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  • Convert Audio File to text using System.Speech

    - by Kushal Kalambi
    I am looking to convert a .wav file recorded through an android phone at 16000 to text using C#; namely the System.Speech namespace. My code is mentioned below; recognizer.SetInputToWaveFile(Server.MapPath("~/spoken.wav")); recognizer.LoadGrammar(new DictationGrammar()); RecognitionResult result = recognizer.Recognize(); label1.Text = result.Text; The is working perfectly with sample .wav "Hello world" file. However when i record something on teh phone and try to convert to on the pc, the converted text is no where close to what i had recoreded. Is there some way to make sure the audio file is transcribed accurately?

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  • Playing audio from a wav file in iPhone SpeakHere example

    - by Mo
    I'm working with the iPhone SpeakHere example, and I would like to be able to play audio from either the mic (as in the example) or from a wav file. I have working code to play from a particular wav file, which looks like this: NSString *path = [[NSBundle mainBundle] pathForResource:@"basketBall" ofType:@"wav"]; AVAudioPlayer* theAudio=[[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:path] error:NULL]; theAudio.delegate = self; [theAudio play]; So I'm fine with actually getting the wav to play in the application (I can hook it up to a button, etc.) but I would like it to also behave the same way pushing the "Play" button does after recorded speech, in that it should be connected to the same visualization (which I have modified quite a bit, but essentially shows the current volume, among other things). Thanks for your help!

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  • Linux, C++ audio capturing (just microphone) library

    - by TheOm3ga
    I'm developing a musical game, it's like a singstar but instead of singing, you have to play the recorder. It's called oFlute, and it's still in early development stage. In the game, I capture the microphone input, then run a simple FFT analysis and compare the results to typical recorder's frequencies, thus getting the played note. At the beginning, the audio library I was using was RtAudio, but I don't remember why I switched to PortAudio, which is what I'm currently using. The problem is that, from time to time, either it crashes randomly or stops capturing, like if there were no sound coming from the microphone. My question is, what's the best option to capture microphone input on Linux? I just need to open, read, and close a flow of bytes from the microphone. I've been reading this guide, and (un)surprisingly it says: I don't think that PortAudio is very good API for Unix-like operating systems. So, what do you recommend me?

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  • How to play audio file ios

    - by Camus
    I am trying to play an audio file but I can get it working. I imported the AVFoundation framework. Here is the code: NSString *fileName = [[NSBundle mainBundle] pathForResource:@"Alarm" ofType:@"caf"]; NSURL *url = [[NSURL alloc] initFileURLWithPath:fileName]; NSLog(@"Test: %@ ", url); AVAudioPlayer *audioFile = [[AVAudioPlayer alloc] initWithContentsOfURL:url error:NULL]; audioFile.delegate = self; audioFile.volume = 1; [audioFile play]; I am receiving an error nil string parameter I copied the file to the supporting files folder so the file is there. Can you guys help me? Thanks

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  • Audio File continues to play even on leaving the view

    - by Swastik
    What I am doing is -(void)viewWillAppear:(BOOL)animated{ [NSTimer scheduledTimerWithTimeInterval:0.3 target:self selector:@selector(clickEvent:) userInfo:nil repeats:YES]; } -(void)clickEvent:(NSTimer *)aTimer{ NSDate* finishDate = [NSDate date]; if([finishDate timeIntervalSinceDate: self.startDate] 11 && touched == NO){ NSString *mp3Path = [[[NSBundle mainBundle] resourcePath] stringByAppendingPathComponent:@"test.mp3"]; [self playMusicFile:mp3Path]; NSLog(@"Timer from First Page"); [aTimer invalidate]; //[touchCheckTimer release]; aTimer = nil; } else{ } -(void)playMusicFile:(NSString *)mp3Path{ NSURL *mp3Url = [NSURL fileURLWithPath:mp3Path]; NSError *err; AVAudioPlayer *audPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:mp3Url error:&err]; [self setAudioPlayer1:audPlayer]; if(audioPlayer1) [audioPlayer1 play]; [audPlayer release]; } Now, on pushing another view this audio file keeps playing in the background. Please help!

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  • iphone - Images (slide show) and audio snychronization

    - by Qaiser
    I have 20 images and some audio. I would like to show a single image at a time and change the images at (unequal) intervals. For example, I want to show image 1 for 1.44 seconds and image 2 for 1.67 seconds and so on. Can someone suggest how to go about doing this please? What I have seen are examples that show how to setup an array of images with one field that denotes total time. This causes the images to show for an equal amount of time (each). ... and that not what I am looking for ...

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  • Background audio not working in windows 8 store / metro app

    - by roryok
    I've tried setting background audio through both a mediaElement in XAML <MediaElement x:Name="MyAudio" Source="Assets/Sound.mp3" AudioCategory="BackgroundCapableMedia" AutoPlay="False" /> And programmatically async void setUpAudio() { var package = Windows.ApplicationModel.Package.Current; var installedLocation = package.InstalledLocation; var storageFile = await installedLocation.GetFileAsync("Assets\\Sound.mp3"); if (storageFile != null) { var stream = await storageFile.OpenAsync(Windows.Storage.FileAccessMode.Read); _soundEffect = new MediaElement(); _soundEffect.AudioCategory = AudioCategory.BackgroundCapableMedia; _soundEffect.AutoPlay = false; _soundEffect.SetSource(stream, storageFile.ContentType); } } // and later... _soundEffect.Play(); But neither works for me. As soon as I minimise the app the music fades out

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  • Streamed mp3 only plays for 1 second

    - by angel6
    Hi, I'm using the plaympeg.c (modified) code of smpeg as a media player. I've got ffserver running as a streaming server. I'm a streaming an mp3 file over http. But when I run plaympeg.c, it plays the streamed file only for a second. When I run plaympeg again, it starts off from where it left and plays for 1 second. Does anyone know why this happens an how to fix it? I've tested it out on WMP and it plays the entire file in one go. So, i guess it's not a problem with the streaming or ffserver.conf include include include include /* #ifdef unix */ include include include include include include include define NET_SUPPORT /* General network support */ define HTTP_SUPPORT /* HTTP support */ ifdef NET_SUPPORT include include include include endif include "smpeg.h" ifdef NET_SUPPORT int tcp_open(char * address, int port) { struct sockaddr_in stAddr; struct hostent * host; int sock; struct linger l; memset(&stAddr,0,sizeof(stAddr)); stAddr.sin_family = AF_INET ; stAddr.sin_port = htons(port); if((host = gethostbyname(address)) == NULL) return(0); stAddr.sin_addr = *((struct in_addr *) host-h_addr_list[0]) ; if((sock = socket(AF_INET, SOCK_STREAM, IPPROTO_TCP)) < 0) return(0); l.l_onoff = 1; l.l_linger = 5; if(setsockopt(sock, SOL_SOCKET, SO_LINGER, (char*) &l, sizeof(l)) < 0) return(0); if(connect(sock, (struct sockaddr *) &stAddr, sizeof(stAddr)) < 0) return(0); return(sock); } ifdef HTTP_SUPPORT int http_open(char * arg) { char * host; int port; char * request; int tcp_sock; char http_request[1024]; char c; printf("\nin http_open passed parameter = %s\n",arg); /* Check for URL syntax */ if(strncmp(arg, "http://", strlen("http://"))) return(0); /* Parse URL */ port = 80; host = arg + strlen("http://"); if((request = strchr(host, '/')) == NULL) return(0); request++ = 0; if(strchr(host, ':') != NULL) / port is specified */ { port = atoi(strchr(host, ':') + 1); *strchr(host, ':') = 0; } /* Open a TCP socket */ if(!(tcp_sock = tcp_open(host, port))) { perror("http_open"); return(0); } /* Send HTTP GET request */ sprintf(http_request, "GET /%s HTTP/1.0\r\n" "User-Agent: Mozilla/2.0 (Win95; I)\r\n" "Pragma: no-cache\r\n" "Host: %s\r\n" "Accept: /\r\n" "\r\n", request, host); send(tcp_sock, http_request, strlen(http_request), 0); /* Parse server reply */ do read(tcp_sock, &c, sizeof(char)); while(c != ' '); read(tcp_sock, http_request, 4*sizeof(char)); http_request[4] = 0; if(strcmp(http_request, "200 ")) { fprintf(stderr, "http_open: "); do { read(tcp_sock, &c, sizeof(char)); fprintf(stderr, "%c", c); } while(c != '\r'); fprintf(stderr, "\n"); return(0); } return(tcp_sock); } endif endif void update(SDL_Surface *screen, Sint32 x, Sint32 y, Uint32 w, Uint32 h) { if ( screen-flags & SDL_DOUBLEBUF ) { SDL_Flip(screen); } } /* Flag telling the UI that the movie or song should be skipped */ int done; void next_movie(int sig) { done = 1; } int main(int argc, char *argv[]) { int use_audio, use_video; int fullscreen; int scalesize; int scale_width, scale_height; int loop_play; int i, pause; int volume; Uint32 seek; float skip; int bilinear_filtering; SDL_Surface *screen; SMPEG *mpeg; SMPEG_Info info; char *basefile; SDL_version sdlver; SMPEG_version smpegver; int fd; char buf[32]; int status; printf("\nchecking command line options "); /* Get the command line options */ use_audio = 1; use_video = 1; fullscreen = 0; scalesize = 1; scale_width = 0; scale_height = 0; loop_play = 0; volume = 100; seek = 0; skip = 0; bilinear_filtering = 0; fd = 0; for ( i=1; argv[i] && (argv[i][0] == '-') && (argv[i][1] != 0); ++i ) { if ( strcmp(argv[i], "--fullscreen") == 0 ) { fullscreen = 1; } else if ((strcmp(argv[i], "--seek") == 0)||(strcmp(argv[i], "-S") == 0)) { ++i; if ( argv[i] ) { seek = atol(argv[i]); } } else if ((strcmp(argv[i], "--volume") == 0)||(strcmp(argv[i], "-v") == 0)) { ++i; if (i >= argc) { fprintf(stderr, "Please specify volume when using --volume or -v\n"); return(1); } if ( argv[i] ) { volume = atoi(argv[i]); } if ( ( volume < 0 ) || ( volume 100 ) ) { fprintf(stderr, "Volume must be between 0 and 100\n"); volume = 100; } } else { fprintf(stderr, "Warning: Unknown option: %s\n", argv[i]); } } printf("\nuse video = %d, use audio = %d\n",use_video, use_audio); printf("\ngoing to check input parameters\n"); if defined(linux) || defined(FreeBSD) /* Plaympeg doesn't need a mouse */ putenv("SDL_NOMOUSE=1"); endif /* Play the mpeg files! */ status = 0; for ( ; argv[i]; ++i ) { /* Initialize SDL */ if ( use_video ) { if ((SDL_Init(SDL_INIT_VIDEO) < 0) || !SDL_VideoDriverName(buf, 1)) { fprintf(stderr, "Warning: Couldn't init SDL video: %s\n", SDL_GetError()); fprintf(stderr, "Will ignore video stream\n"); use_video = 0; } printf("\ninitialised video\n"); } if ( use_audio ) { if ((SDL_Init(SDL_INIT_AUDIO) < 0) || !SDL_AudioDriverName(buf, 1)) { fprintf(stderr, "Warning: Couldn't init SDL audio: %s\n", SDL_GetError()); fprintf(stderr, "Will ignore audio stream\n"); use_audio = 0; } } /* Allow Ctrl-C when there's no video output */ signal(SIGINT, next_movie); printf("\nchecking defined supports\n"); /* Create the MPEG stream */ ifdef NET_SUPPORT printf("\ndefined NET_SUPPORT\n"); ifdef HTTP_SUPPORT printf("\ndefined HTTP_SUPPORT\n"); /* Check if source is an http URL */ printf("\nabout to call http_open\n"); printf("\nhere we go\n"); if((fd = http_open(argv[i])) != 0) mpeg = SMPEG_new_descr(fd, &info, use_audio); else endif endif { if(strcmp(argv[i], "-") == 0) /* Use stdin for input */ mpeg = SMPEG_new_descr(0, &info, use_audio); else mpeg = SMPEG_new(argv[i], &info, use_audio); } if ( SMPEG_error(mpeg) ) { fprintf(stderr, "%s: %s\n", argv[i], SMPEG_error(mpeg)); SMPEG_delete(mpeg); status = -1; continue; } SMPEG_enableaudio(mpeg, use_audio); SMPEG_enablevideo(mpeg, use_video); SMPEG_setvolume(mpeg, volume); /* Print information about the video */ basefile = strrchr(argv[i], '/'); if ( basefile ) { ++basefile; } else { basefile = argv[i]; } if ( info.has_audio && info.has_video ) { printf("%s: MPEG system stream (audio/video)\n", basefile); } else if ( info.has_audio ) { printf("%s: MPEG audio stream\n", basefile); } else if ( info.has_video ) { printf("%s: MPEG video stream\n", basefile); } if ( info.has_video ) { printf("\tVideo %dx%d resolution\n", info.width, info.height); } if ( info.has_audio ) { printf("\tAudio %s\n", info.audio_string); } if ( info.total_size ) { printf("\tSize: %d\n", info.total_size); } if ( info.total_time ) { printf("\tTotal time: %f\n", info.total_time); } /* Set up video display if needed */ if ( info.has_video && use_video ) { const SDL_VideoInfo *video_info; Uint32 video_flags; int video_bpp; int width, height; /* Get the "native" video mode */ video_info = SDL_GetVideoInfo(); switch (video_info->vfmt->BitsPerPixel) { case 16: case 24: case 32: video_bpp = video_info->vfmt->BitsPerPixel; break; default: video_bpp = 16; break; } if ( scale_width ) { width = scale_width; } else { width = info.width; } width *= scalesize; if ( scale_height ) { height = scale_height; } else { height = info.height; } height *= scalesize; video_flags = SDL_SWSURFACE; if ( fullscreen ) { video_flags = SDL_FULLSCREEN|SDL_DOUBLEBUF|SDL_HWSURFACE; } video_flags |= SDL_ASYNCBLIT; video_flags |= SDL_RESIZABLE; screen = SDL_SetVideoMode(width, height, video_bpp, video_flags); if ( screen == NULL ) { fprintf(stderr, "Unable to set %dx%d video mode: %s\n", width, height, SDL_GetError()); continue; } SDL_WM_SetCaption(argv[i], "plaympeg"); if ( screen->flags & SDL_FULLSCREEN ) { SDL_ShowCursor(0); } SMPEG_setdisplay(mpeg, screen, NULL, update); SMPEG_scaleXY(mpeg, screen->w, screen->h); } else { SDL_QuitSubSystem(SDL_INIT_VIDEO); } /* Set any special playback parameters */ if ( loop_play ) { SMPEG_loop(mpeg, 1); } /* Seek starting position */ if(seek) SMPEG_seek(mpeg, seek); /* Skip seconds to starting position */ if(skip) SMPEG_skip(mpeg, skip); /* Play it, and wait for playback to complete */ SMPEG_play(mpeg); done = 0; pause = 0; while ( ! done && ( pause || (SMPEG_status(mpeg) == SMPEG_PLAYING) ) ) { SDL_Event event; while ( use_video && SDL_PollEvent(&event) ) { switch (event.type) { case SDL_VIDEORESIZE: { SDL_Surface *old_screen = screen; SMPEG_pause(mpeg); screen = SDL_SetVideoMode(event.resize.w, event.resize.h, screen->format->BitsPerPixel, screen->flags); if ( old_screen != screen ) { SMPEG_setdisplay(mpeg, screen, NULL, update); } SMPEG_scaleXY(mpeg, screen-w, screen-h); SMPEG_pause(mpeg); } break; case SDL_KEYDOWN: if ( (event.key.keysym.sym == SDLK_ESCAPE) || (event.key.keysym.sym == SDLK_q) ) { // Quit done = 1; } else if ( event.key.keysym.sym == SDLK_RETURN ) { // toggle fullscreen if ( event.key.keysym.mod & KMOD_ALT ) { SDL_WM_ToggleFullScreen(screen); fullscreen = (screen-flags & SDL_FULLSCREEN); SDL_ShowCursor(!fullscreen); } } else if ( event.key.keysym.sym == SDLK_UP ) { // Volume up if ( volume < 100 ) { if ( event.key.keysym.mod & KMOD_SHIFT ) { // 10+ volume += 10; } else if ( event.key.keysym.mod & KMOD_CTRL ) { // 100+ volume = 100; } else { // 1+ volume++; } if ( volume 100 ) volume = 100; SMPEG_setvolume(mpeg, volume); } } else if ( event.key.keysym.sym == SDLK_DOWN ) { // Volume down if ( volume 0 ) { if ( event.key.keysym.mod & KMOD_SHIFT ) { volume -= 10; } else if ( event.key.keysym.mod & KMOD_CTRL ) { volume = 0; } else { volume--; } if ( volume < 0 ) volume = 0; SMPEG_setvolume(mpeg, volume); } } else if ( event.key.keysym.sym == SDLK_PAGEUP ) { // Full volume volume = 100; SMPEG_setvolume(mpeg, volume); } else if ( event.key.keysym.sym == SDLK_PAGEDOWN ) { // Volume off volume = 0; SMPEG_setvolume(mpeg, volume); } else if ( event.key.keysym.sym == SDLK_SPACE ) { // Toggle play / pause if ( SMPEG_status(mpeg) == SMPEG_PLAYING ) { SMPEG_pause(mpeg); pause = 1; } else { SMPEG_play(mpeg); pause = 0; } } else if ( event.key.keysym.sym == SDLK_RIGHT ) { // Forward if ( event.key.keysym.mod & KMOD_SHIFT ) { SMPEG_skip(mpeg, 100); } else if ( event.key.keysym.mod & KMOD_CTRL ) { SMPEG_skip(mpeg, 50); } else { SMPEG_skip(mpeg, 5); } } else if ( event.key.keysym.sym == SDLK_LEFT ) { // Reverse if ( event.key.keysym.mod & KMOD_SHIFT ) { } else if ( event.key.keysym.mod & KMOD_CTRL ) { } else { } } else if ( event.key.keysym.sym == SDLK_KP_MINUS ) { // Scale minus if ( scalesize > 1 ) { scalesize--; } } else if ( event.key.keysym.sym == SDLK_KP_PLUS ) { // Scale plus scalesize++; } else if ( event.key.keysym.sym == SDLK_f ) { // Toggle filtering on/off if ( bilinear_filtering ) { SMPEG_Filter *filter = SMPEGfilter_null(); filter = SMPEG_filter( mpeg, filter ); filter-destroy(filter); bilinear_filtering = 0; } else { SMPEG_Filter *filter = SMPEGfilter_bilinear(); filter = SMPEG_filter( mpeg, filter ); filter-destroy(filter); bilinear_filtering = 1; } } break; case SDL_QUIT: done = 1; break; default: break; } } SDL_Delay(1000/2); } SMPEG_delete(mpeg); } SDL_Quit(); if defined(HTTP_SUPPORT) if(fd) close(fd); endif return(status); }

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  • Record/Playback with AudioQueue on iPhone

    - by Biranchi
    Hi, I am currently using Audio Queues on the iPhone to record and playback audio. What I would like to be able to do is to record some audio, allow the user to pause the record queue, and to seek back and forward through the audio to select a position from where they can start recording from again. I have got over the seeking issue by making the playback AudioQueueBuffer sizes small enough so that the play audio queue callback happens at a rate that allows the user to use a slider control to hear the audio as they adjust the slider back and forth. I think I can achieve the recording at a new position by setting the inStartingPacket parameter of the AudioFileWritePackets function that I call from the Audio Recording Queue callback. The trouble is this only inserts audio over the previously recorded audio. The file length obviously doesn't change so if the user were to go backwards and record less audio than before, the old audio still remains after the end of the newly recorded audio. Is there a way I can get the AudioFile to truncate at the point the user starts to insert the new audio, is there some other way I can remove the old audio starting at the insert position or is there a better way about going about this task? Thanks

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  • Android SDK : Playing video using mms protocol

    - by GX
    Hello, Using the Android SDK, is it possible to play a video stream using the MMS protocol I am streaming video from a PC using windows media. I can use Windows Media Player to play the stream by just inputting the following URL in Windows Media Player mms://192.168.223.194:8081 Is it possible to play the same stream using the Android SDK ? Thanks

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  • Play RTSP/MMS/Http live video feeds in WPF

    - by James Cadd
    I'd like to pull a live video feed into WPF but the MediaElement doesn't appear to support these protocols. An example video stream is here (BP oil leak live feed): http://mfile.akamai.com/97892/live/reflector:45683.asx?bkup=45684 Are there any solutions for playing live streaming formats in WPF?

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  • Simple 2-color differential image compression

    - by Groo
    Is there an efficient, quick and simple example of doing differential b/w image compression? Or even better, some simple (but lossless) streaming technique which could accept a number of frames as input? I have a simple b/w image (320x200) stream, displaying something similar to a LED display, which is updated about once a second using AJAX. Images are pretty similar most of the time, so if I subtracted them, result would compress pretty well (even with simple RLE). Is something like this available?

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  • How to input live video Stream to Microsoft Encoder.

    - by GautamB
    There is a facility in Microsoft Encoder to give input from a file or from a device such as webcam for streaming. But i haven't found a way to give live video stream to encoder. i.e. How to make encoder listen to particular UDP port. From which encoder will take input stream and encode it and push to windows media server. Any help will be appreciated. Thanks, Gautam B.

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  • How to stream XML data using XOM?

    - by Jonik
    Say I want to output a huge set of search results, as XML, into a PrintWriter or an OutputStream, using XOM. The resulting XML would look like this: <?xml version="1.0" encoding="UTF-8"?> <resultset> <result> [child elements and data] </result> ... ... [1000s of result elements more] </resultset> Because the resulting XML document could be big (tens or hundreds of megabytes, perhaps), I want to output it in a streaming fashion (instead of creating the whole Document in memory and then writing that). The granularity of outputting one <result> at a time is fine, so I want to generate one <result> after another, and write it into the stream. Assume there's already a method that helps with iterating the results and generating Element objects: public nu.xom.Element getNextResult(); So I'd simply like to do something like this pseudocode (automatic flushing enabled, so don't worry about that) : open stream/writer write declaration write start tag for <resultset> while more results: write next <result> element write end tag for <resultset> close stream/writer I've been looking at Serializer, but the necessary methods, writeStartTag(Element), writeEndTag(Element), write(DocType) are protected, not public! Is there no other way than to subclass Serializer to be able to use those methods, or to manually write the start and end tags directly into the stream as Strings, bypassing XOM altogether? (The latter wouldn't be too bad in this simple example, but in the general case it would get quite ugly.) Am I missing something or is XOM just not made for this? With dom4j I could do this easily using XMLWriter - it has constructors that take a Writer or OutputStream, and methods writeOpen(Element), writeClose(Element), writeDocType(DocumentType) etc. Compare to XOM's Serializer where the only public write method is the one that takes a whole Document. Please refrain from answering if you're not familiar with XOM! I specifically want to know if and how you can do this kind of streaming with that library. (This is related to my question about the best dom4j replacement where XOM is a strong contender.)

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  • How many copies of files are needed by video server?

    - by Trilok
    A quick question. How many copies of the same movie are kept in a video server (a video streaming server)? Suppose a particular video is at max requested by 1000 users at the same instant of time, how many copies would be sufficient so that parallel streams can be provided to each user? Ideally 1 copy would solve the purpose, but what is the optimum number keeping the bandwidth and simultaneous access in mind?

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  • Protect flash video from download/right protect

    - by Smyles
    Is it possible to protect flv files from download? I'd like to protect my files from download but I don't have the money for a streaming server which I think provides some sort of protection. The files are streamed via PHP and are located in an upload folder on my server. I've used PHP to ensure that only subscribers can view the video but I basically want to go a step further and prevent subscribers from, upon login, downloading my videos with downloaders such as Sothink Flv Downloader for Firefox.

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  • I want to know when a file is down downloading

    - by paulj3000
    Hi, I have a file which I want users to download only once. After it's done downloading, the file is no longer available. Outside of setting up a streaming system, is there any way I can set some sort of callback up to say the file is done downloading on the client's computer? Thanks

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  • tool to find out distance in terms of no. of hops in unix

    - by mawia
    Hi! all, I am writing an application for video streaming.In the application server is required to know the distance of the client from it self in terms of hop number.My question is,is there any tool/method other than traceroute available in unix environment to find it? I also need to find out the geographical location of the client.So is their any tool/method for this as well? Any help in this regard will be highly appreciated. Thanks in advance. Mawia

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  • audio cd s not burning to mp3 format-burning to wav format in k3b and brasero using ubuntu 12.04.2

    - by robert
    It started in ubuntu 13.04-I was doing what I usually do,I opened brasero to make an audio cd from a few mp3 audio files..When burned I noticed the files on cd were in wav format.I then tried k3b with the same result.At that point and because of several issues with 13.04 I formatted my hdd and dropped back to ubuntu 12.04.On 12.04 I tried brasero and k3b once again with same results.I know that when I used to burn cd s using brasero they were burned to cd in mp3 format not wave.Can anyone tell me a fix for this?I have restricted codecs installed.

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  • Using Audio Queue Services to play PCM data over a socket connection

    - by Rohan
    I'm writing a remote desktop client for the iPhone and I'm trying to implement audio redirection. The client is connected to the server over a socket connection, and the server sends 32K chunks of PCM data at a time. I'm trying to use AQS to play the data and it plays the first two seconds (1 buffer worth). However, since the next chunk of data hasn't come in over the socket yet, the next AudioQueueBuffer is empty. When the data comes in, I fill the next available buffer with the data and enqueue it with AudioQueueEnqueueBuffer. However, it never plays these buffers. Does the queue stop playing if there are no buffers in the queue, even if you later add a buffer? Here's the relevant part of the code: void wave_out_write(STREAM s, uint16 tick, uint8 index) { if(items_in_queue == NUM_BUFFERS){ return; } if(!playState.busy){ OSStatus status; status = AudioQueueNewOutput(&playState.dataFormat, AudioOutputCallback, &playState, CFRunLoopGetCurrent(), NULL, 0, &playState.queue); if(status == 0){ for(int i=0; i<NUM_BUFFERS; i++){ AudioQueueAllocateBuffer(playState.queue, 40000, &playState.buffers[i]); } AudioQueueAddPropertyListener(playState.queue, kAudioQueueProperty_IsRunning, MyAudioQueuePropertyListenerProc, &playState); status = AudioQueueStart(playState.queue, NULL); if(status ==0){ playState.busy = True; } else{ return; } } else{ return; } } playState.buffers[queue_hi]->mAudioDataByteSize = s->size; memcpy(playState.buffers[queue_hi]->mAudioData, s->data, s->size); AudioQueueEnqueueBuffer(playState.queue, playState.buffers[queue_hi], 0, 0); queue_hi++; queue_hi = queue_hi % NUM_BUFFERS; items_in_queue++; } void AudioOutputCallback(void* inUserData, AudioQueueRef outAQ, AudioQueueBufferRef outBuffer) { PlayState *playState = (PlayState *)inUserData; items_in_queue--; } Thanks!

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  • Stopping and Play button for Audio (Android)

    - by James Rattray
    I have this problem, I have some audio I wish to play... And I have two buttons for it, 'Play' and 'Stop'... Problem is, after I press the stop button, and then press the Play button, nothing happens. -The stop button stops the song, but I want the Play button to play the song again (from the start) Here is my code: final MediaPlayer mp = MediaPlayer.create(this, R.raw.megadeth); And then the two public onclicks: (For playing...) button.setOnClickListener(new View.OnClickListener() { public void onClick(View v) { // Perform action on click button.setText("Playing!"); try { mp.prepare(); } catch (IllegalStateException e) { // TODO Auto-generated catch block e.printStackTrace(); } catch (IOException e) { // TODO Auto-generated catch block e.printStackTrace(); } mp.start(); // } }); And for stopping the track... final Button button2 = (Button) findViewById(R.id.cancel); button2.setOnClickListener(new View.OnClickListener() { public void onClick(View v) { mp.stop(); mp.reset(); } }); Can anyone see the problem with this? If so could you please fix it... (For suggest) Thanks alot... James

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