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• Concatenation of a 2 second silence audio with a normal audio not working

  - by user1665130

  I have a code for concatenation of files using ffmpeg.Here silence.wav is a mute audio file with 2 seconds length. I need to prepend this mut audio file to REC00096_Jun-06-2014 16.47.28.wav. I tried the folowing code. ffmpeg -i D:\vishnu\silence.wav -i D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav \-filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' \-map '[out]' output.wav Following is the error i am getting. D:\vishnu>ffmpeg -i silence.wav -i "D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav" -filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' -map '[out]' outp ut.wav ffmpeg version N-59036-g5d8e4f6 Copyright (c) 2000-2013 the FFmpeg developers built on Dec 12 2013 22:01:01 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 58.100 / 52. 58.100 libavcodec 55. 45.101 / 55. 45.101 libavformat 55. 22.100 / 55. 22.100 libavdevice 55. 5.102 / 55. 5.102 libavfilter 3. 92.100 / 3. 92.100 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Input #0, wav, from 'silence.wav': Metadata: encoder : Lavf55.22.100 Duration: 00:00:02.02, bitrate: 4234 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 5.1, s16, 4 233 kb/s Guessed Channel Layout for Input Stream #1.0 : mono Input #1, wav, from 'D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav': Duration: 00:00:08.04, bitrate: 384 kb/s Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 24000 Hz, mono, s16, 384 kb/s [wav @ 036f5e40] Invalid stream specifier: '[out]'. Last message repeated 1 times Stream map ''[out]'' matches no streams. D:\vishnu>

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• iPhone Web Application + zoom in zoom out issue

  - by pratik

  Hello, I am developing a web application for iPhone safari browser. There is one strange scenario: A web page is loaded. User can zoom out that web page to see the contents in zoomed out manner. There is one button in the web page. Click of this button will hide one of the 'div' in same page and another 'div' of that same page will be visible. Problem, is that suppose the user has zoomed out the web page to see the contacts in zoomed out manner. Now, user clicks the button (the page is zoomed out only), the next 'div' which is get visible is also disaplayed in zoomed out manner. What is want to do is that even if user has zoomed out the page and he clicks on the button, the next 'div' which get visible should have the display as zoomed in only.

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• HTML5 <audio> Safari live broadcast vs not

  - by Peter Parente

  I'm attempting to embed an HTML5 audio element pointing to MP3 or OGG data served by a PHP file . When I view the page in Safari, the controls appear, but the UI says "Live Broadcast." When I click play, the audio starts as expected. Once it ends, however, I can't start it playing again by clicking play. Even using the JS API on the audio element and setting currentTime to 0 fails with an index error exception. I suspected the headers from the PHP script were the problem, particularly missing a content length. But that's not the case. The response headers include a proper Content- Length to indicate the audio has finite size. Furthermore, everything works as expected in Firefox 3.5+. I can click play on the audio element multiple times to hear the sound replay. If I remove the PHP script from the equation and serve up a static copy of the MP3 file, everything works fine in Safari. Does this mean Safari is treating audio src URLs with query parameters differently than URLs that don't have them? Anyone have any luck getting this to work? My simple example page is: <!DOCTYPE html> <html> <head></head> <body> <audio controls autobuffer> <source src="say.php?text=this%20is%20a%20test&format=.ogg" /> <source src="say.php?text=this%20is%20a%20test&format=.mp3" /> </audio> </body> </html> HTTP Headers from PHP script: HTTP/1.x 200 OK Date: Sun, 03 Jan 2010 15:39:34 GMT Server: Apache X-Powered-By: PHP/5.2.10 Content-Length: 8993 Keep-Alive: timeout=2, max=98 Connection: Keep-Alive Content-Type: audio/mpeg HTTP Headers from direct file access: HTTP/1.x 200 OK Date: Sun, 03 Jan 2010 20:06:59 GMT Server: Apache Last-Modified: Sun, 03 Jan 2010 03:20:02 GMT Etag: "a404b-c3f-47c3a14937c80" Accept-Ranges: bytes Content-Length: 8993 Keep-Alive: timeout=2, max=100 Connection: Keep-Alive Content-Type: audio/mpeg I tried hard-coding the Accept-Ranges header into the script too, but no luck.

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• iphone uiscrollview and uiimageview - setting initial zoom

  - by Brodie4598

  I use the following code to load an image into an scroll view. The image always loads at 100% zoom. Is there a way to set it to load to another zoom level, say .37? UIImageView *tempImage = [[UIImageView alloc]initWithImage:[UIImage imageWithData:data]]; self.imageView = tempImage; scrollView.contentSize = CGSizeMake(imageView.frame.size.width , imageView.frame.size.height); scrollView.maximumZoomScale = 1; scrollView.minimumZoomScale = .37; scrollView.clipsToBounds = YES; scrollView.delegate = self; [scrollView addSubview:imageView];

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• CSS Margin/Padding Issue on Browser Zoom

  - by korymath

  I am having difficulty getting my page to show up correctly at various browser zoom settings. http://andstones.ca/newsite The content is aligned if the user is zoomed out on the page, but once they zoom too far out and refresh the page, or if they are too zoomed in and the page loads, then the content pane is shifted. This does not seem to be an issue in firefox, but it is a very serious issue in chrome/safari/iexplore and on the iPad. What can I do to ensure that the main content loads in the correct position regardless of user browser settings? Thanks , Kory

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• Transform/redraw view after pinch zoom on x-axis

  - by Jonathan

  My setup. UIScrollView (scrollView) - UIView (contentView) - UIView (subView) I have managed to do so I can zoom contentView only on the x-axis. The problem is that subView is containing a graph. When contentView is zoomed and transformed the graph gets unsharp and distorded since the scaling only effect the x-axis. What I need help with is to somehow redraw the contentView after I'm done with the zooming without distorting the graph. Is it possible to transform the contentView so that the graph stays sharp even if you only zoom in the x-axis or redraw the content in the zoomed/stretched view somehow? I have tried the solution in this thread link but haven't succeeded to get it running.

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• I want pan and zoom functionality (like a MapView) on my own map image

  - by fantius

  How can I get my own image to be able to pan and zoom (like a MapView)?

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• General zoom algorithm for drawing program

  - by Steven Sproat

  My GUI toolkit, wxPython provides some methods for implementing a user zoom factor, however the quality isn't too good. I'm looking for ideas on how to create a zooming feature, which I know is complicated. I have a bitmap representing my canvas which is drawn to. This is displayed inside a scrolled window. Problems I forsee: - performance when zoomed in and panning around the canvas - difficulties with "real" coordinates and zoomed in coordinates - image quality not degrading with the zoom Using wxPython's SetUserScale() on its device contexts presents image quality like this - this is with a 1px line, at 30% zoomed in. I'm just wondering the general steps I'll need to take and the challenges I'll encounter. Thanks for any suggestions

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• Windows Audio Issue

  - by Nikki

  This one is driving me nuts. Hoping someone can shed some light. I'm running windows 7 using onboard audio. It's been fine for over 2 years but lately there's a problem every time I play audio. I hear a small soft burst of static and the volume turns itself down from 50% to 23%. Once at 23%, it plays fine. No related events logged in viewer. No reported problems with the device. Different headphones, same problem. I played around with audio settings for hours but the problem persists. EDIT: ok more info: Motherboard: ECS G31T-M LGA775 System info displays this: Name High Definition Audio Device Manufacturer Microsoft Status OK PNP Device ID HDAUDIO\FUNC_01&VEN_1106&DEV_E721&SUBSYS_10192683&REV_1001\4&3D4E739&0&0001 Driver c:\windows\system32\drivers\hdaudio.sys (6.1.7600.16385, 297.00 KB (304,128 bytes), 14/07/2009 9:51 AM) I'll keep adding info as I find it. The question I want resolved is; Is it faulty hardware? If so, I can buy a sound card. I can't imagine software is responsible since I haven't installed anything new for weeks. Virus scans are clear as well. The static burst is irritating to say the least. Tried 2 different headphones and separate speakers. Same problem. I know it's not an easy problem but I was hoping someone had encountered the same thing.

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• Simulating audio playback on headless linux server

  - by afro

  Hi people, We have a headless linux server (Debian 5) we use for runnin integration tests of our web-page code. Among these tests are ones implemented using Selenium, which practically simulates a user browsing our pages and clicking on things. One of these tests is failing now, because it involves starting a flash-based audio player and checking to see whether the progress bar gets displayed properly. The reason this test fails is that there is no way to play the audio, and no sound card on the machine, which has simple webserver hardware. So, my question would be: Is there a simple way of giving a program the impression that its audio output is being processed, and playback is taking place? I don't have to record the playback, or redirect it or anything like that, just a dummy soundcard, like the dummy X-server we aer using, which actually does not need to display stuff. I have tried using JACK, but it's too complicated, and the documentation does not even answer this very simple question. I also installed alsa on the server; it 'pretends' to run, but when a program tries to play audio, just spews error and debug information having to do with the non-existence of a soundcard. It would be really awesome if one of you has a simple answer to this question. Cheers, Ulas

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• Configure audio on HP ENVY 4 ultrabook

  - by phodu_insaan

  I want to configure audio for ubuntu 12.04 on my laptop. Currently the audio just does not play. If i try and plug in headphones then somewhere midway to being fully plugged in the audio plays on the headphones, I plug in further and the sound disappears. How do I get this to work? lspci | grep audio Audio device: Intel Corporation Panther Point High Definition Audio Controller My laptop is one the beats edition HP laptops, and the driver for win7 was an IDT HD audio driver. --EDIT-- The output for cat /proc/asound/card0/codec* | grep Codec is Codec: IDT 92HD91BXX Codec: Intel PantherPoint HDMI I need to get both the IDT card to work and the HDMI card to work with my TV. --EDIT-- --EDIT 2-- I have added blacklist snd-usb-audio to the end of the file /etc/modprobe.d/alsa-base.conf Now the sound plays from my laptop speakers only when I plug in a headphones/external speaker. Otherwise no sound. :( Please help getting everything working as it should. --EDIT 2-- Thanks

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• About AMR audio file playing issue on different devices

  - by user352537

  I have got a quite strange problem here. I am developing an IM software and need to play audio files recorded by another client on Android. The same audio file I've got can be played with AVAudioPlayer on 3GS(IOS 4.2.1) device and simulator 4.2. But when I tried by play it on iPhone4(iOS 4.3.3), the function "play" always return NO. I also tried with two iPhone devices, the audio files recorded by iPhone client can be played on both 3GS and iPhone4. So I asked the Android developers about the record parameters they've used. They said that the "AudioEncoder" used by them was "DEFAULT". There are also some other parameters as following: **private AudioEncoder() {} public static final int DEFAULT = 0; /** AMR (Narrowband) audio codec */ public static final int AMR_NB = 1; /** @hide AMR (Wideband) audio codec */ public static final int AMR_WB = 2; /** @hide AAC audio codec */ public static final int AAC = 3; /** @hide enhanced AAC audio codec */ public static final int AAC_PLUS = 4; /** @hide enhanced AAC plus audio codec */ public static final int EAAC_PLUS = 5;** Does anybody know what's the matter?

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• Writing an audio player in C#

  - by Malki

  Hi, I have a pretty cool idea for a very special media player. I like to think about this project as a mini-startup, since I don't yet know if my idea is practical. Anyways, before implementing my idea, I first need to be able to implement a simple audio player. My preferred language for this project is C#, simply because it's so easy to use, but any other object oriented language would be fine too I guess. I started out with no knowledge whatsoever about audio. My main goals right now are: Being able to play audio files - as many formats as possible (sort of a VLC type player, but only audio for now). Being able to analyze audio files - as in, reading frequency, amplitude, volume, and other information about the audio. I think maybe a good idea here is to be able to analyze one file format (PCM?), and then temporarily converting any file I want to analyze to that format. This is in order to later implement a mechanism that compares songs and identifies similar songs to recommend to the user (this feature isn't part of my idea, but I figured since it exists in many players nowadays, I need to have it too if I want be able to compete with them). BTW - I currently don't have any knowledge about audio/wavelengths/frequencies and such, so I'd appreciate it if someone could point me in the right direction about this analyzation feature. Maybe in the future I'd expand to playing video files as well, but for now I'm concentrating on audio. After searching the Internet for a while, I've come across LAME. Problem is, it's not C#, and I'm not sure how to use it. I know there is something called "Interoperability", that is supposed to let me work with native DLL files through C#. Any information about that would be helpful as well. Any help would be much appreciated. Thanks, Malki :)

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• iPhone SDK Smaller CAF files: lower recording quality with Audio Queues?

  - by Nick

  In my iPhone app I have voice recording functionality the utilizes Audio Queue voice recording functions of the SDK. I'm saving directly to CAF format and using the following settings for the AudioStreamBasicDescription reference: audioFormat.mFormatID = kAudioFormatLinearPCM; I can see that there are other format ids I could use like: kAudioFormatLinearPCM kAudioFormatAppleLossless kAudioFormatAppleIMA4 kAudioFormatiLBC kAudioFormatULaw kAudioFormatALaw My knowledge of sound formats is very limited so my question is... which of these should I use to create the lowest compressed audio recording files? Plus, are there other settings I should apply to lower the quality and filesize even further?

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• How to program a real-time accurate audio sequencer on the iphone?

  - by Walchy

  Hi... I want to program a simple audio sequencer on the iphone but I can't get accurate timing. The last days I tried all possible audio techniques on the iphone, starting from AudioServicesPlaySystemSound and AVAudioPlayer and OpenAL to AudioQueues. In my last attempt I tried the CocosDenshion sound engine which uses openAL and allows to load sounds into multiple buffers and then play them whenever needed. Here is the basic code: init: int channelGroups[1]; channelGroups[0] = 8; soundEngine = [[CDSoundEngine alloc] init:channelGroups channelGroupTotal:1]; int i=0; for(NSString *soundName in [NSArray arrayWithObjects:@"base1", @"snare1", @"hihat1", @"dit", @"snare", nil]) { [soundEngine loadBuffer:i fileName:soundName fileType:@"wav"]; i++; } [NSTimer scheduledTimerWithTimeInterval:0.14 target:self selector:@selector(drumLoop:) userInfo:nil repeats:YES]; In the initialisation I create the sound engine, load some sounds to different buffers and then establish the sequencer loop with NSTimer. audio loop: - (void)drumLoop:(NSTimer *)timer { for(int track=0; track<4; track++) { unsigned char note=pattern[track][step]; if(note) [soundEngine playSound:note-1 channelGroupId:0 pitch:1.0f pan:.5 gain:1.0 loop:NO]; } if(++step>=16) step=0; } Thats it and it works as it should BUT the timing is shaky and instable. As soon as something else happens (i.g. drawing in a view) it goes out of sync. As I understand the sound engine and openAL the buffers are loaded (in the init code) and then are ready to start immediately with alSourcePlay(source); - so the problem may be with NSTimer? Now there are dozens of sound sequencer apps in the appstore and they have accurate timing. I.g. "idrum" has a perfect stable beat even in 180 bpm when zooming and drawing is done. So there must be a solution. Does anybody has any idea? Thanks for any help in advance! Best regards, Walchy

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• Best practice for C++ audio capture API under Linux?

  - by braddock

  I need to create a C++ application with a simple audio recording from microphone functionality. I can't say that there aren't enough audio APIs to do this! Pulse, ALSA, /dev/dsp, OpenAL, etc. My question is what is the current "Best practice" API? Pulse seems supported by most modern distros, but seems almost devoid of documentation. Will OpenAL be supported across different distros, or is it too obscure? Have I missed any? Is there not a simple answer? thanks!

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• Why does my audio keep reverting to stereo?

  - by GraemeF

  I have an ASUS P7P55D LE motherboard with onboard sound and I am running Windows 7 64-bit, and I am using the SPDIF output from my motherboard to my receiver. On my receiver I can see which audio channels are in use (i.e. stereo or 5.1) In the audio interface properties I can test DTS Audio and Dolby Digital outputs and both work fine, but when I try to play a game with 5.1 sound (I've tried Left 4 Dead and Dragon Age Origins) it reverts to stereo. I was getting this behaviour with the default Microsoft drivers so I installed the latest from the ASUS website but there is no difference. I notice that in the Advanced tab the Default Format only allows me to choose from various 2 channel formats so maybe that is something to do with it? How can I get 5.1 output all of the time?

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• Windows 7 Audio problem..

  - by Marcx

  hi, I've a dell studio 1555 bought on september with Windows 7 64bit Professional on it. Audio devide works proprerly, while listening/watching audio/video contents (from disk or internet) Using VoIP Program Ventrilo Audio from others people works good, and I hear their voices clearly... Using VoIP Program such as Teamspeak 2/3, MSN, Skype I hear a really disturbed voice, and it's impossible to comprehend somethings... Anyway I can say that the first week that I've bought and installed Win7 Skype worked good, this problem happened after I installed Ventrilo... I tried to remove it (ventrilo) but nothing happened:( Please help me :D Thanks Marcx

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• Integrated Graphics and Audio as Media Center?

  - by Will

  I'm considering setting up a PC as a Media Center. Mainly to watch movies (ideally HD quality) and listening to music, but also to perform tasks like e-mail, web browsing, ... I quite like the looks and the price of this barebone: http://www.asus.de/Barebone_PC/S_Series_7L/S2P8H61E However it comes with integrated graphics and audio and only has one free PCI-Express slot. Which would mean in the worst case, where both integrated graphics and audio turn out to be insufficient, I could only upgrade one. So is integrated graphics and audio sufficient for a media center solution? Cheers, Will

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• Audio broken on clean install of ThinkPad T60

  - by Ben Alpert

  I just reinstalled a clean version of XP SP3 on a ThinkPad T60. After downloading all of the missing drivers from the Lenovo site, everything seems to work except for the audio. I installed the audio driver from http://www-307.ibm.com/pc/support/site.wss/MIGR-62928.html (version 5.10.1.4326) so SoundMAX shows up in the control panel and everything in the Sound preferences looks okay, but no sound comes out of the speakers when I try to play music or hear the interface sound effects. There's also no sound if I plug in headphones and listen. Some forums suggested installing hotfix Q888111 but it's already included in SP3 so I can't install it. I also tried reinstalling the audio driver mentioned above in case something went wrong the first time, but it still doesn't work. Does anyone know how to make the sound work?

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• Enable multiple audio output on Windows 7

  - by patrick

  For Windows 7, 64 bit: I have a digital SPDIF output to my stereo, which controls speakers in other rooms. I also have a set of speakers connected to the regular audio jack at the computer. This allows me to send music to the kitchen while my child plays games on the computer. Works great. Except when I'm playing games and still want to listen to music. ;-D I know I can manually switch WMP to play through the speakers instead of SPDIF, but I was wondering if there's any way to enable simultaneous audio out in Windows 7? Virtual Audio Card is a non-starter because I'm running 64 bits and the VAC driver isn't signed.

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• HDMI Audio drops out when display enters powersave

  - by Jared Tritsch

  I have a Windows 8 machine with an AMD APU attached to my Home Theater system through HDMI (HDMI routes through a Home Theater AMP, then into the TV). Here's my problem, Whenever the display is interrupted, usually by the TV being turned off or into powersave mode, the audio device lists as "Disconnected" in windows audio devices and I cant get it to re-recognize that the HDMI audio is, in fact, plugged in. The only solution I have found so far is to restart the machine, which will then recognize the device without any problems, until the next time the TV turns off and the problem once again resurfaces. Has anyone else seen this phenomenon? I have no idea if its the GPU, the HDMI interface, the AMP, or even the TV itself, as there really isn't much a way to tell...

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• Logitech bluetooth audio receiver quality

  - by lietus

  I bought a Logitech Bluetooth Audio receiver, and have run into a problem: When playing audio, it contains short hisses/clicks on higher tones or vocal+heavy instrumental music (Sound quality appears to be unclean). I'm not sure what could be the reason, but this only happens when I play from certain devices. My laptops (a Dell XPS Win7, and an Asus Eee seashell Win7) and my phone (Samsung Player 5) both have the problem. However, when I tried using a Samsung S2 phone, the audio was crystal clear. Seems that this could be something with Bluetooth transiting device.. Has anyone encountered a similar problem?

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• Ubuntu 11.04: No Audio Output

  - by Jason George

  I installed a fresh copy of Ubuntu 11.04 earlier this week and I'm having trouble getting my audio online. It was working fine in 10.04 and all the resources I can find on troubleshooting seem to be fairly dated so I'm not sure if they apply. CMI8788 [Oxygen HD Audio] Analog Stereo Duplex Playing a WMA file shows 0.00db output when I mouse over the sound controller in the status bar. Obviously, no output from my speakers. I tried adjusting the profile, thinking I might have the wrong one. That seems to have made things worse. Where mouse over originally said something along the lines of "Oxygen HD Audio," it now reads "Dummy Output." Selecting "Test Speakers" in sound preferences crashes the dialog. Any pointers would be great.

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• FFmpeg add multiple audio files to video at specific points

  - by Arran

  I have two audio files, each about 3 minutes long. I want to take the first 10 seconds of each file and add them each to a video file at specific points - 0 seconds and 10 seconds. So the resulting video should be 20 seconds long. I've got this far: ffmpeg -i video.mov -ss 0 -t 20 -itsoffset 0 -i audio1.mp3 -itsoffset 10 -i audio2.mp3 -acodec copy -vcodec copy out.mov ...but the resulting video has 20 seconds of the first audio file only, the second audio file doesn't start at 10 seconds like it should. Any help would be appreciated, thanks!

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