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• Can I get the Waves Maxx speaker effects to work in Ubuntu?

  - by Rafael

  I have a new machine that comes with JBL 2.1 Speakers with Waves Maxx Audio 3. On Windows it sounds perfect, though in Ubuntu 12.04 I get cheap/sound with simple mp3 files. I have tried a few things on different blogs but no luck so far. Any ideas? aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC665 Analog [ALC665 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC665 Digital [ALC665 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: N700 [Logitech Speaker Lapdesk N700], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0

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• Cheap sound on speakers - Dell XPS L502X

  - by Rafael

  I have a new machine that comes with JBL 2.1 Speakers with Waves Maxx Audio 3. On Windows it sounds perfect, though in Ubuntu 12.04 I get cheap/sound with simple mp3 files. I have tried a few things on different blogs but no luck so far. Any ideas? aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC665 Analog [ALC665 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC665 Digital [ALC665 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: N700 [Logitech Speaker Lapdesk N700], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0

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• There's no Sound Mixer menu, missing menu option in Sound Recorder

  - by AlexN

  I am using: -Ubuntu 11.10 -Skype -PS3 Eye Toy camera to input video and sound This setup has been properly working in former Ubuntu releases. To use the mic already built in on the PS3 Eye Toy camera I open de Sound Recorder app (notice: not inside Skype, from inside Skype it is not possible to do this) that is included in Gnome and then I go to FileSound Mixer, from this menu I can choose Gnome to get the input audio from the PS3 Eye Toy, instead of from the Audio-In of the computer. Now in Ubuntu 11.10 this Sound Mixer menu inside Sound Recorder is missing, Gnome says something like this: gnome-volume-control is not installed in the proper directory Note: I have tried this on Unity, Unity 2D, Gnome Classic, Gnome Classic 2D and Gnome Shell. In all of them the problem is the same. What can I do? Basically what I want to do is to be able to tell the computer to get the audio in from the PS3 Camera. Thanks in advance.

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• Is there now any way to convert mp3 files to m4a or aac 192kbit?

  - by piedro

  Since about two years now I am trying to find a way to convert high quality mp3 files to m4a or aac files with a fixed bitrate 192k. Please don't suggest using another format - i thought this through as far as it goes. The problem here is: ffmpeg obvioulsy can't convert to a higher bitrate than 152k. Even when it says it does so the resulting files still have 152k instead of 192k. ffmpeg also has/had a bug not writing the bitrate into the audio file tags which means when testing you have to calculate the bitrate manually by dividing the filesize by the length of the audio in seconds (resulting in 152k - see above) choosing faac as converter gets me the same results other programs don't work reliably (see this thread Howto convert audio files to *.m4a? I know that this is not an original new problem but I am wondering if there is still no way to convert with ubuntu/kubuntu 12.04 after a lot time passed and I can't find some of the bug issues mentioned in the other thread anymore. So: Is there a solution after all?

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• Problem with sound in Kubuntu 12.10

  - by Mihkel

  I'm really enjoying Kubuntu 12.10 experience, but the problem starts with sound. It wasn't here before, but today sound sounds garbled and echoed and wrong. It happens in Audacity and VLC. It doesn't happen when I test the sound devices nor when I use Amarok to play the music files (but come on, who uses Amarok to listen to a random music file, it's much more natural to use VLC for that ;-) ) Kubuntu/Phonon recognizes 2 sound devices: 1) RV770 HDMI Audio [Radeon HD 4850/4870] Digital Stereo [HDMI] 2) Built-in Audio Analog Stereo I know it has to use the second option, and it probably does, but that's not the case. What I did find out was that I had to rescan for audio devices in Audacity (and probably select "sysdefault") for it to sound normal. Why does it happen? I've tried following some other questions, but well.

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• How to Install Linux on my PC

  - by Holic

  Hi i need some help to install the drivers from my pc, on Ubuntu 10.10 i just installed it, and i a newbie on Ubuntu, but i understand a bit of Windows...but i want to try ubuntu and then Maybe change to UBUNTU!!! My hardware: QuadCore Intel Core i7-870, 3266 MHz (24 x 136) Asus P7P55D-E (2 PCI, 3 PCI-E x1, 2 PCI-E x16, 4 DDR3 DIMM, Audio, Gigabit LAN, IEEE-1394) NVIDIA GeForce GTX 480 (1536 MB) nVIDIA HDMI @ nVIDIA GF100 - High Definition Audio Controller VIA VT1828S @ Intel Ibex Peak PCH - High Definition Audio Controller [B-3] DIMM1: G Skill F3-12800CL9-2GBRL 2 GB DDR3-1333 DDR3 SDRAM (8-8-8-22 @ 609 MHz) (7-7-7-20 @ 533 MHz) (6-6-6-17 @ 457 MHz) DIMM3: G Skill F3-12800CL9-2GBRL 2 GB DDR3-1333 DDR3 SDRAM (8-8-8-22 @ 609 MHz) (7-7-7-20 @ 533 MHz) (6-6-6-17 @ 457 MHz) my pc is not connected to the internet with a wire(RJ45) but with a wireless LAn Asus WL-167G-V3(wich i also whant to install if possible) Anything would've help me :) Cheers & Thank you!

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• 12.04 sound keeps auto-muting when idle

  - by fali

  I just installed 12.04 on an HP8510W. Everything works fine except for one weird behavior which I have noticed. When ever there is no audio playing, the audio mute indicator on the laptop is on. As soon as I start playing a you tube video the mute indicator turns off and I get sound. Here is my pulse audio output which says that the sink is suspended because it is idle: Welcome to PulseAudio! Use "help" for usage information. list-sinks 1 sink(s) available. index: 0 name: <alsa_output.pci-0000_00_1b.0.analog-stereo> driver: <module-alsa-card.c> flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY state: SUSPENDED suspend cause: IDLE I tried running alsamixer, but I don't see the auto-mute option.

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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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• SSIS - XML Source Script

  - by simonsabin

  The XML Source in SSIS is great if you have a 1 to 1 mapping between entity and table. You can do more complex mapping but it becomes very messy and won't perform. What other options do you have? The challenge with XML processing is to not need a huge amount of memory. I remember using the early versions of Biztalk with loaded the whole document into memory to map from one document type to another. This was fine for small documents but was an absolute killer for large documents. You therefore need a streaming approach. For flexibility however you want to be able to generate your rows easily, and if you've ever used the XmlReader you will know its ugly code to write. That brings me on to LINQ. The is an implementation of LINQ over XML which is really nice. You can write nice LINQ queries instead of the XMLReader stuff. The downside is that by default LINQ to XML requires a whole XML document to work with. No streaming. Your code would look like this. We create an XDocument and then enumerate over a set of annoymous types we generate from our LINQ statement XDocument x = XDocument.Load("C:\\TEMP\\CustomerOrders-Attribute.xml");   foreach (var xdata in (from customer in x.Elements("OrderInterface").Elements("Customer")                        from order in customer.Elements("Orders").Elements("Order")                        select new { Account = customer.Attribute("AccountNumber").Value                                   , OrderDate = order.Attribute("OrderDate").Value }                        )) {     Output0Buffer.AddRow();     Output0Buffer.AccountNumber = xdata.Account;     Output0Buffer.OrderDate = Convert.ToDateTime(xdata.OrderDate); } As I said the downside to this is that you are loading the whole document into memory. I did some googling and came across some helpful videos from a nice UK DPE Mike Taulty http://www.microsoft.com/uk/msdn/screencasts/screencast/289/LINQ-to-XML-Streaming-In-Large-Documents.aspx. Which show you how you can combine LINQ and the XmlReader to get a semi streaming approach. I took what he did and implemented it in SSIS. What I found odd was that when I ran it I got different numbers between using the streamed and non streamed versions. I found the cause was a little bug in Mikes code that causes the pointer in the XmlReader to progress past the start of the element and thus foreach (var xdata in (from customer in StreamReader("C:\\TEMP\\CustomerOrders-Attribute.xml","Customer")                                from order in customer.Elements("Orders").Elements("Order")                                select new { Account = customer.Attribute("AccountNumber").Value                                           , OrderDate = order.Attribute("OrderDate").Value }                                ))         {             Output0Buffer.AddRow();             Output0Buffer.AccountNumber = xdata.Account;             Output0Buffer.OrderDate = Convert.ToDateTime(xdata.OrderDate);         } These look very similiar and they are the key element is the method we are calling, StreamReader. This method is what gives us streaming, what it does is return a enumerable list of elements, because of the way that LINQ works this results in the data being streamed in. static IEnumerable<XElement> StreamReader(String filename, string elementName) {     using (XmlReader xr = XmlReader.Create(filename))     {         xr.MoveToContent();         while (xr.Read()) //Reads the first element         {             while (xr.NodeType == XmlNodeType.Element && xr.Name == elementName)             {                 XElement node = (XElement)XElement.ReadFrom(xr);                   yield return node;             }         }         xr.Close();     } } This code is specifically designed to return a list of the elements with a specific name. The first Read reads the root element and then the inner while loop checks to see if the current element is the type we want. If not we do the xr.Read() again until we find the element type we want. We then use the neat function XElement.ReadFrom to read an element and all its sub elements into an XElement. This is what is returned and can be consumed by the LINQ statement. Essentially once one element has been read we need to check if we are still on the same element type and name (the inner loop) This was Mikes mistake, if we called .Read again we would advance the XmlReader beyond the start of the Element and so the ReadFrom method wouldn't work. So with the code above you can use what ever LINQ statement you like to flatten your XML into the rowsets you want. You could even have multiple outputs and generate your own surrogate keys.        

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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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• 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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• 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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