUSIC); int ok = am.getRouting(AudioManager.ROUTE_EARPIECE); Log.d("ROUTING", "getRouting = " + ok); setVolumeControlStream(AudioManager.STREAM_VOICE_CALL); //am.setSpeakerphoneOn(true); Log.d("SPEAKERPHONE", "Is speakerphone on? : " + am.isSpeakerphoneOn()); am.setSpeakerphoneOn(false); Log.d("SPEAKERPHONE", "Is speakerphone on? : " + am.isSpeakerphoneOn()); atrack.setPlaybackRate(11025); byte[] buffer = new byte[buffersize]; arec.startRecording(); atrack.play(); while(isRecording) { arec.read(buffer, 0, buffersize); atrack.write(buffer, 0, buffer.length); } arec.stop(); atrack.stop(); isRecording = false; } } } As you can see if the code I have tried using the AudioManager class and its methods including the deprecated setRouting method and nothing works, the setSpeatPoneOn method seems to have no effect at all, neither does the routing method. Has anyone got any ideas on how to get it to play via the earpiece instead of the spaker phone?

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  • Core-audio - constructing an AudioBufferList struct (Q about c struct definition)

    - by mustISignUp
    The definition of AudioBufferList looks weird to me… i guess my C is not so good struct AudioBufferList { UInt32 mNumberBuffers; AudioBuffer mBuffers[kVariableLengthArray]; }; typedef struct AudioBufferList AudioBufferList; Why AudioBuffer mBuffers[kVariableLengthArray]; and not AudioBuffer *mBuffers; ? kVariableLengthArray appears to be == 1. Eh? I think i have it working but would appreciate it if anyone could set me straight.

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  • Finding out estimated duration of a stream using Core Audio

    - by Reflog
    I am streaming a MP3 over network using custom feeding code, not AVAudioPlayer (which only works with URLs) using APIs like AudioFileStreamOpen and etc. Is there any way to estimate a length of the stream? I know that I can get a 'elapsed' property using: if(AudioQueueGetCurrentTime(queue.audioQueue, NULL, &t, &b) < 0) return 0; return t.mSampleTime / dataFormat.mSampleRate; But what about total duration to create a progress bar? Is that possible?

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  • Core-audio - constructing an AudioBufferList

    - by mustISignUp
    The definition of AudioBufferList looks weird to me… i guess my C is not so good struct AudioBufferList { UInt32 mNumberBuffers; AudioBuffer mBuffers[kVariableLengthArray]; }; typedef struct AudioBufferList AudioBufferList; Why AudioBuffer mBuffers[kVariableLengthArray]; and not AudioBuffer *mBuffers; ? kVariableLengthArray appears to be == 1. Eh? I think i have it working but would appreciate it if anyone could set me straight.

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  • processing an audio wav file with C

    - by sa125
    Hi - I'm working on processing the amplitude of a wav file and scaling it by some decimal factor. I'm trying to wrap my head around how to read and re-write the file in a memory-efficient way while also trying to tackle the nuances of the language (I'm new to C). The file can be in either an 8- or 16-bit format. The way I thought of doing this is by first reading the header data into some pre-defined struct, and then processing the actual data in a loop where I'll read a chunk of data into a buffer, do whatever is needed to it, and then write it to the output. #include <stdio.h> #include <stdlib.h> typedef struct header { char chunk_id[4]; int chunk_size; char format[4]; char subchunk1_id[4]; int subchunk1_size; short int audio_format; short int num_channels; int sample_rate; int byte_rate; short int block_align; short int bits_per_sample; short int extra_param_size; char subchunk2_id[4]; int subchunk2_size; } header; typedef struct header* header_p; void scale_wav_file(char * input, float factor, int is_8bit) { FILE * infile = fopen(input, "rb"); FILE * outfile = fopen("outfile.wav", "wb"); int BUFSIZE = 4000, i, MAX_8BIT_AMP = 255, MAX_16BIT_AMP = 32678; // used for processing 8-bit file unsigned char inbuff8[BUFSIZE], outbuff8[BUFSIZE]; // used for processing 16-bit file short int inbuff16[BUFSIZE], outbuff16[BUFSIZE]; // header_p points to a header struct that contains the file's metadata fields header_p meta = (header_p)malloc(sizeof(header)); if (infile) { // read and write header data fread(meta, 1, sizeof(header), infile); fwrite(meta, 1, sizeof(meta), outfile); while (!feof(infile)) { if (is_8bit) { fread(inbuff8, 1, BUFSIZE, infile); } else { fread(inbuff16, 1, BUFSIZE, infile); } // scale amplitude for 8/16 bits for (i=0; i < BUFSIZE; ++i) { if (is_8bit) { outbuff8[i] = factor * inbuff8[i]; if ((int)outbuff8[i] > MAX_8BIT_AMP) { outbuff8[i] = MAX_8BIT_AMP; } } else { outbuff16[i] = factor * inbuff16[i]; if ((int)outbuff16[i] > MAX_16BIT_AMP) { outbuff16[i] = MAX_16BIT_AMP; } else if ((int)outbuff16[i] < -MAX_16BIT_AMP) { outbuff16[i] = -MAX_16BIT_AMP; } } } // write to output file for 8/16 bit if (is_8bit) { fwrite(outbuff8, 1, BUFSIZE, outfile); } else { fwrite(outbuff16, 1, BUFSIZE, outfile); } } } // cleanup if (infile) { fclose(infile); } if (outfile) { fclose(outfile); } if (meta) { free(meta); } } int main (int argc, char const *argv[]) { char infile[] = "file.wav"; float factor = 0.5; scale_wav_file(infile, factor, 0); return 0; } I'm getting differing file sizes at the end (by 1k or so, for a 40Mb file), and I suspect this is due to the fact that I'm writing an entire buffer to the output, even though the file may have terminated before filling the entire buffer size. Also, the output file is messed up - won't play or open - so I'm probably doing the whole thing wrong. Any tips on where I'm messing up will be great. Thanks!

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  • audio processing using java

    - by Sukhhhh
    We have a requirement where we need to convert from .wav file to .mp3 and we are currently using "Tritonus" library to do that . The concern with that library is that requires "installation" of some "dll" files to the class path. I am wondering are there any API's those allow better processing without local installation. And other question is ,having mp3 format files will make it easier to join the files into a single file than having .wav files ?

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  • Play multiple audio files using AVAudioPlayer

    - by inScript09
    Hi all, I am planning on releasing 10 of my song recordings for free but bundled in an iphone app. They are not available on web or itunes or anywhere as of now. I am new to iphone sdk (latest) as you can imagine, so I have been going through the developer documentation, various forums and stackoverflow to learn. Apple's avTouch sample application was a great start. But I want my app to play all the 10 tracks one by one. All the songs are added to resources folder and are named as track1, track2...track10. In the avTouch app code I can see the following 2 parts which is where I think I need to make changes to achieve what I am looking for. But I am lost. // Load the array with the sample file NSURL *fileURL = [[NSURL alloc] initFileURLWithPath: [[NSBundle mainBundle] pathForResource:@"sample" ofType:@"m4a"]]; - (void)audioPlayerDidFinishPlaying:(AVAudioPlayer *)player successfully:(BOOL)flag { if (flag == NO) NSLog(@"Playback finished unsuccessfully"); [player setCurrentTime:0.]; [self updateViewForPlayerState]; } can anyone please help me on 1. how to load the array with all the 10 tracks which are added to resources folder 2. and when I hit play, player should start the first track. when the 1st track ends 2nd track should start and so on for the remaining tracks. Thank You

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  • blackberry implement audio player

    - by Prasad
    Hi, I am developing an application which let users to hear songs online. And I used Blackberry Player and Manager APIs. My application works fine and I can play songs. Now I wan't to add more controls to it. As an example I want pause, play songs. Mute the sound, Control the volume. Display the progress of the play back. Display the current time position of the song like that. I started research on that. And I tried to do that with PlayerListener. But unfortunately all the time I am getting IllegalStateException. So I can't go ahead with that research. As a help can someone please tell me how can I implement above kind of controls for a player. Appreciate if someone can post a sample code to do that. Further I will put my playback source code here. public void run() { try { p = Manager.createPlayer(requestedSong + SystemSettings.strNetwork); p.setLoopCount(1); p.start(); } catch (IOException ioe) { } catch (MediaException me) { } } public void run() { try { p = Manager.createPlayer(strSongURL); p.setLoopCount(1); p.start(); } catch (IOException ioe) { } catch (MediaException me) { } } Thank you very much. Prasad

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  • Recognising tone of the audio

    - by terabytest
    Hi, I have a guitar and I need my pc to be able to tell what note is being played, recognizing the tone. Is it possible to do it in python, also is it possible with pygame? Being able of doing it in pygame would be very helpful.

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  • Streaming audio (YouTube)

    - by wvd
    Hello all, I'm writing a CLI for a music-media-platform. One of the features is going to be that you can directly play YouTube videos from the CLI. I don't really have an idea to do it but this one sounded the most reasonable: I'm going to use of those sites where you can download music from YouTube, e.g. http://keepvid.com/ - then I directly stream & play this -- but I have one problem. Is there any Python library capable of doing this and if so, do you have any concrete examples? I've been looking but found nothing, even not with gstreamer. Thanks, William van Doorn

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