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• Sound issue in Lubuntu

  - by jvsa90

  I'm recently having a problem in my Lubuntu deskptop: sound through the speakers doesn't seem to work. The funny thing is: it works when I plug in my earphones. I've tried to unmute everything with pavucontrol and alsamixer, but everything seems to be OK. $ sudo aplay -l **** Liste der Hardware-Geräte (PLAYBACK) **** Karte 0: Intel [HDA Intel], Gerät 0: HDA Generic [HDA Generic] Sub-Geräte: 0/1 Sub-Gerät #0: subdevice #0 $ lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation NM10/ICH7 Family High Definition Audio Controller (rev 02) Subsystem: Acer Incorporated [ALI] Device 034a Flags: bus master, fast devsel, latency 0, IRQ 44 Memory at 58200000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd-hda-intel Can anyone guess what's happening? It has worked until recently and it definitely works in my Windows partition.

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• Generic Content Player?

  - by Jantire

  The general idea on the web appears to be that video/audio are to be separated with plain text. By separated, I mean you have a place that plays video/audio and a place that you read text. This is because it is widely understood that they are vastly different. However, audio and video are just another way of communication, just like text. So why do we separate the two even if they are nearly the same thing? Correct me if I'm wrong but, most tutorials are either plain text how-to's (wiki-style) or visual/auditory instructional videos (YouTube). Why aren't the two combined? Or, if it's already been done can someone reply with the link? This might be bordering off-topic and if it is off-topic then please point me to the right place so it won't be. This might also appear to be an obvious question, however I'm not sure if this subject has really been deeply thought-out by more than a few individuals.

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• 12.04 - sound is laggy when running games through Wine

  - by orzechowskid

  Lenovo U400 Wine 1.5.5 Ubuntu 12.04 with all updates applied I'm experiencing severe (~500ms) audio lag in all games run in Wine. Portal 2, Half-Life, World of Goo, and Fallout are all exhibiting this problem. When I run winecfg though and click the "Test Sound" button at the bottom of the Audio tab, the sound effect appears to play immediately. So I'm not sure what's going on. I don't think it's a problem with PulseAudio by itself since totem videos and Youtube clips both play in perfect sync. Anyone have any ideas on where to start fixing this? thanks! (edit: I thought this was limited to Steam games but I installed a non-Steam game and I now see that's not the case. I get audio lag in other apps too.)

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• Are HDMI to VGA Adapters Really Device-Specific?

  - by allquixotic

  There are a lot of devices on the market right now (especially mobile devices) with a Micro-HDMI or Mini-HDMI port and no VGA or D-Sub output. Most manufacturers of said devices sell a cable that looks something like this: I have yet to find a cable like this that claims to work on a wide array of devices. In general, these cables claim to work with one specific device only. The way these cables work, I think, is that analog VGA signals are sent from the HDMI port on the device. This should work for devices that have special hardware on the motherboard/GPU capable of driving this. Is it the case that these cables have to be custom designed for each device? Or, is it rather that any device which possesses this special "signaling of analog VGA over the HDMI port" can be made to work with a cable that is physically compatible (i.e. the HDMI end plugs into the device and the VGA end accepts a VGA monitor cable)? Note that I am not looking for a product recommendation, just a conceptual clarification on what exactly these devices are doing. Also, a few remarks: The cables like the one depicted here are not digital to analog converters. I know about these: they are expensive, and they are the ONLY solution if your device only outputs a digital signal and is incapable of driving analog VGA over the HDMI port. The cables like the one depicted here are not straight crossover cables from VGA to HDMI, either. The crossover cables are designed to send a digital HDMI signal over the VGA port's wires; that is, the wire protocol is HDMI (digital) but the physical pinout is the same as VGA, even though nothing analog is happening. Once again, this is not the behavior that, I believe, the devices which I'm talking about in this question are doing. The cabling and devices that this question is about transmit the analog VGA data over the HDMI port (the HDMI port is in the device outputting the data, and the VGA side is the monitor/projector).

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• Building a Media Center PC with Comcast Cable...?

  - by Rob

  Alright - so this might be a stupid question but I've never been all that much into TV. I currently have Comcast cable. I've just got the 'basic' 2-60 package or whatever; I've just always plugged the cable into the back of my TV. I've never had a cable box. Recently, Comcast has been pulling channels off of my line-up. Most recently, the stole the TV Guide channel from me. I'm told this is part of a push to get customers to switch to their digital line-up. But, I'm also told it requires some sort of digital receiver for each TV you've got. I don't want to buy a bunch of these digital receivers and I don't want to pay the monthly rental fee...but I have heard of how awesome media center PCs are and some really cool things they can do. And, I've got loads of PC parts sitting around. So, can someone guide me through this a bit? Are there computer video cards or TV tuners that are going to work with Comcast's digital cable? What kind of price range are we looking at?

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• How can I make sound over hdmi in kubuntu work?

  - by user32509

  I have used a hdmi cable to connect my lcd (which is connected with my speakers) with my nvida 275 gtx grafic card. I can not get the sound output to work. The hardware itself is working probably - I tested it under windows. Currently I am running Kubuntu 9.10 64 with Nvidia 190.53. The sound output worked fine before I installed the hdmi connection. (German output - i can change it, if you tell me how :)) aplay -l **** Liste von PLAYBACK Geräten **** Karte 0: Intel [HDA Intel], Gerät 0: ALC889A Analog [ALC889A Analog] Untergeordnete Geräte: 1/1 Untergeordnetes Gerät '0: subdevice #0 Karte 0: Intel [HDA Intel], Gerät 1: ALC889A Digital [ALC889A Digital] Untergeordnete Geräte: 1/1 Untergeordnetes Gerät '0: subdevice #0 aplay -L front:CARD=Intel,DEV=0 HDA Intel, ALC889A Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, ALC889A Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, ALC889A Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, ALC889A Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, ALC889A Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, ALC889A Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Intel,DEV=0 HDA Intel, ALC889A Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) pulse Playback/recording through the PulseAudio sound server And i disabled mute in kmix an all channels :)

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• Use DivX settings to encode to mp4 with ffmpeg

  - by sjngm

  I'm used to use VirtualDub to encode a video to AVI container with DivX-codec (and MP3 for audio). Now I'm planning to use ffmpeg to encode videos to MP4 container with h264-codec. What I've figured out is that I need to use libx264 and one of those presets to make anything work. However, I'm amazed about the video bitrate ffmpeg uses for encoding. What I currently have is this little batch file: @ECHO OFF SETLOCAL SET IN=source.avs SET FFMPEG_PATH=C:\Program Files (x86)\ffmpeg SET PRESET=-fpre "%FFMPEG_PATH%\presets\libx264-lossless_slow.ffpreset" SET AUDIO=-acodec libmp3lame -ab 128000 SET VIDEO=-vcodec libx264 -vb 1978000 "%FFMPEG_PATH%\ffmpeg.exe" -i %IN% %AUDIO% %VIDEO% %PRESET% test.mp4 ENDLOCAL With this I tell ffmpeg to use 1978k as the bitrate, but ffmpeg uses 15000k+! I tried other presets, but they don't use my specified bitrate. Here are the presets I have: libx264-baseline.ffpreset libx264-ipod320.ffpreset libx264-ipod640.ffpreset libx264-lossless_fast.ffpreset libx264-lossless_max.ffpreset libx264-lossless_medium.ffpreset libx264-lossless_slow.ffpreset libx264-lossless_slower.ffpreset libx264-lossless_ultrafast.ffpreset ffmpeg version: FFmpeg git-N-29181-ga304071 libavutil 50. 40. 1 / 50. 40. 1 libavcodec 52.120. 0 / 52.120. 0 libavformat 52.108. 0 / 52.108. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 79. 0 / 1. 79. 0 libswscale 0. 13. 0 / 0. 13. 0 Note that I don't use the latest version as it has problems with spaces in filenames. Here's what seems to be the full parameter list DivX 6.9.2 uses: -bvnn 1978000 -vbv 218691200,100663296,100663296 -dir "C:\Users\sjngm\AppData\Roaming\DivX\DivX Codec" -w -b 1 -use_presets=1 -preset=10 -windowed_fullsearch=2 -thread_delay=1 What command line parameters would that be for ffmpeg? EDIT: Going with slhck's suggestion I tried a new 32-bit version. I have no idea if that is 0.9 or newer, I can't find that info. ffmpeg version N-36890-g67f5650 libavutil 51. 34.100 / 51. 34.100 libavcodec 53. 56.105 / 53. 56.105 libavformat 53. 30.100 / 53. 30.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 59.100 / 2. 59.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 51. 2.100 / 51. 2.100 I reworked my batch file to look like this (interestingly enough I can't find parameter -vprofile in the documentation): @ECHO OFF SETLOCAL SET IN=VTS_01_1.avs SET FFMPEG_PATH=C:\Program Files (x86)\ffmpeg SET PRESET=-vprofile high -preset veryslow SET AUDIO=-acodec libmp3lame -ab 128000 SET VIDEO=-vcodec libx264 -vb 1978000 "%FFMPEG_PATH%\ffmpeg.exe" -i %IN% %AUDIO% %PRESET% %VIDEO% test.mp4 ENDLOCAL I see that it now uses the bitrate properly (thanks to LongNeckbeard for pointing out that the lossless-stuff ignores the bitrate!). Just in case you wonder how I came up with the 1978000, I'm using this formula which I found valid for DivX-files (I'm guessing the bitrate won't change that much for h264): width * height * 25 * 0.22 / 1000 I'm not sure if the 0.22 correlates with the CRF somehow. Overall I forgot to say the I will use a two-pass scenario, which is why I don't use the CRF here. I will try to read more about this. Currently I'm just trying to get something running that shows me that I'm doing something right (ffmpeg isn't the easiest tool to understand ;)). C:\Program Files (x86)\ffmpeg\ffmpeg.exe" -i VTS_01_1.avs -acodec libmp3lame -ab 128000 -vcodec libx264 -vb 1978000 -vprofile high -preset veryslow test.mp4 The output is now: ffmpeg version N-36890-g67f5650 Copyright (c) 2000-2012 the FFmpeg developers built on Jan 16 2012 21:57:13 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 34.100 / 51. 34.100 libavcodec 53. 56.105 / 53. 56.105 libavformat 53. 30.100 / 53. 30.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 59.100 / 2. 59.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 51. 2.100 / 51. 2.100 Input #0, avs, from 'VTS_01_1.avs': Duration: 00:58:46.12, start: 0.000000, bitrate: 0 kb/s Stream #0:0: Video: rawvideo (YV12 / 0x32315659), yuv420p, 576x448, 77414 kb/s, 25 tbr, 25 tbn, 25 tbc Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s File 'test.mp4' already exists. Overwrite ? [y/N] y w:576 h:448 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param: [libx264 @ 05A2C400] using cpu capabilities: MMX2 SSE2Fast FastShuffle SSEMisalign LZCNT [libx264 @ 05A2C400] profile High, level 3.1 [libx264 @ 05A2C400] 264 - core 120 r2120 0c7dab9 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=16 deblock=1:0:0 analyse=0x3:0x133 me=umh subme=10 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=24 chroma_me=1 trellis=2 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=3 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=8 b_pyramid=2 b_adapt=2 b_bias=0 direct=3 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=60 rc=abr mbtree=1 bitrate=1978 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'test.mp4': Metadata: encoder : Lavf53.30.100 Stream #0:0: Video: h264 (![0][0][0] / 0x0021), yuv420p, 576x448, q=-1--1, 1978 kb/s, 25 tbn, 25 tbc Stream #0:1: Audio: mp3 (i[0][0][0] / 0x0069), 48000 Hz, 2 channels, s16, 128 kb/s Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> libx264) Stream #0:1 -> #0:1 (pcm_s16le -> libmp3lame) Press [q] to stop, [?] for help frame= 0 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 0 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 0 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 3 fps= 1 q=22.0 size= 39kB time=00:00:00.04 bitrate=8063.8kbits/ frame= 8 fps= 2 q=22.0 size= 82kB time=00:00:00.24 bitrate=2801.3kbits/ frame= 13 fps= 3 q=23.0 size= 120kB time=00:00:00.44 bitrate=2229.5kbits/ frame= 16 fps= 4 q=23.0 size= 147kB time=00:00:00.56 bitrate=2156.7kbits/ frame= 20 fps= 4 q=22.0 size= 175kB time=00:00:00.72 bitrate=1987.4kbits/ : video:4387kB audio:273kB global headers:0kB muxing overhead 0.260038% [libx264 @ 05A2C400] frame I:2 Avg QP:19.53 size: 29850 [libx264 @ 05A2C400] frame P:76 Avg QP:22.24 size: 19541 [libx264 @ 05A2C400] frame B:359 Avg QP:25.93 size: 8210 [libx264 @ 05A2C400] consecutive B-frames: 0.5% 0.5% 0.0% 8.2% 17.2% 52.2% 16.0% 5.5% 0.0% [libx264 @ 05A2C400] mb I I16..4: 5.4% 75.3% 19.3% [libx264 @ 05A2C400] mb P I16..4: 1.3% 16.5% 2.2% P16..4: 36.3% 28.6% 12.7% 1.8% 0.2% skip: 0.4% [libx264 @ 05A2C400] mb B I16..4: 0.4% 3.8% 0.3% B16..8: 40.0% 18.4% 4.7% direct:18.5% skip:13.9% L0:45.4% L1:38.1% BI:16.5% [libx264 @ 05A2C400] final ratefactor: 20.35 [libx264 @ 05A2C400] 8x8 transform intra:83.1% inter:68.5% [libx264 @ 05A2C400] direct mvs spatial:99.2% temporal:0.8% [libx264 @ 05A2C400] coded y,uvDC,uvAC intra: 64.9% 83.4% 49.2% inter: 49.0% 50.4% 4.4% [libx264 @ 05A2C400] i16 v,h,dc,p: 25% 22% 27% 26% [libx264 @ 05A2C400] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 10% 7% 23% 9% 10% 10% 10%10% 13% [libx264 @ 05A2C400] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 12% 11% 13% 9% 12% 11% 10% 9% 12% [libx264 @ 05A2C400] i8c dc,h,v,p: 42% 28% 16% 14% [libx264 @ 05A2C400] Weighted P-Frames: Y:18.4% UV:7.9% [libx264 @ 05A2C400] ref P L0: 29.1% 11.3% 15.7% 7.3% 6.9% 4.9% 5.1% 3.4%3.9% 2.7% 2.8% 1.8% 1.7% 1.2% 1.4% 0.9% [libx264 @ 05A2C400] ref B L0: 68.8% 11.4% 5.5% 2.9% 2.3% 1.9% 1.5% 1.1%1.1% 1.0% 0.9% 0.7% 0.5% 0.3% 0.1% [libx264 @ 05A2C400] ref B L1: 91.9% 8.1% [libx264 @ 05A2C400] kb/s:2055.88 As far as I'm concerned it doesn't look that bad to me.

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• Sound Effects/Manipulation?

  - by Adam

  Hello, I am creating an android app that basically records an applies an "Effect" on the audio track then plays it back. I got my app to record an play back but I am stuck an not sure where do go from here. I have been Googling for days now trying to find a open source audio library or some way to change the audio after I record it. I currently have it setup to play back using SoundPool an I't lets me speed up an slow down the audio. I would like to do things like change pitch an add echo etc. I will appreciate any responses because I am totally stumped right now. Thanks Adam

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• VU meter implementaion in iphone

  - by Sreelal

  Hi, I am developing an aplication for iphone which records audio and save that audio file .I need to create a UI similar to that in Voice Memo app with VU meter .I implemented codes to record audio,but i have no idea about VU meter implementation.Looking forward for a reply ......Thanks in advance

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• avaudioplayer interferes with mpmovieplayer on ipad

  - by user175826

  my app plays video and audio. however, i have a problem where once i play an audio file using avaudioplayer, the video refuses to play. when i play the video first, everything is fine. but if the audio is played first, any time i try to play the video it simply pops up the video player but will not play the actual video (you can use the scroller to go to any point in the video, but no playback will happen). this issue does not come up on the iphone, nor on the ipad simulator. clearly there is some resource conflict here, probably related to the audio, and i'd welcome some input on how to address it.

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• Sound File editing in Objective C

  - by Biranchi

  Hi All, I am able to record and create audio files using AudioFileCreateWithURL in the AudioToolbox Framework. I want to figure out if there is any way to edit the .caf sound files. I want to insert another recoreded audio inside the main audio file. Any thoughts or suggestions how to proceed ?? Thanks.

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• Submit information to url, but also open PDF

  - by Mad Ducky Digital Branding

  I have a client whose desire is to have her Wordpress blog show a MailChimp form on her home page as a gateway to a .pdf. I need the following behavior to occur when the user clicks "Submit": execute the included MailChimp's javascript file; this ensures the form was properly filled, and then performs the sign-up to the newsletter list (don't need help with this part) then show the user an informational PDF for download or viewing EDIT: The logical order was flipped from when I originally posted this. The script should execute, and only if the script gets executed properly should the PDF show to the user Note: My experience level with HTML and PHP is 3/4, and with JS I am 2/4 EDIT: (seems more like 1/4 at this point lol). If my research is correct, PHP (server-side language) would be used to do that which the client wants. Additional validation is not necessary beyond what MailChimp's script provides (it ensures that user has submitted a completed form) is not necessary in this case (the client says it's ok if the e-mail isn't valid at all). EDIT: Reworded this sentence from original post to be more clear The .pdf URL and content is static, and simply needs to be shown, not generated. ----RESEARCH---- I know that the Mailchimp form uses the following line to actually submit the information, but I want to do the action mentioned below, as well as open the aforementioned .pdf: <form action="http://*BLAH*.us2.list-manage.com/subscribe/post?u=*BLAHBLAH*&amp;id=*BLAHBLAHBLAH*" method="post" id="mc-embedded-subscribe-form" name="mc-embedded-subscribe-form" class="validate" target="_blank"> I am reading on other sites that I can conceivably point "action" to a .php file, but if there is a way to do this with javascript - since its using the .js file that I created for that already anyways, then I would be most happy. Barring that, I'll take what I can get.. ----SOLUTION?---- ...

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• RoR | how to get content_tags to nest?

  - by Digital Cake

  As you can see I have a helper with a method that I'm trying to render out to the view. The nested content_tags do not render what is my disconnect about this tag? def draw_calendar(selected_month, month, current_date) content_tag(:table) do content_tag(:thead) do content_tag(:tr) do I18n.t(:"date.abbr_day_names").map{ |day| content_tag(:th, day, :escape => false) } end #content_tag :tr end #content_tag :thead content_tag(:tbody) do month.collect do |week| content_tag(:tr, :class => "week") do week.collect do |date| content_tag(:td, :class => "day") do content_tag(:div, date.day, :class => (Date.today == current_date ? "today" : nil)) end #content_tag :td end #week.collect end #content_tag :tr end #month.collect end #content_tag :tbody end #content_tag :table end #draw_calendar

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• How do i pipe stdout/stderr in .NET?

  - by acidzombie24

  I want to do something like this ffmpeg -i audio.mp3 -f flac - | oggenc2.exe - -o audio.ogg i know how to do ffmpeg -i audio.mp3 -f flac using the process class in .NET but how do i pipe it to oggenc2? Any example of how to do this (it doesnt need to be ffmpeg or oggenc2) would be fine.

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• Avoiding shutdown hook

  - by meryl

  Through the following code I can play and cut and audio file. Is there any other way to avoid using a shutdown hook? The problem is that whenever I push the cut button , the file doesn't get saved until I close the application 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(); // We add a shutdown hook, an anonymous inner class. Runtime.getRuntime().addShutdownHook(new Thread() { public void run() { // We're now in the hook, which means the program is shutting down. // You would need to use better exception handling in a production application. try { // Stop the audio clip. c.stop(); // Create a new input stream, with the duration set to the frame count we reached. Note that we use the previously determined audio format AudioInputStream startStream = new AudioInputStream(new FileInputStream(inputAudio), audioFormat, c.getLongFramePosition()); // Write it out to the output file, using the same file type. AudioSystem.write(startStream, fileType, outputAudio); } catch(IOException e) { e.printStackTrace(); } } }); // After setting up the hook, we start the clip. c.start(); } catch (UnsupportedAudioFileException e) { e.printStackTrace(); } catch (IOException e) { e.printStackTrace(); } catch (LineUnavailableException e) { e.printStackTrace(); } }// end play_cut ......................

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• casting void* to float* creates only zeros

  - by Paperflyer

  I am reading an audio file using CoreAudio (Extended Audio File Read Services). The audio data gets converted to 4-byte float and handed to me as a void* buffer. It can be played with Audio Queue Services, so its content is correct. Next, I want to draw a waveform and thus need access to the actual samples. So, I cast void* audioData to float*: Float32 *floatData = (Float32 *)audioData; When accessing this data however, I only get 0.0 regardless of the index. Float32 value = floatData[index]; // Is always zero for any index Am I doing something wrong with the cast?

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• Virtual microphone, networks and vb.net

  - by Jonathan

  I would like to add a virtual microphone (similar to how you can have a virual CD drive and then mount ISO files on it.) so that it can be selectable in programs like MSN and skype. But have the source of the audio be streamed from over a network(I know how to stream the audio over the network in VB.net) but how do I get that audio which has been streamed as the input to the virtual microphone? Jonathan

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• Virtual microphone, networks and vb.net

  - by Jonathan

  I would like to add a virtual microphone (similar to how you can have a virual CD drive and then mount ISO files on it.) so that it can be selectable in programs like MSN and skype. But have the source of the audio be streamed from over a network(I know how to stream the audio over the network in VB.net) but how do I get that audio which has been streamed as the input to the virtual microphone? Jonathan

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• Is Streaming Video possible with Sql Filestream?

  - by Lieven Cardoen

  We have stored all media in Sql Filestream, but now we'll need Video and Audio streaming... Will this be possible with Sql Filestream or will I have to take all of the Video and Audio out of the database? Which technology would you use to enable Video/Audio Streaming? WebORB FluorineFX Wowza (way better I think than the first two) IIS Media (haven't looked into this yet)

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• Using Mapkit to create a local searchable Map

  - by Digital D

  Using MapKit as a base, I'm planning on adding a map to a project with 'local search' capabilities. I think 'local search' describes the feature I want to design into the map. Here is my vision. The map is displayed on the bottom half of a view. The user's current location is highlighted by default. When the user pushes the 'search' button annotation pins drop onto the map. The search is programmatically fixed to a certain item....for example supermarkets. So supermarkets in a 5 mile radius of the user's current location will populate the map. How would I add this local search feature to the already amazing MapKit? I've learned an incredible amount as a new developer in the last few months, and look forward to learning googles...correction googols more. Thanks in anticipation.

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• Detecting the type of iPhone interrupt

  - by Prashant

  I can detect that the iPhone went to sleep and came back from sleep, by using the applicationWillResignActive and applicationDidBecomeActive. But how do I find out what kind of interrupt it was. I am making an audio player application, and need to keep the audio playing when the iPhone goes to sleep (which I know how to do). But I need to interrupt the audio when a message, alarm or low battery interrupt occurs. Also I need to resume the audio when the event is over. So how do I differentiate between these different interrupts.

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• Processing file uploads before object is saved

  - by Dominic Rodger

  I've got a model like this: class Talk(BaseModel): title = models.CharField(max_length=200) mp3 = models.FileField(upload_to = u'talks/', max_length=200) seconds = models.IntegerField(blank = True, null = True) I want to validate before saving that the uploaded file is an MP3, like this: def is_mp3(path_to_file): from mutagen.mp3 import MP3 audio = MP3(path_to_file) return not audio.info.sketchy Once I'm sure I've got an MP3, I want to save the length of the talk in the seconds attribute, like this: audio = MP3(path_to_file) self.seconds = audio.info.length The problem is, before saving, the uploaded file doesn't have a path (see this ticket, closed as wontfix), so I can't process the MP3. I'd like to raise a nice validation error so that ModelForms can display a helpful error ("You idiot, you didn't upload an MP3" or something). Any idea how I can go about accessing the file before it's saved? p.s. If anyone knows a better way of validating files are MP3s I'm all ears - I also want to be able to mess around with ID3 data (set the artist, album, title and probably album art, so I need it to be processable by mutagen).

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• How do you insert 9 MB file into a Blob Field Using Oracle.DataAccess?

  - by discwiz

  Trying to insert a large audio file into an Oracle 10g database and keep getting this error: ORA-01460: unimplemented or unreasonable conversion requested The byte array length of the audio file is 2702577. The procedure works with smaller array lengths, but not the larger ones. Here is my code and Thanks! Dim oracleConnection As New OracleClient.OracleConnection Dim Cmd As New OracleClient.OracleCommand Dim oracleDataAdapter As New OracleDataAdapter oracleConnection.ConnectionString = System.Configuration.ConfigurationManager.AppSettings("MasterConnectionODT") Cmd.Connection = oracleConnection Cmd.CommandText = "Audio.ADD_AUDIO" Cmd.CommandType = CommandType.StoredProcedure Dim aParam As New OracleClient.OracleParameter aParam.ParameterName = "I_FACILITY_ID_C" aParam.OracleType = OracleType.Char aParam.Value = FacID aParam.Direction = ParameterDirection.Input Cmd.Parameters.Add(aParam) aParam = New OracleParameter aParam.ParameterName = "I_TARP_ID_N" aParam.OracleType = OracleType.Number aParam.Value = TarpID aParam.Direction = ParameterDirection.Input Cmd.Parameters.Add(aParam) aParam = New OracleParameter aParam.ParameterName = "I_AUDIO_BLOB" aParam.OracleType = OracleType.Blob aParam.Value = Audio aParam.Direction = ParameterDirection.Input Cmd.Parameters.Add(aParam) Using oracleConnection oracleConnection.Open() Cmd.ExecuteNonQuery() End Using

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• Echo certain value from smarty array

  - by zx

  Hi, So currently I have an array with smarty.. {foreach from=$_sequences key=k item=v} Name => {$v.menu} Type => {$v.type} Step => {$v.pri} Data =>{$v.data} {/foreach} which gives me Name = Test Type = Audio Step = 1 Data = audio1 Name = Test2 Type = Audio Step = 2 Data = audio2 Name = Test3 Type = Audio Step = 3 Data = audio3 Now how would I get the data for step = 2 to echo out? So from that foreach I only want to display "audio2"

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• Javascript force GC collection? / Forcefully free object?

  - by plash

  I have a js function for playing any given sound using the Audio interface (creating a new instance for every call). This works quite well, until about the 32nd call (sometimes less). This issue is directly related to the release of the Audio instance. I know this because I've allowed time for the GC in Chromium to run and it will allow me to play another 32 or so sounds again. Here's an example of what I'm doing: <html><head> <script language="javascript"> function playSound(url) { snd = new Audio(url); snd.play(); delete snd; snd = null; } </script> </head> <body> <a href="#" onclick="playSound('blah.mp3');">Play sound</a> </body></html> I also have this, which works well for pages that have less than 32 playSound calls: var AudioPlayer = { cache: {}, play: function(url) { if (!AudioPlayer.cache[url]) AudioPlayer.cache[url] = new Audio(url); AudioPlayer.cache[url].play(); } }; But this will not work for what I want to do (dynamically replace a div with other content (from separate files), which have even more sounds on them - 1. memory usage would easily skyrocket, 2. many sounds will never play). I need a way to release the sound immediately. Is it possible to do this? I have found no free/close/unload method for the Audio interface. The pages will be viewed locally, so the constant loading of sounds is not a big factor at all (and most sounds are rather short).

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