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  • Performance: float to int cast and clippling result to range

    - by durandai
    I'm doing some audio processing with float. The result needs to be converted back to PCM samples, and I noticed that the cast from float to int is surprisingly expensive. Whats furthermore frustrating that I need to clip the result to the range of a short (-32768 to 32767). While I would normally instictively assume that this could be assured by simply casting float to short, this fails miserably in Java, since on the bytecode level it results in F2I followed by I2S. So instead of a simple: int sample = (short) flotVal; I needed to resort to this ugly sequence: int sample = (int) floatVal; if (sample > 32767) { sample = 32767; } else if (sample < -32768) { sample = -32768; } Is there a faster way to do this? (about ~6% of the total runtime seems to be spent on casting, while 6% seem to be not that much at first glance, its astounding when I consider that the processing part involves a good chunk of matrix multiplications and IDCT)

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  • Playing NSF music in FMOD.net

    - by Tesserex
    So, as the title says, I want to be able to play NSF files using FMOD, because my project already uses FMOD and I'd rather not replace it. This will involve figuring out how existing players and emulators work and porting it. I haven't yet found an existing player that uses FMOD. My starting point is the MyNes source from http://sourceforge.net/projects/mynes/. There are two big steps between here and what I'm looking for. MyNes plays from a ROM, not NSF. So, I have to rip out the APU and get it to play NSF files. The MyNes APU uses SlimDX, so I have to convert that to FMOD.NET. I am really stuck about how to go about either of these, because I'm not that familiar with audio formats and it's hard finding resources online. So here are a few questions: From what I can tell from the NSF spec at http://kevtris.org/nes/nsfspec.txt, it's just contains the relevant memory section of the ROM, plus the header. If anyone can verify or correct this that would be great. The emulator APU uses data from the rest of the emulator to play, including things like cycle counts. I'm not sure what replaces this in a standalone player. Can't I just load all the music data at once into a stream and play it? Joining #1 and #2, does the header data from the NSF substitute for some of the ROM data in the emulator code? Using FMOD, will I be following the usercreatedsound example for loading a stream? And does this format count as PCM? Specifically MyNes says PCM8. Any tips on loading / playing the stream in FMOD are appreciated. As an aside, I don't really understand the loading / playing sections of the spec I linked at all. It seems to apply to 6502 systems / emulators only and not to my situation. I know it's a long shot for anyone here to have enough experience in this area to help, but anything you can provide is definitely appreciated. A link to an existing .NET library that does this would be even better, but I don't believe one exists.

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  • Playing NSF music in FMOD.net

    - by Tesserex
    So, as the title says, I want to be able to play NSF files using FMOD, because my project already uses FMOD and I'd rather not replace it. This will involve figuring out how existing players and emulators work and porting it. I haven't yet found an existing player that uses FMOD. My starting point is the MyNes source from http://sourceforge.net/projects/mynes/. There are two big steps between here and what I'm looking for. MyNes plays from a ROM, not NSF. So, I have to rip out the APU and get it to play NSF files. The MyNes APU uses SlimDX, so I have to convert that to FMOD.NET. I am really stuck about how to go about either of these, because I'm not that familiar with audio formats and it's hard finding resources online. So here are a few questions: From what I can tell from the NSF spec at http://kevtris.org/nes/nsfspec.txt, it's just contains the relevant memory section of the ROM, plus the header. If anyone can verify or correct this that would be great. The emulator APU uses data from the rest of the emulator to play, including things like cycle counts. I'm not sure what replaces this in a standalone player. Can't I just load all the music data at once into a stream and play it? Joining #1 and #2, does the header data from the NSF substitute for some of the ROM data in the emulator code? Using FMOD, will I be following the usercreatedsound example for loading a stream? And does this format count as PCM? Specifically MyNes says PCM8. Any tips on loading / playing the stream in FMOD are appreciated. As an aside, I don't really understand the loading / playing sections of the spec I linked at all. It seems to apply to 6502 systems / emulators only and not to my situation. I know it's a long shot for anyone here to have enough experience in this area to help, but anything you can provide is definitely appreciated. A link to an existing .NET library that does this would be even better, but I don't believe one exists.

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  • ALSA samples capture: cannot open device

    - by Randagio
    I'm quite new to Linux (Lubuntu 12.04 for sake of precision) and ALSA programming at all. I'm trying to write a C program to capture audio from internal PC microphone for processing it. So as first step I google a bit and I found this article for capturing audio samples A tutorial on using the ALSA Audio API but when I compile it and execute it with: ./capture "default" or ./capture "hw:0,0" and all the possible variants on theme it always raises the error: cannot open device hw:0,0 (no such file or directory). So the issue is: what is the name of the mic audio device to pass as parameter to record the audio from mic ? The mic is working ok because the Sound Recorder program records sounds perfectly and I can playback them. The output of the aplay -l is the following : **** List of PLAYBACK Hardware Devices **** card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 0: Intel ICH [Intel 82801DB-ICH4] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 4: Intel ICH - IEC958 [Intel 82801DB-ICH4 - IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 and this is the amixer output (cut) Simple mixer control 'Master',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 4 [13%] [-40.50dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [on] Front Right: Playback 31 [100%] [12.00dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 22 [71%] [-1.50dB] [on] Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Stereo Mic',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] so for aplay it seems I have no recording device, but for amixer I've got the mic, a mic boost and mic stereo as well with all those gorgeous stuffs on their place !!. If so, how could my Sound Recorder record the audio without any problem at all ?!?! For sure I'm giving the wrong device name to the command line for capturing audio but I'm loosing the hope for finding the correct one ! Please help....before I tear my hair out !!!

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  • HDMI sound gone, can't figure out how to turn it back on

    - by Oli
    I have had an Acer Revo box as a media centre for a while. I recently installed Ubuntu Server (10.10) on it and polished it up with nodm (one of the most simple ways to launch an X session) and installed boxee. It's been working fine for over a month. It's just running ALSA. I've had problems with PulseAudio/Boxee/HDMI before so I wanted to keep it simple. And that worked. It pushed both PCM and digital (AAC and various Dolby codecs) over HDMI perfectly. But I restarted it the other day after mucking around with some nfs configuration and now there isn't any sound. The hardware is an ION chipset. Nvidia 9400M graphics with Nvidia MCP79/7A audio. One thing I have noticed is there doesn't appear to be any sign of a IEC958 device. A traditional fix in the past for fresh installs has been to load alsamixer, find the IEC device and toggle its mute but I can't. I'm certain this used to represent the HDMI output. It just doesn't seem to exist any more unless I run sudo alsa-utils restart while boxee is running, when I see it in an error message: * Shutting down ALSA... [ OK ] * Setting up ALSA... * warning: 'alsactl restore' failed with error message 'alsactl: set_control:1388: Cannot write control '2:0:0:IEC958 Playback Default:0' : Operation not permitted'... ...done. When nodm (and thus boxee) aren't running, I don't see this error but alsamixer still doesn't show the IEC channel. aplay -l gives: card 0: NVidia [HDA NVidia], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 0/1 Subdevice #0: subdevice #0 Its section in lshw reads: *-multimedia description: Audio device product: MCP79 High Definition Audio vendor: nVidia Corporation physical id: 8 bus info: pci@0000:00:08.0 version: b1 width: 32 bits clock: 66MHz capabilities: pm bus_master cap_list configuration: driver=HDA Intel latency=0 maxlatency=5 mingnt=2 resources: irq:22 memory:fae78000-fae7bfff I was running on the stock PAE kernel but now it's running on 2.6.37.1. I upgraded to see if that fixed things; it didn't. I'm considering a reinstall but I hate doing that because a) there's a bit of custom configuration in getting X and Boxee to start on boot and b) I don't know what the problem is. If I reinstall this time, I'll end up doing that every time the sound breaks. I love Ubuntu but I don't want to install it once a month. Is there any way to forcibly reset all alsa settings and restart from scratch (without doing a reinstall)? Any other tips? If you need more information, just ask.

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  • 5.1 surround sound

    - by rocker9455
    Ok, So i've always had trouble with enabling 5.1 in ubuntu. Running 'alsamixer': I have: Master, Heaphones, PCM, Front, Front Mi, Front Mi, Surround, Center All are at 100% Card:HDA Intel Chip:Realtek ALC888 (This is my onboard sound, Its a dell studio, with 7.1 integrated sound) Running "speaker-test -c6 -twav" I only get the front 2 speakers (Right/Left) making any noise. The others make no noise at all. I have no other sound card to use as all my PCI slots are used up. Daemon.conf: ; daemonize = no ; fail = yes ; allow-module-loading = yes ; allow-exit = yes ; use-pid-file = yes ; system-instance = no ; enable-shm = yes ; shm-size-bytes = 0 # setting this 0 will use the system-default, usually 64 MiB ; lock-memory = no ; cpu-limit = no ; high-priority = yes ; nice-level = -11 ; realtime-scheduling = yes ; realtime-priority = 5 ; exit-idle-time = 20 ; scache-idle-time = 20 ; dl-search-path = (depends on architecture) ; load-default-script-file = yes ; default-script-file = ; log-target = auto ; log-level = notice ; log-meta = no ; log-time = no ; log-backtrace = 0 resample-method = speex-float-1 ; enable-remixing = yes ; enable-lfe-remixing = no flat-volumes = no ; rlimit-fsize = -1 ; rlimit-data = -1 ; rlimit-stack = -1 ; rlimit-core = -1 ; rlimit-as = -1 ; rlimit-rss = -1 ; rlimit-nproc = -1 ; rlimit-nofile = 256 ; rlimit-memlock = -1 ; rlimit-locks = -1 ; rlimit-sigpending = -1 ; rlimit-msgqueue = -1 ; rlimit-nice = 31 ; rlimit-rtprio = 9 ; rlimit-rttime = 1000000 ; default-sample-format = s16le ; default-sample-rate = 44100 ; default-sample-channels = 6 ; default-channel-map = front-left,front-right default-fragments = 8 default-fragment-size-msec = 10

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  • Mplayer no sound when playing some movies

    - by Ivan Peevski
    Ok, that's a bit of a strange problem, that somehow crept into my system. It used to work fine. Here is the problem as far as I can identify it. When I try to play certain video files with mplayer, there is no sound. As far as I can tell, it is only an issue with ac3 and dts sound tracks (using the ffmpeg decoder). Mplayer says: ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 48000 Hz, 6 ch, s16le, 1536.0 kbit/33.33% (ratio: 192000->576000) Selected audio codec: [ffdca] afm: ffmpeg (FFmpeg DTS) ========================================================================== [AO_ALSA] Playback open error: Device or resource busy Failed to initialize audio driver 'alsa' Could not open/initialize audio device -> no sound. Audio: no sound (similar with ac3 sound, but using the ffac3 audio codec). Trying different audio output (-ao oss/pcm/sdl) doesn't fix the problem. The strange thing is that if I play these files directly with ffplay, they work fine. mplayer sound with mp3/ogg is fine My alsa configuration is standard (no /etc/asound.conf or ~/.asound*) OS: Linux Gentoo Mplayer: 1.0_rc4_p20100213 (SVN-r30554-4.3.4) FFMpeg: 0.5_p20601-r1 (SVN-r20601) Any other information I can provide?

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  • No digital audio output with Asus Xonar DG

    - by Lunatik
    I've purchased an Asus Xonar DG as replacement for faulty onboard audio in a Medion 8822 as it has an optical output which is all I really need to feed my HTPC. I uninstalled the previous drivers/devices, switched the PC off, inserted the Asus card, powered up, disabled the onboard audio in the BIOS, then installed the driver that came on the CD (same version as on Asus' website as of today) and everything went perfectly - no errors. I set the audio devices up in Windows and in the Asus utility (SPDIF enabled, 6-ch audio) as I would expect to see them work, but the only thing is I have no digital audio from test tones within Windows/the Asus utility, PCM audio or Dolby Digital from DVD. Analogue audio is fine. I've uninstalled things and reinstalled a couple of times now, as well as trying almost all combinations of analogue/digital outputs but can't get it sorted. Does anyone have any tips on how to get this working? This card has just been released so there isn't much out there to go on. Notes: The light on the toslink port is lit. OS is Vista 32-bit SP2 and all up to date, pretty much a fresh install with almost no 3rd party applications installed This page seems to suggest that a digital output device in Windows is not needed with Xonar cards as it was with the previous Realtek so I have it set to Analog. The only other output device is S/PDIF pass-thru

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  • No digital audio output with Asus Xonar DG

    - by Lunatik
    I've purchased an Asus Xonar DG as replacement for faulty onboard audio in a Medion 8822 as it has an optical output which is all I really need to feed my HTPC. I uninstalled the previous drivers/devices, switched the PC off, inserted the Asus card, powered up, disabled the onboard audio in the BIOS, then installed the driver that came on the CD (same version as on Asus' website as of today) and everything went perfectly - no errors. I set the audio devices up in Windows and in the Asus utility (SPDIF enabled, 6-ch audio) as I would expect to see them work, but the only thing is I have no digital audio from test tones within Windows/the Asus utility, PCM audio or Dolby Digital from DVD. Analogue audio is fine. I've uninstalled things and reinstalled a couple of times now, as well as trying almost all combinations of analogue/digital outputs but can't get it sorted. Does anyone have any tips on how to get this working? This card has just been released so there isn't much out there to go on. Notes: The light on the toslink port is lit. OS is Vista 32-bit SP2 and all up to date, pretty much a fresh install with almost no 3rd party applications installed This page seems to suggest that a digital output device in Windows is not needed with Xonar cards as it was with the previous Realtek so I have it set to Analog. The only other output device is S/PDIF pass-thru

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  • Looking for an application to record audio and video on a linux "embedded" device

    - by Luke404
    I am working with a linux x86 device with limited CPU resources (as a prototype we just use a pentium-m netbook). We'd like to record video from one V4L2 device (we'll probably end up using just USB Video Class devices like all modern webcams) and one audio stream from an ALSA source. The thing will not have screen and keyboard, and obviously no X11 environment. Goals are: do as little work as possible to cope with little cpu resources - for example I'd like to record video in the native MJPEG I get out of the UVC devices encoding audio to MPEG3 Layer-2 (aka mp2) is ok since it let us save a lot of space (compared to raw pcm samples) and does use little cpu power I don't mind loosing some video frames here and there (UVC devices do that) as long as I can get audio and video streams syncronized not require user input to start the thing (a python script takes care of initialization, startup, shutdown, etc...) be able to open the resulting files for postprocessing without too much effort (ie, if mplayer or vlc can play it, it's fine) So far the only app I found that could be started from command line and record V4L2 video + ALSA audio is mencoder but I'm having some difficulties with it. It should be able to do that but I cannot record audio and video together - just one of the two. And if I use two different processes to record to two different files I have no means to get them in sync (audio is more or less always correct, but video framerate will vary over time and it seems to lack timestamps to correctly play it back to the correct time). Long story short, how do you record an unconverted MJPEG stream (from an UVC device) and an audio stream (from an ALSA device, possibly encoding to any standard format) using a command line tool, to a single file (MPEG or any other container), keeping audio and video in sync?

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  • Is ffmpeg incorrectly interpreting .aif files?

    - by marue
    Being on an Ubuntu 10.04 server i installed the ffmpeg packages with apt. ffmpeg is working afterwards, and doing as it should. Almost. For testing purposes i uploaded a few audiofiles. One of them, an aif file, is not being correctly interpreted. While on my workhorse (Mac SnowLeopard) ffmpeg tells the format as Stream #0.0: Audio: pcm_s24be, 44100 Hz, 2 channels, s32, 2116 kb/s my Ubuntu server says it is: Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s which is the wrong bitdepth. Ubuntu then fails to convert the file with the error message [pcm_s24be @ 0xcd4b580]invalid PCM packet Error while decoding stream #0.0 which certainly is not true. The file is perfectly valid. Are there any know issues for ffmpeg interpreting the aif format? How can i find out which version of the aif-codec ffmpeg is using? Any ideas how to approach this issue? ffprobe output: FFprobe version SVN-r20090707, Copyright (c) 2007-2009 Stefano Sabatini libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 built on Jan 20 2010 00:13:01, gcc: 4.4.3 20100116 (prerelease) Input #0, aiff, from 'testfile.aif': Duration: 00:00:04.00, start: 0.000000, bitrate: 2117 kb/s Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s update 2: Forcing the conversion with -sample_fmt s32 doesn't change anything. Strange thing is: Even without using -sample_fmt s32 i just realized that the conversion is working and creates valid audiofiles. There just is the error message from above.

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  • This .mpg video clip doesn't play well

    - by Roey
    I've installed K-lite mega codec pack v6.9.0 with playback essentials without player. My default and only media player is windows media player. here are the clip's media info: General Complete name : D:\Users\Roey\Downloads\B384MV.mpg Format : MPEG-PS File size : 273 MiB Duration : 4mn 59s Overall bit rate : 7 643 Kbps Video ID : 224 (0xE0) Format : MPEG Video Format version : Version 2 Format profile : Main@High Format settings, BVOP : No Format settings, Matrix : Default Format settings, GOP : M=1, N=15 Duration : 4mn 57s Bit rate mode : Variable Bit rate : 7 363 Kbps Nominal bit rate : 9 000 Kbps Width : 1 920 pixels Height : 1 080 pixels Display aspect ratio : 16:9 Frame rate : 25.000 fps Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Compression mode : Lossy Bits/(Pixel*Frame) : 0.142 Stream size : 261 MiB (96%) Audio ID : 192 (0xC0) Format : MPEG Audio Format version : Version 1 Format profile : Layer 3 Mode : Joint stereo Duration : 4mn 59s Bit rate mode : Constant Bit rate : 128 Kbps Channel(s) : 2 channels Sampling rate : 44.1 KHz Compression mode : Lossy Stream size : 4.56 MiB (2%) Menu When I play it there is no sound (just a little "kahhhh" noise every 10-20 seconds) and the frames are moving very slow - it "jumps" frames. A blue tray icon [FFa] "ffdshow audio decoder" pops with the following details: Input:MP3, stereo, 44100 Hz (libavocodec) Output:PCM, stereo, 44100 Hz, 16-bit integer Any help will be much appreciated. Thanks

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  • Codecs, Premiere Pro & Quicktime: Import or Play Error

    - by Nchpmn
    Original Question I've been using a FS-H200 (not the Pro variant) recorder with a JVC ProHD camera. I have been shooting with the DTE FORMAT to Quicktime (.mov). I copied the files to an external hard drive and am now trying to edit. The files will play back in VLC, as they would be expected to. However they will not import into Adobe Premiere CS5.5, instead giving an error: Unsupported format or damaged file. Quicktime gives the following error when attempting to play the files: Error -2002: a bad public movie atom was found in the movie (Filename) To try and fix this, I have installed the following codec packs: K-Lite Codec Pack 64-bit Full (version 5.9, latest) K-Lite Codec Pack 32-bit Full (version 8.4, latest) MainConcept Codec Suite (Broadcast) v5.1 for Adobe CS5 Reinstalled Quicktime with new download from Apple The same errors and problems still exist. From this I can assume that there is an issue with Quicktime and that is what Premiere is using as an encoder/decoder for the codec. Is there any way to fix this? From looking at the "Codec Information" from VLC: Stream 0 Type: Video Codec: MPEG-1/2 (mpgv) Language: English Resolution: 1280 x 720 Frame Rate: 25 Stream 1 Type: Audio Codec: PCM S16 BE (twos) Language: English Channels: Stereo Sample Rate: 48000 Hz Bits per sample: 16 Other computer specs: Windows 7 Professional 64-bit (SP1) Gigabyte Z68X-UD3-B3 Intel i7-2600K 16GB DDR3 2TB WD 7200RPM SATA 6Gb/s LaCie d2 Quadra 2TB v3 7200RPM (External HDD) NVIDIA GeForce GTX 560 Ti Golden Sample Updates 2012-03-11 @ 2050 AEDT MPEG Steamclip doesn't recognise, play or convert the footage. File open error: unrecognised file type. [Open Anyway] File open error: can't find video or audio tracks. 2012-03-24 @ 1920 AEDT Had to transcode the footage. :(

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  • DVI splitter not working as expected/confusion between DVI-D and -I

    - by Freakishly
    Hey guys, thanks for looking. I have an ATI FirePro™ V3700 in my desktop machine, and I have been running a dual-monitor setup quite effortlessly, thanks to the two DVI ports on the card. I came upon a third monitor, and wanted to extend my desktop to 3 screens, so I purchased a DVI splitter from Amazon. Now, I can only duplicate the second monitor onto the third, not extend it. I've tried all possible combinations of input to no avail. Here's the setup: The ATI FirePro™ V3700 has two Dual-Link DVI-I outputs The splitter splits a single Dual-Link DVI-I port into two Dual-Link DVI-I outputs Two of the monitors are NEC E222W, and the third monitor is a Dell 2001FP. Each monitor has one D-Sub and one Dual-Link DVI-D input. Cables going from the video card to the monitors are two Dual-Link DVI-D to the NECs and one Single-Link DVI-D to the Dell. Is the problem likely with the DVI-D/DVI-I mismatch? Or is it with the cable on the Dell that is only a Single-Link? The cables are easily replaceable, the monitors not so much. Thanks for your time, I really appreciate it. http://www.amd.com/us/products/workstation/graphics/ati-firepro-3d/v3700/Pages/v3700-specs.aspx http://www.amazon.com/Cables-Unlimited-DVI-D-Splitter-PCM-2260/product-reviews/B000H09RFM/ref=dp_top_cm_cr_acr_txt?ie=UTF8&showViewpoints=1 www dot newegg dot com/Product/Product.aspx?Item=N82E16824002495 accessories dot us dot dell dot com/sna/PopupProductDetail.aspx?cs=19&l=en&c=us&sku=320-1578 Apologies for the fudged links, I'm new here and they won't let me post more than two :P

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

    - by user187532
    Hi Android experts, I am trying to play the bigger size audio wav file(which is 20 mb) using the following code(AudioTrack) on my Android 1.6 HTC device which basically has less memory. But i found device crash as soon as it executes reading, writing and play. But the same code works fine and plays the lesser size audio wav files(10kb, 20 kb files etc) very well. P.S: I should play PCM(.wav) buffer sound, the reason behind why i use AudioTrack here. Though my device has lesser memory, how would i read bigger audio files bytes by bytes and play the sound to avoid crashing due to memory constraints. private void AudioTrackPlayPCM() throws IOException { String filePath = "/sdcard/myWav.wav"; // 8 kb file byte[] byteData = null; File file = null; file = new File(filePath); byteData = new byte[(int) file.length()]; FileInputStream in = null; try { in = new FileInputStream( file ); in.read( byteData ); in.close(); } catch (FileNotFoundException e) { // TODO Auto-generated catch block e.printStackTrace(); } int intSize = android.media.AudioTrack.getMinBufferSize(8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_8BIT); AudioTrack at = new AudioTrack(AudioManager.STREAM_MUSIC, 8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_8BIT, intSize, AudioTrack.MODE_STREAM); at.play(); at.write(byteData, 0, byteData.length); at.stop(); at.release(); } Could someone guide me please to play the AudioTrack code for bigger size wav files?

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

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

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  • What can cause my code to run slower when the server JIT is activated?

    - by durandai
    I am doing some optimizations on an MPEG decoder. To ensure my optimizations aren't breaking anything I have a test suite that benchmarks the entire codebase (both optimized and original) as well as verifying that they both produce identical results (basically just feeding a couple of different streams through the decoder and crc32 the outputs). When using the "-server" option with the Sun 1.6.0_18, the test suite runs about 12% slower on the optimized version after warmup (in comparison to the default "-client" setting), while the original codebase gains a good boost running about twice as fast as in client mode. While at first this seemed to be simply a warmup issue to me, I added a loop to repeat the entire test suite multiple times. Then execution times become constant for each pass starting at the 3rd iteration of the test, still the optimized version stays 12% slower than in the client mode. I am also pretty sure its not a garbage collection issue, since the code involves absolutely no object allocations after startup. The code consists mainly of some bit manipulation operations (stream decoding) and lots of basic floating math (generating PCM audio). The only JDK classes involved are ByteArrayInputStream (feeds the stream to the test and excluding disk IO from the tests) and CRC32 (to verify the result). I also observed the same behaviour with Sun JDK 1.7.0_b98 (only that ist 15% instead of 12% there). Oh, and the tests were all done on the same machine (single core) with no other applications running (WinXP). While there is some inevitable variation on the measured execution times (using System.nanoTime btw), the variation between different test runs with the same settings never exceeded 2%, usually less than 1% (after warmup), so I conclude the effect is real and not purely induced by the measuring mechanism/machine. Are there any known coding patterns that perform worse on the server JIT? Failing that, what options are available to "peek" under the hood and observe what the JIT is doing there?

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  • Performance: float to int cast and clipping result to range

    - by durandai
    I'm doing some audio processing with float. The result needs to be converted back to PCM samples, and I noticed that the cast from float to int is surprisingly expensive. Whats furthermore frustrating that I need to clip the result to the range of a short (-32768 to 32767). While I would normally instictively assume that this could be assured by simply casting float to short, this fails miserably in Java, since on the bytecode level it results in F2I followed by I2S. So instead of a simple: int sample = (short) flotVal; I needed to resort to this ugly sequence: int sample = (int) floatVal; if (sample > 32767) { sample = 32767; } else if (sample < -32768) { sample = -32768; } Is there a faster way to do this? (about ~6% of the total runtime seems to be spent on casting, while 6% seem to be not that much at first glance, its astounding when I consider that the processing part involves a good chunk of matrix multiplications and IDCT) EDIT The cast/clipping code above is (not surprisingly) in the body of a loop that reads float values from a float[] and puts them into a byte[]. I have a test suite that measures total runtime on several test cases (processing about 200MB of raw audio data). The 6% were concluded from the runtime difference when the cast assignment "int sample = (int) floatVal" was replaced by assigning the loop index to sample. EDIT @leopoldkot: I'm aware of the truncation in Java, as stated in the original question (F2I, I2S bytecode sequence). I only tried the cast to short because I assumed that Java had an F2S bytecode, which it unfortunately does not (comming originally from an 68K assembly background, where a simple "fmove.w FP0, D0" would have done exactly what I wanted).

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  • trying to center a css menu made on purecssmenu.com

    - by Party Mcfly
    i have a menu that i generated on purecssmenu.com and im having trouble trying to center it on my page here is the code... <!-- Start PureCSSMenu.com STYLE --> <style> #pcm{display:none;} ul.pureCssMenu ul{display:none} ul.pureCssMenu li:hover>ul{display:block} ul.pureCssMenu ul{position: absolute;left:-1px;top:98%;} ul.pureCssMenu ul ul{position: absolute;left:98%;top:-2px;} ul.pureCssMenu,ul.pureCssMenu ul { margin:0px; list-style:none; padding:0px 2px 2px 0px; background-color:#000000; background-repeat:repeat; border-color:#000000; border-width:1px; border-style:solid; } ul.pureCssMenu table {border-collapse:collapse}ul.pureCssMenu { display:block; zoom:1; float: left; } ul.pureCssMenu ul{ width:80.85000000000001px; } ul.pureCssMenu li{ display:block; margin:2px 0px 0px 2px; font-size:0px; } ul.pureCssMenu a:active, ul.pureCssMenu a:focus { outline-style:none; } ul.pureCssMenu a, ul.pureCssMenu li.dis a:hover, ul.pureCssMenu li.sep a:hover { display:block; vertical-align:middle; background-color:#000000; border-width:1px; border-color:#000000; border-style:solid; text-align:left; text-decoration:none; padding:2px 5px 2px 10px; _padding-left:0; font:11px Arial; color: #969696; text-decoration:none; cursor:default; } ul.pureCssMenu span{ overflow:hidden; } ul.pureCssMenu li { float:left; } ul.pureCssMenu ul li { float:none; } ul.pureCssMenu ul a { text-align:left; white-space:nowrap; } ul.pureCssMenu li.sep{ text-align:left; padding:0px; line-height:0; height:100%; } ul.pureCssMenu li.sep span{ float:none; padding-right:0; width:3px; height:100%; display:inline-block; background-color:#cccccc #111111 #111111 #cccccc; background-image:none;} ul.pureCssMenu ul li.sep span{ width:100%; height:3px; } ul.pureCssMenu li:hover{ position:relative; } ul.pureCssMenu li:hover>a{ background-color:#000000; border-color:#000000; border-style:solid; font:11px Arial; color: #ffa500; text-decoration:none; } ul.pureCssMenu li a:hover{ position:relative; background-color:#000000; border-color:#000000; border-style:solid; font:11px Arial; color: #ffa500; text-decoration:none; } ul.pureCssMenu li.dis a { color: #666 !important; } ul.pureCssMenu img {border: none;float:left;_float:none;margin-right:2px;width:16px; height:16px; } ul.pureCssMenu ul img {width:16px; height:16px; } ul.pureCssMenu img.over{display:none} ul.pureCssMenu li.dis a:hover img.over{display:none !important} ul.pureCssMenu li.dis a:hover img.def {display:inline !important} ul.pureCssMenu li:hover > a img.def {display:none} ul.pureCssMenu li:hover > a img.over {display:inline} ul.pureCssMenu a:hover img.over,ul.pureCssMenu a:hover ul img.def,ul.pureCssMenu a:hover a:hover img.over{display:inline} ul.pureCssMenu a:hover img.def,ul.pureCssMenu a:hover ul img.over,ul.pureCssMenu a:hover a:hover img.def{display:none} ul.pureCssMenu a:hover ul{display:block} ul.pureCssMenu span{ display:block; background-image:url(./images/arr_white.gif); background-position:right center; background-repeat: no-repeat; padding-right:12px;} ul.pureCssMenu li:hover>a>span{ background-image:url(./images/arrv_white.gif); } ul.pureCssMenu a:hover span{ _background-image:url(./images/arrv_white.gif)} ul.pureCssMenu ul span,ul.pureCssMenu a:hover table span{background-image:url(./images/arr_white.gif)} </style> <!-- End PureCSSMenu.com STYLE --> and here is the html, witch is probably not even needed in this posted but i figure i would include it.. i just want that menu centered inside my website. <!-- Start PureCSSMenu.com MENU --> <ul class="pureCssMenu pureCssMenum"> <li class="pureCssMenui"><a class="pureCssMenui" href="home.html" target="scare">home</a></li> <li class="pureCssMenui"><a class="pureCssMenui" href="#">about</a></li> <li class="pureCssMenui"><a class="pureCssMenui" href="#"><span>haunts</span><![if gt IE 6]></a><![endif]><!--[if lte IE 6]><table><tr><td><![endif]--> <ul class="pureCssMenum"> <li class="pureCssMenui"><a class="pureCssMenui" href="#">2009</a></li> <li class="pureCssMenui"><a class="pureCssMenui" href="#">2010</a></li> <li class="pureCssMenui"><a class="pureCssMenui" href="#">2011</a></li> <li class="pureCssMenui"><a class="pureCssMenui" href="#">2012</a></li> </ul> <!--[if lte IE 6]></td></tr></table></a><![endif]--></li> <li class="pureCssMenui"><a class="pureCssMenui" href="#">studio</a></li> <li class="pureCssMenui"><a class="pureCssMenui" href="#"><span>products</span><![if gt IE 6]></a><![endif]><!--[if lte IE 6]><table><tr><td><![endif]--> <ul class="pureCssMenum"> <li class="pureCssMenui"><a class="pureCssMenui" href="#">nightmares</a></li> <li class="pureCssMenui"><a class="pureCssMenui" href="#">hauntworks</a></li> <li class="pureCssMenui"><a class="pureCssMenui" href="#">atmosfears</a></li> <li class="pureCssMenui"><a class="pureCssMenui" href="#">frightwears</a></li> </ul> <!--[if lte IE 6]></td></tr></table></a><![endif]--></li> <li class="pureCssMenui"><a class="pureCssMenui" href="#"><span>links</span><![if gt IE 6]></a><![endif]><!--[if lte IE 6]><table><tr><td><![endif]--> <ul class="pureCssMenum"> <li class="pureCssMenui"><a class="pureCssMenui" href="#">haunts</a></li> <li class="pureCssMenui"><a class="pureCssMenui" href="#">suppliers</a></li> <li class="pureCssMenui"><a class="pureCssMenui" href="#">resources</a></li> </ul> <!--[if lte IE 6]></td></tr></table></a><![endif]--></li> <li class="pureCssMenui"><a class="pureCssMenui" href="#">contact</a></li> </ul> <a id="pcm" href="http://www.purecssmenu.com/">CSS Drop Down Menu by PureCSSMenu.com</a> <!-- End PureCSSMenu.com MENU --> thanks in advance for your time.

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  • Writing an audio player in C#

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

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  • Play Multiple iPod Library Songs On iPhone At The Same Time With Pitch Bending & Other Effects

    - by Dino
    Hi, I have been going at this for the past two weeks and it is driving me crazy. I asked this question a couple of days ago (Extract iPod Library raw PCM samples and play with sound effects) and whilst the answer got me half way there I am still stuck. Basically what I am trying to achieve is load up multiple songs from the iPod library for playback with effects such as pitch bend, eq effects etc... I have gone down the route of AVPlayer and AVAudioPlayer which are too simple. The only framework I've seen that can play audio with these effects is OpenAL. I have tried a few objective c wrappers (Finch and ObjectAL) Finch does not play compressed audio whilst ObjectAL will only convert it for me if I pass in a URL for the file (which is something I cannot do because I only have an incompatible iPod library URL). An example of an app that does what I want beautifilly is Tap DJ. It can load up songs from the iPod library fast (unlike TouchDJ and it plays them with all sorts of effects. Any help would be much appreciated.

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  • Real Time Sound Captureing J2ME

    - by Abdul jalil
    i am capturing sound in J2me and send these bytes to remote system, i then play these bytes on remote system.five second voice is capture and send to remote system. i get the repeated sound again .i am making a sound messenger please help me where i am doing wrong i am using the follown code . String remoteTimeServerAddress="192.168.137.179"; sc = (SocketConnection) Connector.open("socket://"+remoteTimeServerAddress+":13"); p = Manager.createPlayer("capture://audio?encoding=pcm&rate=11025&bits=16&channels=1"); p.realize(); RecordControl rc = (RecordControl)p.getControl("RecordControl"); ByteArrayOutputStream output = new ByteArrayOutputStream(); OutputStream outstream =sc.openOutputStream(); rc.setRecordStream(output); rc.startRecord(); p.start(); int size=output.size(); int offset=0; while(true) { Thread.currentThread().sleep(5000); rc.commit(); output.flush(); size=output.size(); if(size0) { recordedSoundArray=output.toByteArray(); outstream.write(recordedSoundArray,0,size); } output.reset(); rc.reset(); rc.setRecordStream(output); rc.startRecord(); }

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  • Virtual audio driver (microphone)

    - by Dalamber
    Hello guys, I want to develop a virtual microphone driver. Please, do not say anything about DirectShow - that's not "the way". I need a solution that will work with any software including Skype and MSN. And DirectShow doesn't fit these requirements. I found AVStream Filter-Centric Simulated Capture Driver (avssamp.sys) in Windows 7 WDK. What I need is an audio part of it. By default it reads avssamp.wav and plays it. But this driver is registered as WDM streaming capture device. And I want it in Audio Capture Device. There are some posts in the web but they are all the same: http://www.tech-archive.net/Archive/Development/microsoft.public.development.device.drivers/2005-05/msg00124.html http://www.winvistatips.com/problem-installing-avssamp-audio-capture-sources-category-t184898.html I think registering this filter-driver as audio capture device will make Skype recognize it as a microphone and thefore I will be able to push any PCM file as if it goes from mic. If someone already faced this problem before, please help. Thanks in advance.

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  • Audio Streaming Latency

    - by killianmcc
    I'm writing a UDP local area network video chat system and have got the video and audio streams working. However I'm experiencing a little latency (about half a second) in the audio and was wondering what codecs would provide the least latency. I'm using NAudio (http://naudio.codeplex.com/) which provides me access to the following codecs for streaming; Speex Narrow Band (VBR) Speex Wide Band (16kHz)(VBR) Speex Ultra Wide Band (32kHz)(VBR) DSP Group TrueSpeech (8.5kbps) GSM 6.10 (13kbps) Microsoft ADPCM (32.8kbps) G.711 a-law (64kbps) G.722 16kHz (64kbps) G.711 mu-law (64kbps) PCM 8kHz 16 bit uncompressed (128kbps) I've tried them out and I'm not noticing much difference. Is there any others that I should download and try to reduce latency? I'm only going to be sending voice over the connection but I'm not really worried about quality or background noises too much. UPDATE I'm sending the audio in blocks like so; waveIn = new WaveIn(); waveIn.BufferMilliseconds = 50; waveIn.DeviceNumber = inputDeviceNumber; waveIn.WaveFormat = codec.RecordFormat; waveIn.DataAvailable += waveIn_DataAvailable; void waveIn_DataAvailable(object sender, WaveInEventArgs e) { if (connected) { byte[] encoded = codec.Encode(e.Buffer, 0, e.BytesRecorded); udpSender.Send(encoded, encoded.Length); } }

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  • DVD playback with Windows Media Player 11 works fine, but when copied to HDD and then played back, t

    - by stakx
    I have several DVDs with short documentaries on it. Since the notebook I'm using (a Dell Latitude E6400) has only one DVD drive, and I might play back those short movies very often, I thought of copying them to the HDD and playing them back from there. However, I've run into a problem, namely stuttering audio. Problem description: When I play back these movies directly from DVD (with Windows Media Player 11 under Windows Vista), everything works fine. Smooth video, no significant audio problems (only the occasional click). But as soon as I copy any of these DVDs to the HDD and try to play them back from there (e.g. using the wmpdvd://drive/title/chapter?contentdir=path protocol, I get stuttering audio — audio playback sounds like a machine gun for a third of a second or so, approx. every 8 seconds. I have tried converting the VOB files from the DVD to another format (ie. ripping), but that resulted in a noticeable downgrade of picture quality. Therefore I thought it best to keep the files in their original format, if possible. Still, I suspect that the stuttering audio is due to some (de-)muxing problem, and that changing the file format might help. (After all, video playback is fine; therefore I don't think that the hardware is too slow for playback.) Only thing is, I don't know how to convert the VOB files to another Windows Media Player-compatible format without quality loss. I hope someone can help me, or give me further pointers on things I could try out to get HDD playback to work without the problem described. Some things I've tried so far, without any success: VOB2MPG, in order to convert the .vob file to a .mpg file. But that changes only the A/V container, not the content. No re-encoding takes place at all. Re-encoding with MPlayer/MEncoder. Lots of quality loss there, and I frankly haven't got the time to test all possible settings combinations available. Disabling all plug-ins, equalizers, etc. in Windows Media Player. Disabling all hardware acceleration on the audio playback device. Further info on the VOB files I'm trying to playback: The video format is MPEG ES, PAL 720x576 pixels @ 24/25 frames per second. The sound stream is uncompressed PCM, 16-bit stereo @ 48kHz. (Might it help if I somehow re-encoded the sound stream at a lower resolution, or as an MP3? If so, how would I do this without changing the video stream?) P.S.: I am limited to using Windows Media Player (11). (I previously tried MPlayer btw., but the video playback quality was surprisingly bad.)

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