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  • Bluetooth Audio and SoftPhone Audio Input/Output

    - by o7th Web Design
    I have a Voip Softphone software that I would like to start using on my Ubuntu 14.04 box. Here's the thing. My system sound right now goes through my HDMI to my speaker system so I can play music all day ;-) I have a bluetooth headset connected to the machine as well. What I am wondering is if there is a way to: Auto-mute the music when a call comes in Auto-switch the sound devices when a call comes in, from my hdmi sound device, to my headset Auto-switch back when the call ends, and auto-un-mute the music Or even just an auto-switch to the headset? I can always pause the music ;)

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  • Mass audio encoder

    - by bessman
    I have a few thousand FLAC files which I would like to transcode to OGG Vorbis, but I can't find any suitable tools for the job. To name a few I have tried so far and why they are unsuitable: oggenc is single-threaded and would require me to automate it myself, mencoder requires the input to also contain video, and abcde assumes the input is a CD. The ideal tool should be multi-threaded, and support inputing multiple files located in different directories simultaneously. CLI or GUI makes no matter. Does such a tool exist?

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  • iOS 5 Audio Alarms Don't Sound Without kAudioSessionProperty_OverrideCategoryMixWithOthers On

    - by coneybeare
    I have an audio app that is having some problems with the way iOS 5 has changed audio behaviors. When my app's audio is playing (AVAudioSessionCategoryPlayback), and a Clock.app alarm or timer is fired from the OS, the UIAlertView notification pops up, but without the audio alert. My application sound ducks fine to get out of the way of the audio alert, but the alarm app's audio alert does not sound. Naturally, tons of support requests poured in over the iOS 5 change. I have solved this temporarily by setting kAudioSessionProperty_OverrideCategoryMixWithOthers which lets the alarm audio come through, but there are a few very undesirable side-effects when doing this: Other app's audio can play with/over mine. The remote control events are not routed to my app, but to iPod.app. None of the above drawbacks are acceptable for my app's requirements. I have been hacking away at this for some time now but haven't been able to crack it. How can I setup my audio such that: My app's audio still uses the AVAudioSessionCategoryPlayback category for background audio. The Clock.app alarms still have their audio alerts make sound The app still responds to remote control notifications

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  • pitchbend (varispeed) audio with iPhone SDK's AudioUnit

    - by fetzig
    hi, I'm trying to manipulate the speed (and pitch) of a sound while playing. so i played around with iphone sdk's AudioUnit. downloaded iPhoneMultichannelMixerTest and tried to add an AUComponent to the graph (in this case a formatconverter). but i get (pretty soon) following error when building: #import <AudioToolbox/AudioToolbox.h> #import <AudioUnit/AudioUnit.h> ... AUComponentDescription varispeed_desc(kAudioUnitType_FormatConverter, kAudioUnitSubType_Varispeed, kAudioUnitManufacturer_Apple); ^^ error: 'kAudioUnitSubType_Varispeed' was not declared in this scope. any ideas why? the documentation on this topic doesn't help me at all (just api doc isn't very helpful when having no clue about the concept behind). there are no examples on how to wire these effects together and manipulating there properties...so maybe i'm totally wrong, anyway any hint is great. thx for help.

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  • Can Flash player play .m3u audio files from a remote location

    - by undefined
    Is it possible to play a .m3u file streamed from a remote location through Flash Player in a browser? I have a player that loads and plays .mp3 files but also want to be able to play .m3u files. I have looked at the as3plsreader on google code but I think this is only for AIR and desktop files. anyone tried this or know where I should start looking for an answer? If I wasnt to use flash, what other ways could I get remote m3u files to play in a browser?

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  • How can I get it the Free Music Archive audio player or is there a better alternative?

    - by Dennis Hodapp
    I'm looking at free streaming audio players for web browsers that I can use in a project. I really like the audio player used on http://freemusicarchive.org/. Are they using an open source audio player and can I get a hold of it? Or is it closed source? Also if there are any open-source audio players that anybody knows about I'd love to know about them (preferable to have one with no flash). Last thing...is HTML5 going to be able to replace audio streaming players?

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  • Windows not remembering default audio device?

    - by Lynda
    I prefer the audio output on my computer to use the standard audio jack output due to volume issues. But I am using a monitor with HDMI. I have chosen to set the default audio device to be "Speakers" But every time I reboot the default audio device is the HDMI Output again. I am running Windows 7 64bit. Why does it not remember the default device? (I do shutdown and boot up properly without errors.)

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  • No audio device detected

    - by Paul
    My computer has no audio. It says NO AUDIO DEVICE. I already installed Realtek AC97 and Realtek High Definition Audio Driver and I also pasted stream.dll to the Windows and system32 folders and I restarted my computer but it still says NO AUDIO DEVICE. Please help me. Thanks

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  • ASUS N45SF - play subwoofer with audio connected

    - by Jaroslav Bucko
    I have notebook ASUS N45SF. It comes with dedicated subwoofer, which is connected to separate audio jack. When I connect any audio device to audio jack, internal speakers remain silent, but subwoofer too. I want to let subwoofer play even with audio device connected to jack. Are there any drivers or settings in OS, which would eneble this behaviour? I have Win7/Ubuntu dualboot so OS doesnt matter. Thanks

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  • PHP & IIS 6 Encoding problem

    - by Alexander
    The server is running Windows 2003 with IIS 6.0.3790.1830 x86 (iis.dll). My database server is Microsoft SQL Server 2000. My PHP version is 5.3. The original application is hosted on appserv1 and it's database is on dbserv1. It's working fine, everything is tuned up, running great. It was needed to place the same application (different modules) on another server, for other uses, so I copied the database on dbserv2, configured appserv2 to host the application, so I achieved 2 almost identical copies. Both dbserv1 and dbserv2 use the same encoding, both appserv1 and appserv2 are on IIS6 with the same PHP configurations. I also tried my best to have the same settings in the IIS. I also made sure that I pass the encoding information both in the HTTP headers and in the meta tags with http-equiv. Both applications use utf-8. The Problem is that the copy of the application doesn't display the non-ASCII characters normally in the browser, even if the browser detects correctly the UTF-8 encoding of the page. First I thought it was a database issue, given the fact that MSSQL 2000 doesn't support UTF-8, and instead it uses UCS-2, but when I redirected the application on appserv2 to work with the database on dbserv1, it had the same encoding problems. This is why I am asking in what way I can make it work. thank you for reading.

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  • Ivar definitions show 'long' type encoding as 'long long' type encoding

    - by Frank C.
    I've found what I think may be a bug with Ivar and Objective-C runtime. I'm using XCode 3.2.1 and associated libraries, developing a 64 bit app on X86_64 (MacBook Pro). Where I would expect the type encoding for the following "longVal" to be 'l', the Ivar encoding is showing a 'q' (which is a 'long long'). Anyone else seeing this? Simplified code and output follows: Code: #import <Foundation/Foundation.h> #import <objc/runtime.h> @interface Bug : NSObject { long longVal; long long longerVal; } @property (nonatomic,assign) long longVal; @property (nonatomic,assign) long long longerVal; @end @implementation Bug @synthesize longVal,longerVal; @end int main (int argc, const char * argv[]) { NSAutoreleasePool * pool = [[NSAutoreleasePool alloc] init]; unsigned int ivarCount=0; Ivar *ivars= class_copyIvarList([Bug class], &ivarCount); for(unsigned int x=0;x<ivarCount;x++) { NSLog(@"Name [%@] encoding [%@]", [NSString stringWithCString:ivar_getName(ivars[x]) encoding:NSUTF8StringEncoding], [NSString stringWithCString:ivar_getTypeEncoding(ivars[x]) encoding:NSUTF8StringEncoding]); } [pool drain]; return 0; } And here is output from debug console: This GDB was configured as "x86_64-apple-darwin".tty /dev/ttys000 Loading program into debugger… sharedlibrary apply-load-rules all Program loaded. run [Switching to process 6048] Running… 2010-03-17 22:16:29.138 ivarbug[6048:a0f] Name [longVal] encoding [q] 2010-03-17 22:16:29.146 ivarbug[6048:a0f] Name [longerVal] encoding [q] (gdb) continue Not a pretty picture! -- Frank

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  • what is the best mid/high-end class audio/music creation audio sound card?

    - by Chris
    Hello, I have a computershop myself, and I repair computers. But one of the things I really don't know (yet) is the performace od audio cards for music creation with midi. I have searched and searched and came up with some good reviews, but after browsing for a couple of hours I could't see the trees trough the forrest :-D (it's a dutch expression) At one moment I thought the M-Audio - Delta 1010LT would be a good PCIe card, later on I read that this card was released years ago. (but that could be false information) Also any personal expierence would be great, but not necessairy. I have searched a few cards, and I hope someone can help me make a choice for a friend of mine. He's buget is between $100 and $350 I know there are audio cards from $ 500 - $1850,- this is just too expensive. The following specs are crucial: ASIO Midi Mic in minimal 5.1, 7.1 recommended it's not for airplay, but just to compose music at home. using Ableton and midi keyboard. 1. M-Audio - Delta 1010LT: 8 x 8 analog I/O 2 mic preamps or line inputs S/PDIF digital I/O (coaxial) with 2-channel PCM SCMS copy protection control digital I/O supports surround-encoded AC-3 and DTS pass-through 1 x 1 MIDI I/O directly drive up to 7.1 surround (bass management software included) software controlled 36-bit internal DSP digital mixing/routing +4dbu/-10dBV operation individually switched in software word clock I/O for sample accurate device synchronization 2. RME HDSP 9632: * Stereo Analog Ein- und Ausgang, symmetrisch*, 24-Bit/192kHz, > 110 dB SNR * Optionale Erweiterungsboards mit je 4 symmetrischen Ein- und Ausgängen * Alle analogen I/Os voll 192 kHz-fähig, also keine Reduzierung der Kanalzahl * 1 x ADAT Digital In/Out, 96 kHz-fähig (S/MUX) * 1 x SPDIF Digital In/Out, 192 kHz-fähig * 1 x Breakout Kabel für koaxialen SPDIF-Betrieb* * Also bis zu 16 Ein-und Ausgänge gleichzeitig nutzbar! * 1 x Stereo Kopfhörerausgang, parallel zum analogen Ausgang, aber eigene Pegelanpassung * 1 x MIDI I/O für 16 Kanäle Hi-Speed MIDI über Breakout Kabel * DIGICheck, RMEs einzigartiges Meter- und Analysetool mit Spectral Analyser, Professionelle Level Meter 2/8/16-Kanalig, Vector Audio Scope und diversen weiteren Analysefunktionen * HDSP Meter Bridge: Frei skalierbare Levelmeter mit Peak- und RMS Berechnung in Hardware * TotalMix: 512-Kanal Mischer mit 40 Bit interner Auflösung 3. EMU 1212M (1212 M) PCIe: * Top kwaliteit convertors 24-bit/192kHz convertors. * Hardware gestuurde effecten. * DSP zero-latency hardware mixen en monitoring. * Analoge en digitale I/O plus MIDI. * EMU Production Tools Software Bundle - Cakewalk SONAR , Steinberg Cubase LE, Ableton Live E-MU Edition **EMU 1212M PCI-e inputs/outputs:** * 2 balanced jack inputs. * 2 balanced jack outputs. * 24-bit/192kHz ADAT I/O. * 24-bit/192kHz Coaxiale S/PDif I/O switchable to AES/EBU. * MIDI I/O. 4. M-Audio Audiophile 192: - Up to 24-bit/192kHz audio - 2 balanced analog inputs (1/4” TRS) - 2 balanced analog outputs (1/4” TRS) - S/PDIF digital I/O (coaxial RCA connectors) with 2-channel PCM - SCMS copy protection control - Digital I/O supports surround-encoded AC-3 and DTS pass-through - Direct hardware input monitoring via separate balanced 1/4” TRS monitor outputs - Software routing of inputs and outputs - Digital I/O can be routed to/from external effects - 16-channel MIDI I/O - ASIO, WDM, GSIF 2 and Core Audio driver support for compatibility with most applications - 64-bit driver support for Windows - PCI 2.2 compatibility - Apple G5 compatible - Incompatible exceptions - Includes Ableton Live Lite music production software, so you can make music right away - Works with other Delta cards Technical Specifcations: - Compatibility - ASIO - WDM - GSIF 2 - Core Audio

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  • file-name encoding problems

    - by tenhouse
    I googled over this topic but couldn't find what I was looking for... the following "happend" to me: I had my files stored on a NTFS-USB Harddisk, because of space problems I moved them to an ext3 system....somehow the filename (content is still ok as far as I saw) encoding screwed up....my files look like the following now: Kküken <--- should have an "ü" Jäger <--- should be an "ä" Zwölf <--- should be an "ö" fünfte <-- should be an "ü" etc .... These are just examples, but already give me my first question Why has the "ü" two different representations? (Maybe I screw up, before I screw up and now I have a mixing of x different encoding-layers? :) ) I tried the following command: convmv -r -f UTF-8 -t ISO-8859-1 * This command work for some files (for example Zwölf) but not for all: iso-8859-1 doesn't cover all needed characters for: "fünfte" So Iguess it must be another encoding - but which? How can I find out this? And is there any way that I can still fix all of this?

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  • Convert audio file to FLAC with ffmpeg?

    - by elpsk
    can I convert one of this format to compatible 16000.0 Sample Rate FLAC file? kAudioFormatLinearPCM = 'lpcm', kAudioFormatAppleIMA4 = 'ima4', kAudioFormatMPEG4AAC = 'aac ', kAudioFormatMACE3 = 'MAC3', kAudioFormatMACE6 = 'MAC6', kAudioFormatULaw = 'ulaw', kAudioFormatALaw = 'alaw', kAudioFormatMPEGLayer1 = '.mp1', kAudioFormatMPEGLayer2 = '.mp2', kAudioFormatMPEGLayer3 = '.mp3', kAudioFormatAppleLossless = 'alac' I tried using ffmpeg ffmpeg -i audio.xxx -acodec flac audio.flac but result is FFmpeg version CVS, Copyright (c) 2000-2004 Fabrice Bellard Mac OSX universal build for ffmpegX configuration: --enable-memalign-hack --enable-mp3lame --enable-gpl --disable-vhook --disable-ffplay --disable-ffserver --enable-a52 --enable-xvid --enable-faac --enable-faad --enable-amr_nb --enable-amr_wb --enable-pthreads --enable-x264 libavutil version: 49.0.0 libavcodec version: 51.9.0 libavformat version: 50.4.0 built on Apr 15 2006 04:58:19, gcc: 4.0.1 (Apple Computer, Inc. build 5250) Input #0, wsaud, from 'audio.alac': Duration: 00:00:03.8, start: 0.000000, bitrate: 199 kb/s Stream #0.0: Audio: adpcm_ima_ws, 24931 Hz, stereo, 199 kb/s Unable for find a suitable output format for 'audio.flac' I also installed flac codec for mac, but nothing... I tried also use convtoflac.sh (from http://legroom.net/software/convtoflac) but result is similar. Any idea to convert in flac?

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  • Change the audio output device in Firefox

    - by Zanami Zani
    I'm trying to play music through Ventrilo and currently I use Virtual Audio Cable. The way it works is that in foobar2000 (a music playing program) I set the output device in preferences to Virtual Audio Cable. Then in Ventrilo I log in to another name and set the input device to Virtual Audio Cable. This routes the music through the Virtual Audio Cable and allows me to play the music through Ventrilo. However, I would also like to change the output device for Firefox (or any other browser) or "Plugin Container for Firefix" to Virtual Audio Cable so that I could play music from Pandora or YouTube on to Ventrilo. Unfortunately I could not find an option for this anywhere.

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  • Record the audio stream from HDMI monitor

    - by Nick
    I am trying to record sound playing on my comupter with Audacity but am running into some troubles. I have the stereo mix set to be the default audio recorder but it doesn't pick up the audio that is being played through my HDMI monitors speakers: Playback Recording When I plug in headphones the stereo mix will pick up the audio stream and I can record but not when playing through the HDMI. I have installed the latest audio drivers and have tried all the different record options to no avail. How can I capture the Audio stream going through the HDMI?

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  • How to auto detect text file encoding?

    - by ???
    There are many plain text files which were encoded in variant charsets. I want to convert them all to UTF-8, but before running iconv, I need to know its original encoding. Most browsers have an Auto Detect option in encodings, however, I can't check those text files one by one because there are too many. Only having known the original encoding, I then can convert the texts by iconv -f DETECTED_CHARSET -t utf-8. Is there any utility to detect the encoding of plain text files? It doesn't have to be a 100% perfect correct, but it should recognize most of them.

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  • Dynamic audio score/music

    - by Joel Martinez
    I'm interested in developing a game who's background music changes with the mood and scenario of the game's action. Of course many existing games do this (halo for example), but I was interested in any resources/papers/articles talking about the techniques to develop a system like this. I have some ideas, and I understand that this will be equally challenging to implement at the code level as it will be to come up or acquire music that fits this model. Any links or, answers with ideas in them would he appreciated. Edit: this is the kind of info I'm looking for :) http://halo.bungie.org/misc/gdc.2002.music/

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  • Remove Audio stream from XVID files

    - by Kyle Brandt
    I have a bunch of Xvid files that each have an audio stream that I do not want. How can I strip the audio track I don't want using the Linux command line? I don't need the whole script (loop), just what command I would use to process each avi file individually (unless the cmd itself has batch modification built into it). I don't believe the file is in an mkv container, as mkvinfo doesn't find anything. Here is part of mplayer's output (thanks ~quack): [aviheader] Video stream found, -vid 0 ID_AUDIO_ID=1 [aviheader] Audio stream found, -aid 1 ID_AUDIO_ID=2 [aviheader] Audio stream found, -aid 2 VIDEO: [XVID] 512x384 12bpp 25.000 fps 1013.4 kbps (123.7 kbyte/s)

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  • Play audio over network with Windows 7?

    - by Josh
    I have a unique situation where I'd like to stream audio (ALL audio, not just mp3s, etc) from my laptop to another computer over the network. I live in a studio apartment and my laptop is my main computer but I'd like it's audio to play on my htpc with a nice stereo system. Since it's a studio, both computers are in the same room so I don't want 2 sets of speakers. I want my computer to directly play back through the stereo. I used to do this with pulseaudio but my job now requires that I run Windows full time. I'm aware of Shoutcast and other similar streaming solutions but I don't want any transcoding done. It's a waste of CPU and not to mention my laptop fans, and I don't mind the network bandwidth that uncompressed audio requires. Is there a way to run Shoutcast without encoding? Also, I know that Windows Remote Desktop can play audio over the network pretty easily. Is this part of .Net that I could just code a simple app that streams the audio without RD'ing in? I also don't want to run it over a physical wire. :)

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  • Audio card with built-in ground isolator?

    - by Dave Jarvis
    What audio cards would you recommend that eliminate hum, and hard-drive & mouse movement signal interference? Hardware components: Motherboard. Asus P5Q SE Audio. Realtek ALC 1200, 8-Channel High-Definition Audio CODEC (on board) Harddrive. WD Caviar 320 GB Mouse. Logitech Marbleman USB Mixer. Mackie d.4 Pro Amplifier. Sonance Sonamp 260 All components are plugged into the same Monster Power HDP 910 powerbar (does not help eliminate noise). I have no other components plugged in. The computer uses a Monster iCable 1000 to go from mini (on board audio) to RCA (mixer). I have moved the cable as far from other cables as possible. A ground loop isolator between the mixer and on board audio eliminates all noise. I would rather not use a ground loop isolator; an internal audio card that is Linux-compatible (Kubuntu) would be ideal. Suggestions?

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  • Downmix ALL SYSTEM audio to mono - Windows 7

    - by Mike K.
    I'm deaf in one ear and want to use my headphones when playing a game and talking with my friends on Skype/TS/Mumble/etc while also sometimes listening to music. I need ALL my system audio to be downmixed to mono so that my ONE hearing ear gets ALL audio channels instead of split stereo audio. No, none of the other similar questions on superuser have a solution. My headphone properties does not have a 'Mono' option, I don't have a 'Headphone Virtualization' option, and my Realtek HD audio driver software doesn't have these options either (driver was updated 11/14/2012). Don't even talk about setting the balance of one side of the headphones to 0. You're not paying attention if you suggest that. JACK and Virtual Audio Cable didn't work. It's possible I configured them wrong, but I followed the steps I found in related questions and still got split stereo out. TL;DR I need a viable, working, software solution (I say software because I have a USB headset) for forcing ALL system audio to mono so that I can hear literally everything through the one earpiece. Thanks!

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  • Windows Media Based video audio converter?

    - by acidzombie24
    This may seem like an odd question. Right now ONLY windows media player, VLC and media player classic opens and plays my audio video correctly. Virtualdub plays it back with the wrong framerate and losses the audio, Avidemux 2.5 seems to be able to dump the audio/video but the video (like all other apps) is either a bad framerate or is wrong (glitches and bad framerate or bad dump). Nothing recognizes the audio file and when playing the video Avidemux (and most other things) die. FFMPEG cant seem to split the video or audio (using copy -an and etc) and this is getting me very angry. VLC dumps the video incorrectly when i try dumping it with that too. What can i use to convert the video? its streaming so it starts at 26mins in and ends at 28 (this is where apps have the problem. They dont know this and fudge everything or crash). I manage to dump the audio with Avidemux but virtualdub and ffmpeg says unreconized codec. Even if i cant convert it (it seems compressed enough) i want to at least attach it back into an AVI.

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  • ffmpeg hangs when creating a video

    - by FearUs
    I am trying to insert an audio channel with a video: first of all I extract the audio from the original video for processing: ffmpeg -i lotr.mp4 lotr.wav I then extract all frames for later processing too: ffmpeg -i lotr.mp4 -f image2 %d.jpg When done processing audio and video streams, I try to create the video ffmpeg -f image2 -r 15 -i %d.jpg new.mp4 then merge with the audio: ffmpeg -i new.mp4 -i lotr.wav -map 0:0 -map 1:0 new_w_audio.mp4 Result: CPU activity = 100%, the process hangs and never returns. PS: I even tried it without modifying the images or the audio (so just trying to unpack then repack the video) but still the same output FFmpeg version SVN-r26400, Copyright (c) 2000-2011 the FFmpeg developers built on Jan 18 2011 04:07:05 with gcc 4.4.2 configuration: --enable-gpl --enable-version3 --enable-libgsm --enable-libvorb is --enable-libtheora --enable-libspeex --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-libvpx --disable-decoder=libvpx --arch=x86 --enable-runtime-cpudetect - -enable-libxvid --enable-libx264 --enable-librtmp --extra-libs='-lrtmp -lpolarss l -lws2_32 -lwinmm' --target-os=mingw32 --enable-avisynth --enable-w32threads -- cross-prefix=i686-mingw32- --cc='ccache i686-mingw32-gcc' --enable-memalign-hack libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'new.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Duration: 00:00:29.66, start: 0.000000, bitrate: 193 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], 192 k b/s, 15 fps, 15 tbr, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 [wav @ 01fed010] max_analyze_duration reached Input #1, wav, from 'lotr.wav': Duration: 00:00:29.90, bitrate: 176 kb/s Stream #1.0: Audio: pcm_s16le, 11025 Hz, 1 channels, s16, 176 kb/s File 'new_w_audio.mp4' already exists. Overwrite ? [y/N] y [buffer @ 01b03820] w:200 h:134 pixfmt:yuv420p Output #0, mp4, to 'new_w_audio.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], q=2-3 1, 200 kb/s, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1: Audio: aac, 11025 Hz, 1 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding

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