Search Results

Search found 4698 results on 188 pages for 'audio recording'.

Page 49/188 | < Previous Page | 45 46 47 48 49 50 51 52 53 54 55 56  | Next Page >

• Microphone doesn't work

  - by mandy

  I'm having a trouble with my built in microphone. Even if I use headphone with mic, it doesn't really work. The weird thing is, if I clap the green lines of the speaker icon jumps, but if I speak it doesn't. I have also tried some recordings, but I can not hear myself and adjusting the volume didn't help at all. I tried to restore it, still no change. I have updated it in the device manager, but it said there that it's up to date and the devices are working properly. Until I decided to recover the whole system (wherein I pressed zero and switch on button) to my surprise, the settings became different, most programs were deleted, even my files. It's like it was formatted and I'm so sad that the mic was not fixed. I really don't know what to do now. My laptop model is Toshiba satellite m840. I want to it return to its settings/set up just before I Recover the system and bring back all the programs that we're installed and of course, most of all, to fix my microphone so I can use Skype again and other video calling application.. I hope someone could help me. Thanks a lot!

  Read the article

• Polycom webcast screen capture software?

  - by Moshe

  We have someone here who uses Polycom to do web conferences to another site overseas. Is there a way to capture the stream on Polycom systems?

  Read the article

• get_iplayer _ new problem

  - by Squiggle

  I have get_iplayer installed on a Windows 7 PC and, generally, it has worked without too many problems. Now I find that I cannot download "The Sinking of the Laconia" from the BBC. Both episodes are still available for viewing (although only episode 2 has a live ' download' option). An error message saying 'ERROR: RTMP_ReadPacket, failed to read RTMP packet header' may be relevant? If anyone can help before the programme becomes unavailable, I'd be grateful

  Read the article

• Android:Playing bigger size audio wav sound file produces crash

  - by user187532

  Hi Android experts, I am trying to play the bigger size audio wav file(which is 20 mb) using the following code(AudioTrack) on my Android 1.6 HTC device which basically has less memory. But i found device crash as soon as it executes reading, writing and play. But the same code works fine and plays the lesser size audio wav files(10kb, 20 kb files etc) very well. P.S: I should play PCM(.wav) buffer sound, the reason behind why i use AudioTrack here. Though my device has lesser memory, how would i read bigger audio files bytes by bytes and play the sound to avoid crashing due to memory constraints. private void AudioTrackPlayPCM() throws IOException { String filePath = "/sdcard/myWav.wav"; // 8 kb file byte[] byteData = null; File file = null; file = new File(filePath); byteData = new byte[(int) file.length()]; FileInputStream in = null; try { in = new FileInputStream( file ); in.read( byteData ); in.close(); } catch (FileNotFoundException e) { // TODO Auto-generated catch block e.printStackTrace(); } int intSize = android.media.AudioTrack.getMinBufferSize(8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_8BIT); AudioTrack at = new AudioTrack(AudioManager.STREAM_MUSIC, 8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_8BIT, intSize, AudioTrack.MODE_STREAM); at.play(); at.write(byteData, 0, byteData.length); at.stop(); at.release(); } Could someone guide me please to play the AudioTrack code for bigger size wav files?

  Read the article

• Chat with Audio/Video and DesktopSharing

  - by RavIncredible

  Hi All, I am working on an application where I like to bundle 3 things: 1. Chat 2. Audio / Video Conversation 3. Desktop Sharing I would like to know the approach and where to look in for this, few of the things that I am aware of are: Chat and Audio – I can go with Jabber server and configure any SIP server like asterix for audio calls. Desktop Sharing – I have read about silverlite coming up with Desktop Sharing modules, but what would be only targeted to Windows. I would like to have sharing for windows, mac and linux OS. I don’t mind building separate clients for each. But I like to know which common protocol has to be used for Desktop sharing. In other words something similar to team viewer. Please suggest. Video Conference – I totally don’t have any idea about this. The application that I am supposed to build has to target the below platforms: 1. Window, Mac, Linux Desktop 2. iPhone, iPad and Android Devices. Would appreciate any help or reference or links to any of the topics (Chat, A/V and desktop sharing). Thanks Ravi

  Read the article

• Nginx flv audio pseudo stream works but video is not loading

  - by sarah

  I am working on a development server for a company & they want nginx webserver to work with. So the requirements for their company is, it should be capable of doing following things i.e hotlink protection, mp4 & flv pseudo stream & secure streaming. However nginx fulfills their requirements and i am configuring their server from past 2 days as i am new to this field so i've only acheived hotlinking prevention in past 2 days. But the problem on which i am stuck is flv pseudo streaming, to make work to mp4 pseudo stream it was just a piece of paper but i am really fuc*ed up with flv pseudo stream. I have converted my flv videos with flvmdi tools to insert many keyframes but the problem is , when i try to seek video from following keyframes that are generated by flvmdi i.e test.flv?start=2681223, video does not load but audio pseudo works fine. So it means no problem with my flv configuration in nginx.conf file. And the forum that i used to compile my nginx-1.2.1 is http://h264.code-shop.com/trac/wiki/Mod-H264-Streaming-Nginx-Version2 & by adding additional module --with-http_flv_module. This forum is really active, hopes i will resolve my problem as soon as you guys will provide me some guide.

  Read the article

• Reading a WAV file into VST.Net to process with a plugin

  - by Paul

  Hello, I'm trying to use the VST.Net and NAudio frameworks to build an application that processes audio using a VST plugin. Ideally, the application should load a wav or mp3 file, process it with the VST, and then write a new file. I have done some poking around with the VST.Net library and was able to compile and run the samples (specifically the VST Host one). What I have not figured out is how to load an audio file into the program and have it write a new file back out. I'd like to be able to configure the properties for the VST plugin via C#, and be able to process the audio with 2 or more consecutive VSTs. Using NAudio, I was able to create this simple script to copy an audio file. Now I just need to get the output from the WaveFileReader into the VST.Net framework somehow. private void processAudio() { reader = new WaveFileReader("c:/bass.wav"); writer = new WaveFileWriter("c:/bass-copy.wav", reader.WaveFormat); int read; while ((read = reader.Read(buffer, 0, buffer.Length)) > 0) { writer.WriteData(buffer, 0, read); } textBox1.Text = "done"; reader.Close(); reader.Dispose(); writer.Close(); writer.Dispose(); } Please help!! Thanks References: http://vstnet.codeplex.com (VST.Net) http://naudio.codeplex.com (NAudio)

  Read the article

• How to read/write high-resolution (24-bit, 8 channel) .wav files in Java?

  - by dB'

  I'm trying to write a Java application that manipulates high resolution .wav files. I'm having trouble importing the audio data, i.e. converting the .wav file into an array of doubles. When I use a standard approach an exception is thrown. AudioFileFormat as = AudioSystem.getAudioFileFormat(new File("orig.wav")); --> javax.sound.sampled.UnsupportedAudioFileException: file is not a supported file type Here's the file format info according to soxi: dB$ soxi orig.wav soxi WARN wav: wave header missing FmtExt chunk Input File : 'orig.wav' Channels : 8 Sample Rate : 96000 Precision : 24-bit Duration : 00:00:03.16 = 303526 samples ~ 237.13 CDDA sectors File Size : 9.71M Bit Rate : 24.6M Sample Encoding: 32-bit Floating Point PCM Can anyone suggest the simplest method for getting this audio into Java? I've tried using a few techniques. As stated above, I've experimented with the Java AudioSystem (on both Mac and Windows). I've also tried using Andrew Greensted's WavFile class, but this also fails (WavFileException: Compression Code 3 not supported). One workaround is to convert the audio to 16 bits using sox (with the -b 16 flag), but this is suboptimal since it increases the noise floor. Incidentally, I've noticed that the file CAN be read by libsndfile. Is my best bet to write a jni wrapper around libsndfile, or can you suggest something quicker? Note that I don't need to play the audio, I just need to analyze it, manipulate it, and then write it out to a new .wav file. * UPDATE * I solved this problem by modifying Andrew Greensted's WavFile class. His original version only read files encoded as integer values ("format code 1"); my files were encoded as floats ("format code 3"), and that's what was causing the problem. I'll post the modified version of Greensted's code when I get a chance. In the meantime, if anyone wants it, send me a message.

  Read the article

• Help file not working

  - by meryl

  Hi, can anyone help me ? I wanto to play an audio file and whenever I press the stop button , the already played part of the file should be saved. Unfortunately , what I get is an audio file (.wav) which actually is unplayable. Thanks //**************************** void play_cut() { try { // First, we get the format of the input file final AudioFileFormat.Type fileType = AudioSystem.getAudioFileFormat(inputAudio).getType(); // Then, we get a clip for playing the audio. c = AudioSystem.getClip(); // We get a stream for playing the input file. AudioInputStream ais = AudioSystem.getAudioInputStream(inputAudio); // We use the clip to open (but not start) the input stream c.open(ais); // We get the format of the audio codec (not the file format we got above) final AudioFormat audioFormat = ais.getFormat(); c.start(); AudioInputStream startStream = new AudioInputStream(new FileInputStream(inputAudio), audioFormat, c.getLongFramePosition()); AudioSystem.write(startStream, fileType, outputAudio); } catch (UnsupportedAudioFileException e) { e.printStackTrace(); } catch (IOException e) { e.printStackTrace(); } catch (LineUnavailableException e) { e.printStackTrace(); } }// end play_cut //****************************

  Read the article

• Why is iTunes starting and stopping play randomly, and how do I stop it?

  - by Chris R

  Since yesterday morning my copy of iTunes has been starting and stopping randomly. If iTunes is not running, then it opens and sometimes begins playing, other times sits idle. Eventually, after a random interval it will begin playing a song, and then stop, and so on... Needless to say, it's driving me mad. (Mac OSX, 10.6.3, on a new-ish (< 1 year old) 24" iMac) I've made five changes to my system that may or may not be connected to this: My office phone was replaced with a Linksys IP Phone, which necessitated a change to my networking; where previously my Mac was connected directly to the office network port, now it is connected through the phone. My network connection now uses auto link detection in lieu of forcing 100Mbit I unpaired my bluetooth headset. I removed the USB audio device associated with another headset. I upgraded to Safari 5. I don't use it as a primary browser, but it's often open to run web apps that I'm developing. All of these things happened in pretty close proximity to each other, so one or more of them may be the culprit. One other thing that may or may not be related; for some reason my built-in microphone is no longer picking up audio. It seems like this might be connected to the iTunes issue, because it happened around the same time. In terms of things that I've tried in order to solve this, I'm at a bit of a loss. I followed the instructions at http://developer.apple.com/mac/library/technotes/tn2004/tn2124.html#SECLAUNCHDLOGGING to enable detailed launchd logging to see if I could track down which process was asking iTunes to open (when it's not already open) but I wasn't able to make heads or tails of the output. I'm not even sure if I'm looking in the right place, to be honest; it actually acts like something is activating the application with AppleScript, but I have no processes running that are doing that, as far as I know. I'm running a few apps that have iTunes integration: Adium, iChat with Chax, Quicksilver. None of these have been changed lately, so I consider them low risks of causing this, but it's not impossible. Moreover, I'm not using any of those features intentionally. This is a snippet of launchd debug logging from around the time it just launched: 10-06-09 9:14:29 AM com.apple.launchd[1] Dispatching kevent... 10-06-09 9:14:29 AM com.apple.launchd[1] KEVENT[0]: udata = 0x10002b230 data = 0x30 ident = 5 filter = EVFILT_READ flags = EV_ADD|EV_RECEIPT fflags = 0x0 10-06-09 9:14:29 AM com.apple.launchd[1] Dispatching kevent... 10-06-09 9:14:29 AM com.apple.launchd[1] KEVENT[0]: udata = 0x100802000 data = 0x0 ident = 26 filter = EVFILT_PROC flags = EV_ADD|EV_RECEIPT|EV_CLEAR fflags = NOTE_FORK 10-06-09 9:14:29 AM com.apple.launchd[1] (com.apple.coreservicesd[26]) Dispatching kevent callback. 10-06-09 9:14:29 AM com.apple.launchd[1] (com.apple.coreservicesd[26]) EVFILT_PROC event for job: 10-06-09 9:14:29 AM com.apple.launchd[1] KEVENT[0]: udata = 0x1004076f0 data = 0x0 ident = 26 filter = EVFILT_PROC flags = EV_ADD|EV_RECEIPT|EV_CLEAR fflags = NOTE_FORK 10-06-09 9:14:29 AM com.apple.launchd[1] (com.apple.coreservicesd[26]) fork()ed 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave) Conceived 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22197]) Created PID 22197 anonymously by PPID 26 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22197]) Looking up per user launchd for UID: 0 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22197]) Per user launchd job found for UID: 505 10-06-09 9:14:29 AM com.apple.launchd[1] System: Looking up service com.apple.system.notification_center 10-06-09 9:14:29 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.system.notification_center 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22197]) Looking up per user launchd for UID: 0 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22197]) Per user launchd job found for UID: 505 10-06-09 9:14:29 AM com.apple.launchd[1] System: Looking up service com.apple.system.DirectoryService.libinfo_v1 10-06-09 9:14:29 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.system.DirectoryService.libinfo_v1 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22197]) Looking up per user launchd for UID: 0 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22197]) Per user launchd job found for UID: 505 10-06-09 9:14:29 AM com.apple.launchd[1] System: Looking up service com.apple.system.DirectoryService.membership_v1 10-06-09 9:14:29 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.system.DirectoryService.membership_v1 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22197]) Looking up per user launchd for UID: 0 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22197]) Per user launchd job found for UID: 505 10-06-09 9:14:29 AM com.apple.launchd[1] System: Looking up service com.apple.CoreServices.coreservicesd 10-06-09 9:14:29 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.CoreServices.coreservicesd 10-06-09 9:14:29 AM com.apple.launchd[1] Dispatching kevent... 10-06-09 9:14:29 AM com.apple.launchd[1] KEVENT[0]: udata = 0x100802000 data = 0x0 ident = 22197 filter = EVFILT_PROC flags = EV_ADD|EV_RECEIPT|EV_CLEAR fflags = NOTE_EXIT 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22197]) Dispatching kevent callback. 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22197]) EVFILT_PROC event for job: 10-06-09 9:14:29 AM com.apple.launchd[1] KEVENT[0]: udata = 0x100401720 data = 0x0 ident = 22197 filter = EVFILT_PROC flags = EV_ADD|EV_RECEIPT|EV_CLEAR fflags = NOTE_EXIT 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22197]) Reaping 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave) Total rusage: utime 0.000000 stime 0.000000 maxrss 0 ixrss 0 idrss 0 isrss 0 minflt 0 majflt 0 nswap 0 inblock 0 oublock 0 msgsnd 0 msgrcv 0 nsignals 0 nvcsw 0 nivcsw 0 10-06-09 9:14:29 AM com.apple.launchd[1] (0x100401720.anonymous.lssave) Removed 10-06-09 9:14:30 AM com.apple.launchd[1] Dispatching kevent... 10-06-09 9:14:30 AM com.apple.launchd[1] KEVENT[0]: udata = 0x100802000 data = 0x0 ident = 22197 filter = EVFILT_PROC flags = EV_ADD|EV_RECEIPT|EV_CLEAR|EV_EOF|EV_ONESHOT fflags = NOTE_REAP 10-06-09 9:14:32 AM com.apple.launchd[1] Dispatching kevent... 10-06-09 9:14:32 AM com.apple.launchd[1] KEVENT[0]: udata = 0x10002b230 data = 0x30 ident = 5 filter = EVFILT_READ flags = EV_ADD|EV_RECEIPT fflags = 0x0 10-06-09 9:14:33 AM com.apple.launchd[1] Dispatching kevent... 10-06-09 9:14:33 AM com.apple.launchd[1] KEVENT[0]: udata = 0x100802000 data = 0x0 ident = 143 filter = EVFILT_PROC flags = EV_ADD|EV_RECEIPT|EV_CLEAR fflags = NOTE_FORK 10-06-09 9:14:33 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Dispatching kevent callback. 10-06-09 9:14:33 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) EVFILT_PROC event for job: 10-06-09 9:14:33 AM com.apple.launchd[1] KEVENT[0]: udata = 0x10041e9a0 data = 0x0 ident = 143 filter = EVFILT_PROC flags = EV_ADD|EV_RECEIPT|EV_CLEAR fflags = NOTE_FORK 10-06-09 9:14:33 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) fork()ed 10-06-09 9:14:33 AM com.apple.launchd[1] System: Looking up service com.apple.distributed_notifications.2 10-06-09 9:14:33 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.distributed_notifications.2 10-06-09 9:14:33 AM com.apple.launchd[1] System: Looking up service com.apple.system.notification_center 10-06-09 9:14:33 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.system.notification_center 10-06-09 9:14:33 AM com.apple.launchd[1] System: Looking up service com.apple.system.DirectoryService.libinfo_v1 10-06-09 9:14:33 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.system.DirectoryService.libinfo_v1 10-06-09 9:14:33 AM com.apple.launchd[1] System: Looking up service com.apple.system.DirectoryService.membership_v1 10-06-09 9:14:33 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.system.DirectoryService.membership_v1 10-06-09 9:14:33 AM com.apple.launchd[1] System: Looking up service com.apple.CoreServices.coreservicesd 10-06-09 9:14:33 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.CoreServices.coreservicesd 10-06-09 9:14:33 AM com.apple.launchd[1] System: Looking up service com.apple.SystemConfiguration.configd 10-06-09 9:14:33 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.SystemConfiguration.configd 10-06-09 9:14:33 AM com.apple.launchd[1] System: Looking up service com.apple.audio.coreaudiod 10-06-09 9:14:33 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.audio.coreaudiod 10-06-09 9:14:34 AM com.apple.launchd[1] System: Looking up service com.apple.system.logger 10-06-09 9:14:34 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.system.logger 10-06-09 9:14:35 AM com.apple.launchd[1] Dispatching kevent... 10-06-09 9:14:35 AM com.apple.launchd[1] KEVENT[0]: udata = 0x10002b230 data = 0x30 ident = 5 filter = EVFILT_READ flags = EV_ADD|EV_RECEIPT fflags = 0x0 10-06-09 9:14:35 AM com.apple.launchd[1] System: Looking up service com.apple.DiskArbitration.diskarbitrationd 10-06-09 9:14:35 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.DiskArbitration.diskarbitrationd 10-06-09 9:14:35 AM com.apple.launchd[1] System: Looking up service com.apple.system.logger 10-06-09 9:14:35 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.system.logger 10-06-09 9:14:36 AM com.apple.launchd[1] System: Looking up service com.apple.FSEvents 10-06-09 9:14:36 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.FSEvents 10-06-09 9:14:36 AM com.apple.launchd[1] System: Looking up service com.apple.SystemConfiguration.configd 10-06-09 9:14:36 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.SystemConfiguration.configd 10-06-09 9:14:38 AM com.apple.launchd[1] Dispatching kevent... 10-06-09 9:14:38 AM com.apple.launchd[1] KEVENT[0]: udata = 0x10002b230 data = 0x30 ident = 5 filter = EVFILT_READ flags = EV_ADD|EV_RECEIPT fflags = 0x0 10-06-09 9:14:39 AM com.apple.launchd[1] Dispatching kevent... 10-06-09 9:14:39 AM com.apple.launchd[1] KEVENT[0]: udata = 0x100802000 data = 0x0 ident = 26 filter = EVFILT_PROC flags = EV_ADD|EV_RECEIPT|EV_CLEAR fflags = NOTE_FORK 10-06-09 9:14:39 AM com.apple.launchd[1] (com.apple.coreservicesd[26]) Dispatching kevent callback. 10-06-09 9:14:39 AM com.apple.launchd[1] (com.apple.coreservicesd[26]) EVFILT_PROC event for job: 10-06-09 9:14:39 AM com.apple.launchd[1] KEVENT[0]: udata = 0x1004076f0 data = 0x0 ident = 26 filter = EVFILT_PROC flags = EV_ADD|EV_RECEIPT|EV_CLEAR fflags = NOTE_FORK 10-06-09 9:14:39 AM com.apple.launchd[1] (com.apple.coreservicesd[26]) fork()ed 10-06-09 9:14:39 AM com.apple.launchd[1] (0x100401720.anonymous.lssave) Conceived 10-06-09 9:14:39 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22211]) Created PID 22211 anonymously by PPID 26 10-06-09 9:14:39 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22211]) Looking up per user launchd for UID: 0 10-06-09 9:14:39 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22211]) Per user launchd job found for UID: 505 10-06-09 9:14:39 AM com.apple.launchd[1] System: Looking up service com.apple.system.notification_center 10-06-09 9:14:39 AM com.apple.launchd[1] (com.apple.launchd.peruser.505[143]) Mach service lookup: com.apple.system.notification_center 10-06-09 9:14:39 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22211]) Looking up per user launchd for UID: 0 10-06-09 9:14:39 AM com.apple.launchd[1] (0x100401720.anonymous.lssave[22211]) Per user launchd job found for UID: 505 10-06-09 9:14:39 AM com.apple.launchd[1] System: Looking up service com.apple.system.DirectoryService.libinfo_v1

  Read the article

• Missing audio and problems playing FLV video converted from 720p .mov file with FFMPEG

  - by undefined

  I have some .mov video files recorded from a JVC GC-FM1 HD video camera in 720p mode. I have FFMPEG running on a Linux box that I upload files to and have them encoded into FLV format. The video appears to be encoding ok but there is no audio in the resulting FLV file and when I play it back in Flash Player in a browser or on Adobe Media Player, the video pauses at the start. It appears that Adobe Media Player waits for the progress bar to reach the end of the video before starting the playback - i.e. the video will load, the picture pauses, the progress bar seeks to the end as if the video was playing then when it reaches the end the video picture starts. There is no audio on the video. I am noticing this in the video player I have built with Flash 8 using an FLVPlayback component and attached seekBar. The seek bar will start moving as if the video is playing but the picture remains paused. Here are some outputs from my FFMPEG log and the command I am using to encode the video - my FFMPEG command called from PHP - $cmd = 'ffmpeg -i ' . $sourcelocation.$filename.".".$fileext . ' -ab 96k -b 700k -ar 44100 -s ' . $target['width'] . 'x' . $target['height'] . ' -ac 1 -acodec libfaac ' . $destlocation.$filename.$ext_trans .' 2>&1'; and here is the output from my error log - FFmpeg version UNKNOWN, Copyright (c) 2000-2010 Fabrice Bellard, et al. built on Jan 22 2010 11:31:03 with gcc 4.1.2 20070925 (Red Hat 4.1.2-33) configuration: --prefix=/usr --enable-static --enable-shared --enable-gpl --enable-nonfree --enable-postproc --enable-avfilter --enable-avfilter-lavf --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libvorbis --enable-libx264 libavutil 50. 7. 0 / 50. 7. 0 libavcodec 52.48. 0 / 52.48. 0 libavformat 52.47. 0 / 52.47. 0 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.17. 0 / 1.17. 0 libswscale 0. 9. 0 / 0. 9. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 0 codec frame rate differs from container frame rate: 119.88 (120000/1001) -> 59.94 (60000/1001) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'uploads/video/60974_v1.mov': Metadata: major_brand : qt minor_version : 0 compatible_brands: qt comment : JVC GC-FM1 comment-eng : JVC GC-FM1 Duration: 00:00:30.41, start: 0.000000, bitrate: 4158 kb/s Stream #0.0(eng): Video: h264, yuv420p, 640x480 [PAR 1:1 DAR 4:3], 4017 kb/s, 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 128 kb/s Output #0, rawvideo, to 'uploads/video/60974_v1.jpg': Stream #0.0(eng): Video: mjpeg, yuvj420p, 320x240 [PAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 90k tbn, 59.94 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding [h264 @ 0x8e67930]B picture before any references, skipping [h264 @ 0x8e67930]decode_slice_header error [h264 @ 0x8e67930]no frame! Error while decoding stream #0.0 [h264 @ 0x8e67930]B picture before any references, skipping [h264 @ 0x8e67930]decode_slice_header error [h264 @ 0x8e67930]no frame! Error while decoding stream #0.0 frame= 1 fps= 0 q=3.8 Lsize= 15kB time=0.02 bitrate=7271.4kbits/s dup=482 drop=0 video:15kB audio:0kB global headers:0kB muxing overhead 0.000000% Which are the important errors here - B picture before any references, skipping? decode_slice_header error? no frame? or Seems stream 0 codec frame rate differs from container frame rate: 119.88 (120000/1001) - 59.94 (60000/1001) Any advice welcome, thanks

  Read the article

• Splitting HDMI sound to 2 devices under Windows 7

  - by Jeramy

  Okay, this is a strange set-up and is frustrating me. I have an HDMI signal from my PC being split to my audio receiver and my HDTV. I need to split it to both so that I can choose to either play audio from the HDTV or from the surround sound speakers in the room. The problem that I am having is in Windows 7, the output is listed under "Playback Devices" and is auto-populated with the HDTV, which only has the option for stereo sound. If I unplug the HDTV from the splitter it will populate with my receiver information and let me set it to 5.1 surround, but as soon as I plug the HDTV back in it reverts. I tried reversing the order of the HDMI cables in the splitter and this seemed to work for a short while, then Windows must have polled the devices again or something because it reverted. It will work as long as Windows identifies the reciever, thereby unlocking the 5.1 surround option, otherwise I am stuck with stereo, which it assumes is all the HDTV is capable of. Is there a way to manually override this and set my own options? Or any other solutions?

  Read the article

• Diverting sound output of MCE to SPDIF

  - by Saxtus

  I have an ASUS SupremeFX II audio card (which in fact is an onboard audio riser slot) with the default drivers that are pre-installed by Windows 7 x64 for this card. I am able to manually switch between analog output and SPDIF output by the means of control panel (or external utilities like STADS), a change that affects all applications. The problem is that by doing that every time I am about to launch Windows Media Center, except that it's not that elegant, also makes all other Windows application's sounds to pass through SPDIF too, bypassing analog output completely, blending with what I am watching at Windows Media Center. Is there a way to make SPDIF as the default playback device for Windows Media Center? I know other programs that have a setting like that (foobar2000 for example) working like charm, allowing me to even have different outputs working at the same time (tested with my current card successfully). But when comes to Windows Media Center... it just use what the default playback device is all the time. The only setting that I know of, is under: Settings General Windows Media Center Setup Set Up Your Speakers and what it does is to just change the default playback device for entire system. Please help!

  Read the article

• How to pipe internet radio into a tuner?

  - by JW

  UPDATE: Thanks everyone for the ideas! This was an area I knew very little about but now I can talk with a little more expertise about it. Much appreciated! Visited my dad this weekend and he wants to pipe some internet radio he's found down to a tuner on quite a distance away in the house. He uses computers for only very basic things: e-mail, getting the Post crossword, checking Yahoo!, checking recipes, etc. There's currently one computer in the house (no router). My initial suggestion (without any research whatsoever) was to get a wireless router and a netbook for downstairs near the tuner, but he initially wasn't too keen about having another computer down there. Anyway, is there any computer hardware that could magically pipe the audio output from the computer down to one set of (RCA) audio inputs on the tuner? Wireless isn't necessary but it probably would be easier. Anyway, thanks for your suggestions! UPDATE Thanks everyone! Voted up all of your suggestions now that I have 15 rep. Much appreciated.

  Read the article

• How can I control which sound card Ubuntu uses for playback?

  - by GorillaSandwich

  I am dual-booting Ubuntu 9.04 and Windows XP but am new to Ubuntu. In Windows, I use an M-Audio Audiophile 2496 sound card for recording (because it has RCA input jacks for my mixer), but I don't use it for playback (because my speakers use a 1/8 inch jack); instead, I use the motherboard's built-in sound card. I tried to recreate this arrangement in Ubuntu, but despite selecting the built-in card for all playback under System > Preferences > Sound, I still have inconsistent results. Rhythmbox plays back through the integrated card, but Flash content in the browser and games in the OS send their audio to the Audiophile card. I have seen recommendations to use a program called "Jack" to control this, but I installed it and found it baffling. How can I control which card is used for playback, other than disabling one card (as I discovered how to do and explain below)? Also, is there a GUI for disabling hardware, or is it necessary to edit a configuration file?

  Read the article

• How to send audio data from Java Applet to Rails controller

  - by cooldude

  Hi, I have to send the audio data in byte array obtain by recording from java applet at the client side to rails server at the controller in order to save. So, what encoding parameters at the applet side be used and in what form the audio data be converted like String or byte array so that rails correctly recieve data and then I can save that data at the rails in the file. As currently the audio file made by rails controller is not playing. It is the following ERROR : LAVF_header: av_open_input_stream() failed while playing with the mplayer. Here is the Java Code: package networksocket; import java.util.logging.Level; import java.util.logging.Logger; import javax.swing.JApplet; import java.net.*; import java.io.*; import java.awt.event.*; import java.awt.*; import java.sql.*; import javax.swing.*; import javax.swing.border.*; import java.awt.*; import java.util.Properties; import javax.swing.plaf.basic.BasicSplitPaneUI.BasicHorizontalLayoutManager; import sun.awt.HorizBagLayout; import sun.awt.VerticalBagLayout; import sun.misc.BASE64Encoder; /** * * @author mukand */ public class Urlconnection extends JApplet implements ActionListener { /** * Initialization method that will be called after the applet is loaded * into the browser. */ public BufferedInputStream in; public BufferedOutputStream out; public String line; public FileOutputStream file; public int bytesread; public int toread=1024; byte b[]= new byte[toread]; public String f="FINISH"; public String match; public File fileopen; public JTextArea jTextArea; public Button refreshButton; public HttpURLConnection urlConn; public URL url; OutputStreamWriter wr; BufferedReader rd; @Override public void init() { // TODO start asynchronous download of heavy resources //textField= new TextField("START"); //getContentPane().add(textField); JPanel p = new JPanel(); jTextArea= new JTextArea(1500,1500); p.setLayout(new GridLayout(1,1, 1,1)); p.add(new JLabel("Server Details")); p.add(jTextArea); Container content = getContentPane(); content.setLayout(new GridBagLayout()); // Used to center the panel content.add(p); jTextArea.setLineWrap(true); refreshButton = new java.awt.Button("Refresh"); refreshButton.reshape(287,49,71,23); refreshButton.setFont(new Font("Dialog", Font.PLAIN, 12)); refreshButton.addActionListener(this); add(refreshButton); Properties properties = System.getProperties(); properties.put("http.proxyHost", "netmon.iitb.ac.in"); properties.put("http.proxyPort", "80"); } @Override public void actionPerformed(ActionEvent e) { try { url = new URL("http://localhost:3000/audio/audiorecieve"); urlConn = (HttpURLConnection)url.openConnection(); //String login = "mukandagarwal:rammstein$"; //String encodedLogin = new BASE64Encoder().encodeBuffer(login.getBytes()); //urlConn.setRequestProperty("Proxy-Authorization",login); urlConn.setRequestMethod("POST"); // urlConn.setRequestProperty("Content-Type", //"application/octet-stream"); //urlConn.setRequestProperty("Content-Type","audio/mpeg");//"application/x-www- form-urlencoded"); //urlConn.setRequestProperty("Content-Type","application/x-www- form-urlencoded"); //urlConn.setRequestProperty("Content-Length", "" + // Integer.toString(urlParameters.getBytes().length)); urlConn.setRequestProperty("Content-Language", "UTF-8"); urlConn.setDoOutput(true); urlConn.setDoInput(true); byte bread[]=new byte[2048]; int iread; char c; String data=URLEncoder.encode("key1", "UTF-8")+ "="; //String data="key1="; FileInputStream fileread= new FileInputStream("//home//mukand//Hellion.ogg");//Dogs.mp3");//Desktop//mausam1.mp3"); while((iread=fileread.read(bread))!=-1) { //data+=(new String()); /*for(int i=0;i<iread;i++) { //c=(char)bread[i]; System.out.println(bread[i]); }*/ data+= URLEncoder.encode(new String(bread,iread), "UTF-8");//new String(new String(bread));// // data+=new String(bread,iread); } //urlConn.setRequestProperty("Content-Length",Integer.toString(data.getBytes().length)); System.out.println(data); //data+=URLEncoder.encode("mukand", "UTF-8"); //data += "&" + URLEncoder.encode("key2", "UTF-8") + "=" + URLEncoder.encode("value2", "UTF-8"); //data="key1="; wr = new OutputStreamWriter(urlConn.getOutputStream());//urlConn.getOutputStream(); //if((iread=fileread.read(bread))!=-1) // wr.write(bread,0,iread); wr.write(data); wr.flush(); fileread.close(); jTextArea.append("Send"); // Get the response rd = new BufferedReader(new InputStreamReader(urlConn.getInputStream())); while ((line = rd.readLine()) != null) { jTextArea.append(line); } wr.close(); rd.close(); //jTextArea.append("click"); } catch (MalformedURLException ex) { Logger.getLogger(Urlconnection.class.getName()).log(Level.SEVERE, null, ex); } catch (IOException ex) { Logger.getLogger(Urlconnection.class.getName()).log(Level.SEVERE, null, ex); } } @Override public void start() { } @Override public void stop() { } @Override public void destroy() { } // TODO overwrite start(), stop() and destroy() methods } Here is the Rails controller function for recieving: def audiorecieve puts "///////////////////////////////////////******RECIEVED*******////" puts params[:key1]#+" "+params[:key2] data=params[:key1] #request.env('RAW_POST_DATA') file=File.new("audiodata.ogg", 'w') file.write(data) file.flush file.close puts "////**************DONE***********//////////////////////" end Please reply quickly

  Read the article

• audio onprogress in chrome not working

  - by user351709

  Hi I am having a problem getting onprogress event for the audio tag working on chrome. it seems to work on fire fox. http://www.scottandrew.com/pub/html5audioplayer/ works on chrome but there is no progress bar update. When I copy the code and change the src to a .wav file and run it on fire fox it works perfectly. <style type="text/css"> #content { clear:both; width:60%; } .player_control { float:left; margin-right:5px; height: 20px; } #player { height:22px; } #duration { width:400px; height:15px; border: 2px solid #50b; } #duration_background { width:400px; height:15px; background-color:#ddd; } #duration_bar { width:0px; height:13px; background-color:#bbd; } #loader { width:0px; height:2px; } .style1 { height: 35px; } </style> <script type="text/javascript"> var audio_duration; var audio_player; function pageLoaded() { audio_player = $("#aplayer").get(0); //get the duration audio_duration = audio_player.duration; $('#totalTime').text(formatTimeSeconds(audio_player.duration)); //set the volume } function update(){ //get the duration of the player dur = audio_player.duration; time = audio_player.currentTime; fraction = time/dur; percent = (fraction*100); wrapper = document.getElementById("duration_background"); new_width = wrapper.offsetWidth*fraction; document.getElementById("duration_bar").style.width = new_width + "px"; $('#currentTime').text(formatTimeSeconds(audio_player.currentTime)); $('#totalTime').text(formatTimeSeconds(audio_player.duration)); } function formatTimeSeconds(time) { var minutes = Math.floor(time / 60); var seconds = "0" + (Math.floor(time) - (minutes * 60)).toString(); if (isNaN(minutes) || isNaN(seconds)) { return "0:00"; } var Strseconds = seconds.substr(seconds.length - 2); return minutes + ":" + Strseconds; } function playClicked(element){ //get the state of the player if(audio_player.paused) { audio_player.play(); newdisplay = "||"; }else{ audio_player.pause(); newdisplay = ">"; } $('#totalTime').text(formatTimeSeconds(audio_player.duration)); element.value = newdisplay; } function trackEnded(){ //reset the playControl to 'play' document.getElementById("playControl").value=">"; } function durationClicked(event){ //get the position of the event clientX = event.clientX; left = event.currentTarget.offsetLeft; clickoffset = clientX - left; percent = clickoffset/event.currentTarget.offsetWidth; duration_seek = percent*audio_duration; document.getElementById("aplayer").currentTime=duration_seek; } function Progress(evt){ $('#progress').val(Math.round(evt.loaded / evt.total * 100)); var width = $('#duration_background').css('width') $('#loader').css('width', evt.loaded / evt.total * width.replace("px","")); } function getPosition(name) { var obj = document.getElementById(name); var topValue = 0, leftValue = 0; while (obj) { leftValue += obj.offsetLeft; obj = obj.offsetParent; } finalvalue = leftValue; return finalvalue; } function SetValues() { var xPos = xMousePos; var divPos = getPosition("duration_background"); var divWidth = xPos - divPos; var Totalwidth = $('#duration_background').css('width').replace("px","") audio_player.currentTime = divWidth / Totalwidth * audio_duration; $('#duration_bar').css('width', divWidth); } </script> </head> <script type="text/javascript" src="js/MousePosition.js" ></script> <body onLoad="pageLoaded();"> <table> <tr> <td valign="bottom"><input id="playButton" type="button" onClick="playClicked(this);" value=">"/></td> <td colspan="2" class="style1" valign="bottom"> <div id='player'> <div id="duration" class='player_control' > <div id="duration_background" onClick="SetValues();"> <div id="loader" style="background-color: #00FF00; width: 0px;"></div> <div id="duration_bar" class="duration_bar"></div> </div> </div> </div> </td> </tr> <tr> <td> </td> <td> <span id="currentTime">0:00</span> </td> <td align="right" > <span id="totalTime">0:00</span> </td> </tr> </table> <audio id='aplayer' src='<%=getDownloadLink() %>' type="audio/ogg; codecs=vorbis" onProgress="Progress(event);" onTimeUpdate="update();" onEnded="trackEnded();" > <b>Your browser does not support the <code>audio</code> element. </b> </audio> </body>

  Read the article

• Creating a music catalog in C# and extracting first 30 seconds as soon as the first words are sung

  - by Rad

  I already read a question: Separation of singing voice from music. I don’t need this complex audio processing. I only need some detection mechanism that would detect that there is some voice/vocal playing while the music is playing (or not playing) I need to extract first 30 seconds when a vocalist starts singing along with full band music. See question 2 below. I want to create a music catalog using ASP.NET MVC 2 and Silverlight clients and C#.NET 4.0 programming language that would be front store. On the backend I would also like to create a desktop WPF/Windows application to create the music catalog from already existing music files, most of which have metadata in them ID3v1, ID3v2.3, ID3v2.4, iTunes MP4, WMA, Vorbis Comments and APE Tags etc. I would possibly like to create a web service that would allow catalog contributors to upload a zipped album and trigger metadata extraction of music data and extraction of music segments as described below. I would be happy if I achieve no. 1 below. Let's say I have 1000ths of songs in mp3 (or other formats) grouped in subfolders using some classification (Genre, Artists, Albums, Composers or other groupings). I want to create tables in DB that would organize songs so they can be searched based on different criteria (year, length, above classification or by song title, description etc) like what iTune store allows to their customers. I want to extract metadata from various formats (I will try to get songs in mp3 format, but there may be other popular formats) and allow music Catalog manager person to add missing data from either desktop or web applications. He or other contributors can upload zipped music via an HTML or Silverlight upload or WPF. Can anybody suggest open source libraries, articles, code snippets that can do that in an automatic way using .NET and possibly SQL Server DB? My main questions are these. This is an audio processing challenge. I want to extract 2 segments of music (questions 1 and 2): 1. How to extract a music segment: 1-2 seconds before a vocal starts singing and up to 30 seconds from that point in time and 2. Much more challenging is to find repeating segments (One would usually find or recognize the names of the songs and songs are usually known by these refrains. How would I go about creating a list of songs that go great together like what Genius from iTune does? Is there any characteristics of music that can be used to match songs? The goal is for people quickly scan and recognize songs i.e. associate melody, words with a title/album so they can make intelligent decisions like buying a song, create similar mood songs. Thanks, Rad

  Read the article

• how to save Audiorecorded file to another location? i m trying but i got exception...

  - by rakesh-bhatt99

  NSString recordFile = [NSTemporaryDirectory() stringByAppendingPathComponent: (NSString)inRecordFile]; NSArray *docPaths=NSSearchPathForDirectoriesInDomains(NSDocumentDirectory,NSUserDomainMask,YES); NSString docDir=[[docPaths objectAtIndex:0]stringByAppendingPathComponent: (NSString)inRecordFile]; url = CFURLCreateWithString(kCFAllocatorDefault, (CFStringRef)docPaths, NULL); // create the audio file XThrowIfError(AudioFileCreateWithURL(url, kAudioFileCAFType, &mRecordFormat, kAudioFileFlags_EraseFile, &mRecordFile), "AudioFileCreateWithURL failed"); CFRelease(url);

  Read the article

• Android RTSP coding problem

  - by NetApex

  I have Googled my butt off trying to find where if there is a surefire way to make rtsp work. I have a radio station that I listen to that streams via rtsp. Of course by default Android doesn't want to play it. If I pop the URL into yourmuze.fm and create a station there it lets me stream it to my phone. After checking how it works I come to find that it streams to the phone via rtsp! So obviously there is something amiss. What makes one stream work and one not? This is the stream I am attempting : rtsp://wms2.christiannetcast.com/yes-fm It is an audio stream so I would be thrilled with most peoples problem of "it only does audio and not video." When yourmuze.fm streams, DDMS states it brings up MovieView to play the audio if that helps at all.

  Read the article

• Record AVAudioPlayer output using AVAudioRecorder

  - by Kieran

  In my app the user plays a sound by pressing a button. There are several buttons which can be played simultaneously. The sounds are played using AVAudioPlayer instances. I want to record the output of these instances using AVAudioRecorder. I have set it all up and a file is created and records but when I play it back it does not play any sound. It is just a silent file the length of the recording. Does anyone know if there is a setting I am missing with AVAudioPlayer or AVAudioRecorder? Thanks

  Read the article

• Embedding wav files in AS3 Flash/Flex project?

  - by aaaidan

  The Flash IDE is capable of embedding many types of uncompressed sound files, including wav, and offers optional compression when publishing. However, the [Embed] tag, only seems to allow embedding of mp3 files. Is it truly impossible to embed an uncompressed wav file, or am I missing some magic, undocumented mimeType? I was hoping for something like: [Embed source="../../audio/wibble.wav" mimeType="audio/wav"] ...but I get no transcoder registered for mimeType 'audio/wav' It's possible to embed wav or other format as an octet-stream and parse at runtime, but that's pretty heavy handed I think. I'm surprised that even though the Flash IDE can embed uncompressed sound data, [Embed] cannot, given that the swf spec can contain uncompressed sound data. Any takers?

  Read the article

• Streaming server required with JW Player?

  - by Aaron

  Currently, a site I developed plays mp3 files directly in JW Player using the file attribute and public URLs to the mp3 file. This is now an issue with the client for legal reasons, and they now need to stream the audio files so that the users can't open up their cache and nab the files directly after downloading. The JW player site has a bunch of examples for streaming video, but nothing for audio. Is it possible to stream audio files with JW player, and do we have to pay a lot of money for a streaming provider? Is it possible to do on the local php server?

  Read the article

• How to do a sample rate conversion in Windows (and OSX)

  - by Paperflyer

  I am about to write an audio file converter for my side job at the university. As part of this I would need sample rate conversion. However, my professor said that it would be pretty hard to write a sample rate converter that was both of good quality and fast. On my research on the subject, I found some functions in the OSX CoreAudio-framework, that could do a sample rate conversion (AudioConverter.h). After all, an OS has to have some facilities to do that for its own audio stack. Do you know a similar method for C/C++ and Windows, that are either part of the OS or open source? I am pretty sure that this function exists within DirectX Audio (XAudio2?), but I seem to be unable to find a reference to it in the MSDN library.

  Read the article

• Can anyone give me a sample DSP script in C/C++

  - by Andrew

  Im working on a (Audio) DSP project and just wondering if there are any sample (Open source) DSP example that are written in c or c++, for my MSP430 Chip. I just want something as a guideline so i can program my own script using the ACD and DCA on my board for sampling. http://focus.ti.com/docs/toolsw/folders/print/msp-exp430f5438.html Thats my board, MSP430F5438 Experimenter Board, from what i herd it can run dsp script via the USB connection with the computer. Im using CCS ( From TI, code composer studio) and Octave/Matlab. Just any DSP example scripts or sites that will help me create my own would be appreciated. What im tying to do, Partial audio (sampled) track -- Nyquist rate sampling -- over- and undersampling -- reconstruction of the audio track.

  Read the article

< Previous Page | 45 46 47 48 49 50 51 52 53 54 55 56  | Next Page >