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  • How to play audio podcast file from libsyn rss feed? (drupal)

    - by Kirk Hings
    Got an established libsyn rss feed, got a new drupal website for the podcast. Libsyn provides a player but not correct aesthetic. I can upload and play mp3 files with audio module and mp3player module, and like the mp3 player's output, a simple flash player, but I don't want to be manually moving the podcast audio files (mp3) over every week. Looked at importing automatically with Feeds, but it's not working and besides that's creating extra files unnecessarily on the drupal site. Just want to use the mp3player modulee's flash player in a drupal page, which feeds the latest mp3 file from a libsyn rss feed. Don't really need to store or play multiple episodes, just the latest episode. How would you do it?

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  • What is the best API in any language for Audio and MIDI music application development?

    - by noneme
    What, in your opinion, is the best API to utilize in developing an application that handles both realtime MIDI and audio input and output? This would be for an application that is used in the process of making music as opposed to playing audio or MIDI files. I'm aware that this may be a subjective question, but if you know of an API that is dominantly used for these purposes, please share it. I'm agnostic about which language the API is for, and I also don't care about portability. The real concern is for an API that is well documented, well designed (e.g. thought out and intuitive to developers using it), and actively maintained. OS portability would be nice, but it is second to having an API/Language that meets the previous requirements. Please note that the emphasis is not on API's for sound synthesis or for composing music with code. It is intended for the handling of sound file and MIDI data in a real-time context.

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  • Rapid spectral analysis of audio file using Python 2.6?

    - by Ephemeralis
    What I want to do is to have a subroutine that analyses every 200 milliseconds of a sound file which it is given and spits out the frequency intensity value (from 0 to 1 as a float) of a specific frequency range into an array which I later save. This value then goes on to be used as the opacity value for a graphic which is supposed to 'strobe' to the audio file. The problem is, I have never ventured into audio analysis before and have no clue where to start. I have looked pymedia and scipy/numpy thinking I would be able to use FFT in order to achieve this, but I am not really sure how I would manipulate this data to end up with the desired result. The documentation on the SpectrAnalyzer class of pymedia is virtually non-existant and the examples on the website do not actually work with the latest release of the library - which isn't exactly making my life easier. How would I go about starting this project? I am at a complete loss as to what libraries I should even be using.

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  • Safari won't request video or audio from HTML 5 media elements?

    - by thure
    So far what I've been developing has worked in Chrome and, using fallbacks, IE8. What I don't get is this: Safari just won't start loading <video> or <audio> content. Safari 6 won't load, and neither will iOS 5's Safari: My code calls .load() on the elements at the appropriate time (at least for Chrome), so what gives? Here is the video declaration: <video width="800" height="600" class="faces" id="facesVideo"> <source src="video/grid.mp4" type="video/mp4" /> <source src="video/grid.ogv" type="video/ogg" /> </video> The audio is declared dynamically, but has the same problem. Do I need to wait for some DOM event that Chrome doesn't need before calling .load()? What does it take to get Safari to start buffering until the elements can fire canplaythrough?

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  • Detecting when HTML5 audio is finished playing (more than once)?

    - by user386911
    I am having a problem detecting when an tag is finished playing an mp3. When I do something like this: myAudio.addEventListener("ended", function() { alert("ended"); }); It only occurs the first time the audi is played. When I play the audio again, nothing happens. The same thing occurs when I use the onended=doThis(); method. I've heard maybe there is a way to do it in jquery, but I haven't been able to get it to work. I've also heard there might be a way to fix it by changing the audio div id everytime the mp3 is played, but this doesn't work for me because I need the id to stay the same. Anyone got any ideas?

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  • Mini-DisplayPort to HDMI Adapter Sound Stopped Working

    - by jimdrang
    I have cancelled cable and want to watch the NCAA tournament games on my TV tonight through my 2011 Macbook Pro where I can stream the game in a browser. I have a cheap Mini-DisplayPort to HDMI converter that I have connected to my TV in the past and had no issues with audio or video, the problem is the audio has stopped working since the last time I used it a few months ago and now just keeps playing through the laptop speakers, but the video works fine. Everything with my setup is the same and when I try to force the audio output to the TV in the Audio system settings, my TV is not listed as an output option at all. I have tried various combinations of power cycling, replugging-in both devices and making sure the TV options are set properly to receive audio through HDMI but no luck. Anyone know what the issue could be?

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  • Joining H264 *without* re-encoding

    - by jdmuys
    Hi, I have two halves of a single show in two .MP4 files, encoded in H264. I would like to join them without re-encoding. Is this possible? I managed to create a joined video as a Quicktime file (.mov) using Quicktime Pro, but then Quicktime Pro will not convert it back to .MP4 without re-encoding. This may be because looking inside the .mov file, the two H264 videos are in there still separated as individual "objects". I am also struggling with MPEG StreamClip without reaching a real solution. But I may have missed something. Note that I have the same issue with MPEG2 files. I can export them to a .MPEG container or a .TS file for example, but I don't know how to join them without re-encoding. Any suggestion welcome, preferably using Mac software.

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  • Command line import of database using latin1 encoding

    - by chrisjlee
    I'm using a particular cloud hosting solution (one which i won't name) and they don't provide ssh access so i'm at a whim on how the database is dumped. I downloaded the dump which is packed into a tar.gz file. I discover that this file utilizes latin1 encoding. Which i don't get to specify the encoding for the host i'm using because i don't have SSH access or DB access. I try to import it via command line for my local development environment (mysql -uroot foodb < file.db) like i usually do with other databases but am having problems. Is it possible to import a database via command line by specifying which encoding (preferably latin1) before importing it? Or do i have to convert it to UTF8?

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  • Sound problems in Windows 7

    - by Plastkort
    hi! I hope I reached the correct forum for this question. I have a media computer with a third party audio card (Soundblaster audigy SE) I use a coaxial digital audio cable connected to a Onkyo TX SR508. if I use normal audio, the sound seems very low, I have to set volume atleast to 62 in my amplifier to hear anything, however if I set digital SPDIF I have no means of controlling the audio volume from the PC (only from the amplifier) and this can be nasty if I toggle between movies that uses Digital AC3/THX and movies wihtouth, if I look at movies capable of AC3 the volume gets VERY loud if amp is set to 62, 32 is more than enough volume when using passthrough. so this bothers me is how can I get the same amount of volume with or without digital output? I tried also other soundcards, internal red light Digital audio cable... if I connect to my television I get ok sound on any sound source via HDMI... help :)

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  • Joining H264 *without* re-encoding

    - by jdmuys
    I have two halves of a single show in two .MP4 files, encoded in H264. I would like to join them without re-encoding. Is this possible? I managed to create a joined video as a Quicktime file (.mov) using Quicktime Pro, but then Quicktime Pro will not convert it back to .MP4 without re-encoding. This may be because looking inside the .mov file, the two H264 videos are in there still separated as individual "objects". I am also struggling with MPEG StreamClip without reaching a real solution. But I may have missed something. Note that I do not have the same issue with MPEG2 files. I can export them to a .MPEG container or a .TS file for example, and then I can join them without re-encoding using MPEG Streamclip. Any suggestion welcome, preferably using Mac software.

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  • AWS RDS Mysql with benstalk Hibernate app: Character encoding issue

    - by TeraTon
    I'm running a webapp from amazon rds with tomcat 7 and spring, which uses hibernate as the persistence layer. The application and utf-8 encoding work properly on localhost, but for some reason when I deploy to amazon, the UTF-8 encoding breaks. I use mysql 5.5.27 on amazon rds and the table that we wish to update has collation set to utf8 - utf8_unicode_ci And in hibernate I have set: < prop key="hibernate.connection.charSet"UTF-8 UTF-8 characters get replaced by ??? and this is of course especially bad for passwords and usernames + email as it basically kills them. Anyone else encountered character encoding breaking when deploying to amazon?

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  • Why does IIS not support chunked transfer encoding?

    - by Graeme Perrow
    I am making an HTTP connection to an IIS web server and sending a POST request with the data encoded using Transfer-Encoding: chunked. When I do this, IIS simply closes the connection, with no error message or status code. According to the HTTP 1.1 spec, All HTTP/1.1 applications MUST be able to receive and decode the "chunked" transfer-coding so I don't understand why it's (a) not handling that encoding and (b) it's not sending back a status code. If I change the request to send the Content-Length rather than Transfer-Encoding, the query succeeds, but that's not always possible. When I try the same thing against Apache, I get a "411 Length required" status and a message saying "chunked Transfer-Encoding forbidden". Why do these servers not support this encoding?

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  • Encoding in python with lxml - complex solution

    - by Vojtech R.
    Hi, I need to download and parse webpage with lxml and build UTF-8 xml output. I thing schema in pseudocode is more illustrative: from lxml import etree webfile = urllib2.urlopen(url) root = etree.parse(webfile.read(), parser=etree.HTMLParser(recover=True)) txt = my_process_text(etree.tostring(root.xpath('/html/body'), encoding=utf8)) output = etree.Element("out") output.text = txt outputfile.write(etree.tostring(output, encoding=utf8)) So webfile can be in any encoding (lxml should handle this). Outputfile have to be in utf-8. I'm not sure where to use encoding/coding. Is this schema ok? (I cant find good tutorial about lxml and encoding, but I can find many problems with this...) I need robust approved solution so I ask you seniors. Many thanks

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  • Check the encoding of text in SQlite

    - by JJG
    I'm having a nightmare dealing with non Eurpean texts in SQlite. I think the problem is that SQlite isn't encoding the text in UTF8. So I want to check what the encoding is, and hopefully change it to utf8. I encoded a CSV in UTF8 and simply imported it to SQlite but the non-roman text is garbled. I would like to know: 1)how to check the encoding. 2)How to change the encoding if it is not utf8. I've been reading about Pragma encoding, but I'm not sure how to use this.

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  • Front audio jack not working or missconfigured?

    - by Nicholas
    I have win xp and asus p5ql-e motherboard. The problem is that when i plug in my headphones on the front audio jack, the computer doesn't recognizes it. They work on the back audio jack but not on the front - how can i be sure that it;s not a software problem (something miss configured or not configured at all), before i conclude that it's a hardware problem (broken front audio jack or miss connected to the motherboard)?

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  • Is there a way to "burn" audio to an ISO? (as an audio CD)

    - by Sootah
    I have an audiobook that I've downloaded via their download manager, and it's loaded into their cutesy little audio program that they force you to use. I can play the book just fine using their proprietary software, and while it's annoying when using my PC, it's utterly UNBEARABLE when I try to listen to it on my Blackberry. The program is INSANELY slow, it literally takes around 30 seconds to switch between tracks, so if I've forgotten where I am in the book it takes me around 15 minutes to finally get to where I was at. I've looked everywhere on how to transcode the book to .MP3, but evidently with their current format it's either extremely convoluted (and I have no desire to dick around with installing some older version of the codec, getting a different transcoding app, and then wrestling with getting it to actually work). Since I'm able to burn a copy of the book to an audio CD, I figure the best way to go about this is to just make the CDs and then rip them off of those to .MP3. In order to avoid wasting two hours, not to mention 14 CD-R's, I was wondering if there's a way to "burn" to an .ISO instead of an actual CD-R. I currently have SlySoft's Virtual CloneDrive installed, so I can mount .ISO's easily enough, but now I want to actually create an ISO via the CD burning process. Just in case I've not explained myself very well, here is an overview of what I intend to do: "Burn" a set of Audio CD .ISOs from the audiobook (hopefully I can do this using Windows Media Player, otherwise I'll be forced to use the audiobook app) Mount an .ISO in Virtual CloneDrive Rip the audio tracks on the mounted .ISO to .MP3s Repeat steps 2-3 until the entire book is in .MP3 format Copy .MP3s to my Blackberry so that I'm not driven insane every time I want to listen to the book in the car, and be able to use Winamp when listening on my computer EDIT: I'd suppose a rather concise way to put it is that I need something that will emulate a CD-R drive, so that you can select it as the output drive in whatever app your burning the audio CD from. (I'd suppose that when you "insert a blank CD-R" the app would then ask you what file to save to)

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  • No audio from USB device on iMac

    - by brandon
    My computer froze the other day and started a buzzing noise. After I restarted in the middle of it, the buzzing persisted. A few restarts later, the buzzing stopped, but I can't get audio to come from my audio interface setup anymore. I'm using an M-Audiophile USB to connect to studio monitors, and there's absolutely no audio. I've tried restarting many times and unplugging - still nothing. Any help would be appreciated.

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  • Flash game size and distribution between asset types

    - by EyeSeeEm
    I am currently developing a Flash game and am planning on a large amount of graphics and audio assets, which has led to some questions about Flash game size. By looking at some of the popular games on NG, there seem to be many in the 5-10Mb and a few in the 10-25Mb range. I was wondering if anyone knew of other notable large-scale games and what their sizes were, and if there have been any cases of games being disadvantaged because of their size. What is a common distribution of game size between code, graphics and audio? I know this can vary strongly, but I would like to hear your thoughts on an average case for a high-quality game.

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  • Lubuntu upgrade to 13.04 killed sound with ALSA. How to troubleshoot?

    - by Sven
    After upgrading to 13.04 from 12.10 Lubuntu lost audio playback after unplugging usb soundcard (Polycom) and plugging it back in. Volume control was gray and leading to pulseaudio mixer (not installed) so I uninstalled the pulseaudio package. I also removed and reinstalled the alsa-base package. After restart I have the alsamixer back everything seemingly as usual(volume 100%, unmute) but every sound program gets me errors no matter what device I select. aplay -L: null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default:CARD=NVidia HDA NVidia, ALC662 rev1 Analog Default Audio Device sysdefault:CARD=NVidia HDA NVidia, ALC662 rev1 Analog Default Audio Device front:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Digital IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output dmix:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Direct sample mixing device dmix:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Direct sample mixing device dmix:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct sample mixing device dsnoop:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Direct sample snooping device dsnoop:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Direct sample snooping device dsnoop:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct sample snooping device hw:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Direct hardware device without any conversions hw:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Direct hardware device without any conversions hw:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct hardware device without any conversions plughw:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Hardware device with all software conversions plughw:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Hardware device with all software conversions plughw:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Hardware device with all software conversions default:CARD=Communicator Default Audio Device sysdefault:CARD=Communicator Default Audio Device front:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio Front speakers surround40:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio IEC958 (S/PDIF) Digital Audio Output dmix:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio Direct sample mixing device dsnoop:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio Direct sample snooping device hw:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio Direct hardware device without any conversions plughw:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio Hardware device with all software conversions etc/asound.conf: defaults.ctl.card 1 defaults.pcm.card 1 defaults.pcm.device 1 Following gets same result with both devices. aplay -vv -D front:CARD=NVidia,DEV=0 "Release the Pressure.wav": Playing WAVE 'Release the Pressure.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono aplay: set_params:1087: Channels count non available Guayadeque mp3 playback: AL lib: alsa_open_playback: Could not open playback device 'default': No such file or directory 21:32:14: Error: Gstreamer error 'Configured audiosink playbackbin is not working.' Audacious: ALSA error: snd_mixer_attach failed: No such file or directory. ALSA error: snd_pcm_open failed: No such device. So How do I fix my audio? UPDATE: I removed the usb soundcard and got rid of all alsa config. Everything is working as before the install but it sure feels fragile.

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  • /etc/init.d Character Encoding Issue

    - by Ryan Rosario
    I have a script in /etc/init.d on an EC2 image that, on machine startup, pulls in source code via SVN, builds it, and then runs it using Ant. The source code is Java. Within this code is a call to the Weka library which writes a file to disk. On most Ubuntu AMIs, and my home machines' versions of Ubuntu, there is no issue. The problem is that with certain versions/AMIs of Ubuntu, Unicode characters in the file are replaced with question marks ('?'). If I run the job manually on the trouble instance, Unicode is output to file correctly, but not when run from /etc/init.d. What might be causing this problem and how can I fix it so that Unicode characters appear correctly in files written from /etc/init.d processes?

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  • How to automatically change volume level when un-/plugging headphones?

    - by htorque
    What I want is the following: When I plug in my headphones, I want the sound to be un-muted and set to a specific volume level. When I unplug my headphones, I want the sound to be muted (or set to a specific volume level). Setting the volume levels isn't the problem, but I somehow need to do this when un-/plugging the headphones, so I'm looking for a way to get notified of those events. I quickly found /proc/asound/card0/codec#0 to indicate whether headphones are plugged in or not, so I tried to monitor it using inotifywait and change the volume level based on modified notifications. Unfortunately inotifywait failed because proc isn't an ordinary filesystem. Are there other ways to do this (maybe via PulseAudio)? Audio device: Intel HDA, audio codec: Conexant CX20585.

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  • Optimal Compression for Speech

    - by ashes999
    I'm designing a game that depends heavily on audio; I will have some 300+ speech files (most of them just a word or two long). This can very quickly escalate the size of my final game. What's the optimal way to encode/compress speech files to keep the size minimal without getting audio artifacts? Please address both per-file compression/encoding, and also zipping/compressing the set of all speech files together in your answer. Because I'm not sure which (or combination of both) factors will give me the best results. Edit: I need this to run in Silverlight and Android, so I'm presumably stuck with only MP3 as my option (other than uncompressed wave files).

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  • Checking rtp stream audio quality.

    - by chills42
    We are working in a test environment and need to monitor the audio quality of an rtp stream that is being captured using tshark. Right now we are able to capture the audio and access the file through wireshark, but we would like to find a way to save the audio to a .wav file (or similar) via the command line. Does anyone know of a tool that can do this?

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