Search Results

Search found 4461 results on 179 pages for 'pic audio'.

Page 61/179 | < Previous Page | 57 58 59 60 61 62 63 64 65 66 67 68  | Next Page >

  • Nested attributes in the index view?

    - by user283179
    I seem to be getting error: uninitialized constant Style::Pic when I'm trying to render a nested object in to the index view the show view is fine. class Style < ActiveRecord::Base #belongs_to :users has_many :style_images, :dependent => :destroy accepts_nested_attributes_for :style_images, :reject_if => proc { |a| a.all? { |k, v| v.blank?} } #found this here http://ryandaigle.com/articles/2009/2/1/what-s-new-in-edge-rails-nested-attributes has_one :cover, :class_name => "Pic", :order => "updated_at DESC" accepts_nested_attributes_for :cover end class StyleImage < ActiveRecord::Base belongs_to :style #belongs_to :style_as_cover, :class_name => "Style", :foreign_key => "style_id" has_attached_file :pic, :styles => { :small => "200x0>", :normal => "600x> " } validates_attachment_presence :pic #validates_attachment_size :pic, :less_than => 5.megabytes end <% for style_image in @style.style_images %> <li><%= style_image.caption %></li> <div id="show_photo"> <%= image_tag style_image.pic.url(:normal) %></div> <% end %> As you can see from the above The main model style has many style_images, all these style_images are displayed in the show view but, in the the index view I wish to show one image which has been name and will act as a cover that is displayed for each style. in the index controller I have tried the following: class StylesController < ApplicationController layout "mini" def index @styles = Style.find(:all, :inculde => [:cover,]).reverse respond_to do |format| format.html # index.html.erb format.xml { render :xml => @styles } end end and the index <% @styles.each do |style| %> <%=image_tag style.cover.pic.url(:small) %> <% end %> class StyleImage < ActiveRecord::Base belongs_to :style #belongs_to :style_as_cover, :class_name => "Style", :foreign_key => "style_id" has_attached_file :pic, :styles => { :small => "200x0>", :normal => "600x> " } validates_attachment_presence :pic #validates_attachment_size :pic, :less_than => 5.megabytes end In the style_images table there is an cover_id also. From the about you can see that I have included the cover in the controller and the model. I have know idea where I'm going wrong here! If any one can help please do!

    Read the article

  • Why isn't this driver install working (sudo code)?

    - by Nick
    I have a soundcard that I'd like to use and I've been trying to install it and being a new Ubuntu user, I get about half way through this in the Terminal and it stops cooperating with me... See the link (soundcard hyperlink) but basically what I have here: I do the following and it works: sudo apt-get install subversion svn co https://line6linux.svn.sourceforge.net/svnroot/line6linux Change to the directory cd line6linux/driver/trunk Time to build from the source but first make sure you have the latest build and headers sudo apt-get install build-essential sudo apt-get install linux-headers Then after this point it says must specify file to install. Not sure how to do this or what it means. Then, running make gives the following output: ./set_revision.sh ./set_revision.sh: 9: test: https://line6linux.svn.sourceforge.net/svnroot/line6linux/driver/trunk: unexpected operator make -C /lib/modules/3.2.0-29-generic-pae/build CONFIG_LINE6_USB=m SUBDIRS=/home/nick/line6linux/driver/trunk modules make[1]: Entering directory /usr/src/linux-headers-3.2.0-29-generic-pae' CC [M] /home/nick/line6linux/driver/trunk/audio.o /home/nick/line6linux/driver/trunk/audio.c: In function ‘line6_init_audio’: /home/nick/line6linux/driver/trunk/audio.c:30:57: error: ‘THIS_MODULE’ undeclared (first use in this function) /home/nick/line6linux/driver/trunk/audio.c:30:57: note: each undeclared identifier is reported only once for each function it appears in make[2]: * [/home/nick/line6linux/driver/trunk/audio.o] Error 1 make[1]: * [module/home/nick/line6linux/driver/trunk] Error 2 make[1]: Leaving directory/usr/src/linux-headers-3.2.0-29-generic-pae' make: * [default] Error 2 This is in Ubuntu 12.04.1 LTS Another thing, semi related. Cut, copy, paste? Seems like it's different from program to program. I was in the terminal and hit Ctrl-C and then Ctrl-Shift-V in Firefox and it won't paste. But in terminal it will paste. I'm confused. Here is what it's giving me after I hit "Make": nick@NickUbuntu:~/line6linux/driver/trunk$ make ./set_revision.sh ./set_revision.sh: 9: test: https://line6linux.svn.sourceforge.net/svnroot/line6linux/driver/trunk: unexpected operator make -C /lib/modules/3.2.0-29-generic-pae/build CONFIG_LINE6_USB=m SUBDIRS=/home/nick/line6linux/driver/trunk modules make[1]: Entering directory /usr/src/linux-headers-3.2.0-29-generic-pae' CC [M] /home/nick/line6linux/driver/trunk/audio.o /home/nick/line6linux/driver/trunk/audio.c: In function ‘line6_init_audio’: /home/nick/line6linux/driver/trunk/audio.c:30:57: error: ‘THIS_MODULE’ undeclared (first use in this function) /home/nick/line6linux/driver/trunk/audio.c:30:57: note: each undeclared identifier is reported only once for each function it appears in make[2]: *** [/home/nick/line6linux/driver/trunk/audio.o] Error 1 make[1]: *** [_module_/home/nick/line6linux/driver/trunk] Error 2 make[1]: Leaving directory/usr/src/linux-headers-3.2.0-29-generic-pae' make: * [default] Error 2 Looks like these folks also had similar problems: http://ubuntuforums.org/showthread.php?t=1163608&page=3

    Read the article

  • alsa doesn't won't in vlc

    - by freebird
    Alsa Audio Output works fine from terminal aplay /usr/share/sounds/alsa/Noise.wav . But i got to change from default to Alsa Audio Output in vlc . Found in Tools Perfernces Audio Outputs The issue lie when i change it to Alsa i Loose all sound. When i leave it defualt i get a annoying Audio delay of like 200ms or 500ms. from what i have found you have to use Alsa Audio Outpu to fix that issue.

    Read the article

  • Automatically change Sound Input Output device

    - by Senthil Kumaran
    I have to plugin my USB Audio adapter ( 4300054 Gigawire USB Audio Adapter) for audio input because has a combo-input-output port for voice. After I do this, I have go open Sound Settings and manually select the USB Audio adapter for Input and Output, if I do not, the system default remains selected. Is there anyway, I can make Ubuntu to automatically select the USB Audio Adapter as the default as soon as I plug-in?

    Read the article

  • Sound not working on an Intel 5 Series/3400

    - by phoenix7
    lspci gives me these two devices: $ lspci | grep Audio 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 05) 02:00.1 Audio device: ATI Technologies Inc RV710/730 There are two devices listed in System Settings|Sound|Output: RV710/730 Digital Stereo (HDMI) Internal Audio Analog Stereo And finally, the are not muted! Also, when I run an application that accesses the sound card, I can see it in the Applications tab. Any ideas?

    Read the article

  • hdmi audio works only with aplay -D alsa test wavs; open source radeon drivers; kernel 3.5 vgaswitcheroo

    - by user108754
    I've trolled the internets to make hdmi work on my system Ubuntu 12.04 software center kernel 3.5 uname: Linux ubuntu 3.5.0-18-generic #29~precise1-Ubuntu SMP...x86_64 x86_64 x86_64 GNU/Linux open source radeon drivers vgaswitcheroo (hybrid intel/radeon gpu): I boot with intel, not radeon, running. (and recall that with kernel 3.5, vgaswitcheroo now gives info on a third item, "DIS-Audio"; it indicates pwr on my system) ( /etc/rc.local: chown user:user /sys/kernel/debug/ # change "username" with your user name echo OFF /sys/kernel/debug/vgaswitcheroo/switch ) grub indeed now has "radeon.audio=1" for testing audio, I did aplay -l which gave me the card and device, which made me try aplay -D plughw:1,3 /usr/share/sounds/alsa/Front_Center.wav and lo! I get crystal clear sound on my hdtv. If I play an mp3 file as the argument to that command, I get noise as, I guess, aplay interprets the mp3 code as a wav. If I play a .wav that is not in the /usr/share/sounds/alsa/ directory, I get nothing. Internet flash video in browser plays no sound over hdmi. Both system sounds control and pavucontrol have hdmi cedar selected. Alas, I can not get sound for any gui test (left, right). Why would only aplay, and only when directed with "-D plughw", yield sound over hdmi? I've also tried only using one sound program at a time, if it was a limitation of alsa, so I tried aplay with web browser and even the sound control gui closed. I tried each of the last two, running alone. No improvement. alsamixer only shows hda intel and I think it's only the intel audio, not the hdmi.

    Read the article

  • How can I find the song position of a song being played with XACT?

    - by DJ SymBiotiX
    So I'm making a game in XNA and I need to use XACT for my songs (rather than media player). I need to use XACT because each song will have multiple layers that combine when played at the same time (bass, lead, drums) etc. I cant use the media player because the media player can only play one song at a time. Anyways, so lets say I have a song playing with XACT in my project with the following code public SongController() { audioEngine = new AudioEngine(@"Content\Song1\Song1.xgs"); waveBank = new WaveBank(audioEngine, @"Content\Song1\Layers.xwb"); soundBank = new SoundBank(audioEngine, @"Content\Song1\SongLayers.xsb"); songTime = new PlayTime(); Vox = soundBank.GetCue("Vox"); BG = soundBank.GetCue("BG"); Bass = soundBank.GetCue("Bass"); Lead = soundBank.GetCue("Lead"); Other = soundBank.GetCue("Other"); Vox.SetVariable("CueVolume", 100.0f); BG.SetVariable("CueVolume", 100.0f); Bass.SetVariable("CueVolume", 100.0f); Lead.SetVariable("CueVolume", 100.0f); Other.SetVariable("CueVolume", 100.0f); _bassVol = 100.0f; _voxVol = 100.0f; _leadVol = 100.0f; _otherVol = 100.0f; Vox.Play(); BG.Play(); Bass.Play(); Lead.Play(); Other.Play(); } So when I look at the variables in Vox, or BG (they are Cue's btw) I cant seem to find any play position in them. So I guess the question is: Is there a variable I can query to find that data, or do I need to make my own class that starts counting up from the time I start the song? Thanks

    Read the article

  • kAudioSessionProperty_CurrentHardwareSampleRate input/output

    - by iter
    kAudioSessionProperty_CurrentHardwareSampleRate seems to describe the input sampling rate. I wonder if there is a way to determine the available output sampling rate on an iPhone / iPad (iPhone supports 44.1K; iPad, 48K). http://developer.apple.com/iphone/library/documentation/AudioToolbox/Reference/AudioSessionServicesReference/Reference/reference.html#//apple_ref/doc/c_ref/kAudioSessionProperty_CurrentHardwareSampleRate

    Read the article

  • Loop OpenAL source with offset

    - by ressaw
    The OpenAL API states that an setting an offset still causes the sound to loop back to zero for looping sources. But is there a way to loop and still have an offset somehow? I have an mp3, and since it contains headers with information at the start of the file, there's a small, but noticable, delay in looping when it rewinds. If not, are there any other compressed formats that don't contain these empty headers?

    Read the article

  • concatenating mp3 files or joining mp3 files using java

    - by Sukhhhh
    We would like to concatenate/merge/join mp3 files seamlessly using "java" in any environment. We are trying the following options at the moment ( please let us know any other options): Using JMF -- ruled out as it supported only in windows http://java.sun.com/javase/technologies/desktop/media/jmf/reference/faqs/index.html Using tritinous , jlayer and Lame combination. Please let us know thoughts and the links those mention about concatenate/merge/join mp3 files using option 2.

    Read the article

  • can not get jplayer plugin to work

    - by Richard
    Hello, I hope somebody has some experience with the jplayer plugin I have been staring at the sourcecode of the demo's and looking in firebug, but I can't see why it is not showing at all. It also try's to use the flash file, but in other examples the embed code does not show up in the container div either. How could I get this to work, or debug? $(document).ready(function(){ $("#jpId").jPlayer( { ready: function () { this.element.jPlayer("setFile", "/mp3/nobodymove.mp3"); // Defines the mp3 } }); }); thanks, Richard

    Read the article

  • VB FFT - stuck understanding relationship of results to frequency

    - by WaveyDavey
    Trying to understand an fft (Fast Fourier Transform) routine I'm using (stealing)(recycling) Input is an array of 512 data points which are a sample waveform. Test data is generated into this array. fft transforms this array into frequency domain. Trying to understand relationship between freq, period, sample rate and position in fft array. I'll illustrate with examples: ======================================== Sample rate is 1000 samples/s. Generate a set of samples at 10Hz. Input array has peak values at arr(28), arr(128), arr(228) ... period = 100 sample points peak value in fft array is at index 6 (excluding a huge value at 0) ======================================== Sample rate is 8000 samples/s Generate set of samples at 440Hz Input array peak values include arr(7), arr(25), arr(43), arr(61) ... period = 18 sample points peak value in fft array is at index 29 (excluding a huge value at 0) ======================================== How do I relate the index of the peak in the fft array to frequency ?

    Read the article

  • FMOD.net streaming, callback and exinfo parameters

    - by Tesserex
    I posted a question on gamedev about how to play nsf files (NES console music) in FMOD. It didn't get any results, but since then I made some progress. I decided that the easiest method was just to compile an existing player into a dll and then call it from C# to populate my buffer. The problem now is getting it to sound right, and making sure all my paremeters are correct. Here are the facts so far: The nsf dll is dealing with shorts, so the data is PCM16. The sample nsf I'm using has a playback rate of 60 Hz. Just for playing around now, I'm using a frequency of 48000. Based on 2 and 3, the dll calculates a necessary buffer size of 48000 / 60hz = 800. This means it will render 800 shorts worth of buffer for every simulated NES frame. I've so far got my C# code to play the nsf, at the correct pitch and tempo, but it's very grainy / fuzzy, which I'm attributing to the fact that the FMOD read callback is giving a data length of 1600, whereas I should be expecting 800. I've tried playing around with all the numbers and it either crashes, or the music changes pitch, tempo, or both. Here's some of my C# code: uint channels = 1, frequency = 48000; FMOD.MODE mode = (FMOD.MODE.DEFAULT | FMOD.MODE.OPENUSER | FMOD.MODE.LOOP_NORMAL); FMOD.Sound sound = new FMOD.Sound(); FMOD.CREATESOUNDEXINFO ex = new FMOD.CREATESOUNDEXINFO(); ex.cbsize = Marshal.SizeOf(ex); ex.fileoffset = 0; ex.format = FMOD.SOUND_FORMAT.PCM16; // does this even matter? It doesn't change my results as long as it's long enough for one update ex.length = frequency; ex.numchannels = (int)channels; ex.defaultfrequency = (int)frequency; ex.pcmreadcallback = pcmreadcallback; ex.dlsname = null; // eventually I will calculate this with frequency / nsf hz, but I'm just testing for now ex.decodebuffersize = 800; // from the dll load_nsf_file("file.nsf", 8, (int)frequency); // 8 is the track number to play var result = system.createSound( (string)null, (mode | FMOD.MODE.CREATESTREAM), ref ex, ref sound); channel = new FMOD.Channel(); result = system.playSound(FMOD.CHANNELINDEX.FREE, sound, false, ref channel); private FMOD.RESULT PCMREADCALLBACK(IntPtr soundraw, IntPtr data, uint datalen) { // from the dll process_buffer(data, (int)800); // if I use datalen, it usually crashes (I can't get datalen to = 800 safely) return FMOD.RESULT.OK; } So here are some of my questions: What is the relationship between exinfo.decodebuffersize, frequency, and the datalen parameter of the read callback? With this code sample, it's coming in as 3200. I don't know where that factor of 4 between it and the decodebuffersize comes from. Is datalen in the callback referring to number of bytes, or shorts? The process_buffer function takes a short array and its length. I would expect fmod is talking about shorts as well because I told it PCM16. Maybe my playback quality is bad for some totally different reason. If so I have no idea where to begin solving that. Any ideas there?

    Read the article

  • NAudio Mp3 Playback in Console

    - by Kurru
    Hi I'm trying to make a helper dll that will simplify the NAudio framework into a subset of functions I'm likely to need but I've hit a stumbling block right off the bat. I'm trying to use the following code to play an mp3 but I'm not hearing anything at all. Any help would be appreciated! static WaveOut waveout; static WaveStream playback; static System.Threading.ManualResetEvent wait = new System.Threading.ManualResetEvent(false); static void Main(string[] args) { System.Threading.Thread t = new System.Threading.Thread(new System.Threading.ThreadStart(PlaySong)); t.Start(); wait.WaitOne(); System.Threading.Thread.Sleep(2 * 1000); waveout.Stop(); waveout.Dispose(); playback.Dispose(); } static void PlaySong() { waveout = new WaveOut(); playback = OpenMp3Stream(@"songname.mp3"); waveout.Init(playback); waveout.Play(); Console.WriteLine("Started"); wait.Set(); } private static WaveChannel32 OpenMp3Stream(string fileName) { WaveChannel32 inputStream; WaveStream mp3Reader = new Mp3FileReader(fileName); WaveStream pcmStream = WaveFormatConversionStream.CreatePcmStream(mp3Reader); WaveStream blockAlignedStream = new BlockAlignReductionStream(pcmStream); inputStream = new WaveChannel32(blockAlignedStream); return inputStream; }

    Read the article

  • Sound/Silence in a wav file.

    - by Vivek
    Hi, I am searching for a utility/code that could detect and let me know if my 1 minute wav file contains sound or not ? Other way, if it could detect the duration of the silence(if exists) at any position in the wav file, that would also server the purpose. Does SOX support any command for that ? I tried with Java, but didnt found anything in JMF. Thanks Vivek

    Read the article

  • SoundManager2 has irregular latency

    - by Stefan Monov
    I'm playing some notes at regular intervals. Each one is delayed by a random number of milliseconds, creating a jarring irregular effect. How do I fix it? Note: I'm OK with some latency, just as long as it's consistent. Answers of the type "implement your own small SoundManager2 replacement, optimized for timing-sensitive playback" are OK, if you know how to do that :) but I'm trying to avoid rewriting my whole app in flash for now. For an example of app with zero audible latency see the flash-based ToneMatrix. Testcase (see it here live or get it in an zip): <head> <title></title> <script type="text/javascript" src="http://www.schillmania.com/projects/soundmanager2/script/soundmanager2.js"> </script> <script type="text/javascript"> soundManager.url = '.' soundManager.flashVersion = 9 soundManager.useHighPerformance = true soundManager.useFastPolling = true soundManager.autoLoad = true function recur(func, delay) { window.setTimeout(function() { recur(func, delay); func(); }, delay) } soundManager.onload = function() { var sound = soundManager.createSound("test", "test.mp3") recur(function() { sound.play() }, 300) } </script> </head> <body> </body> </html>

    Read the article

  • How do I play back a WAV in ActionScript?

    - by Jeremy White
    Please see the class I have created at http://textsnip.com/51013f for parsing a WAVE file in ActionScript 3.0. This class is correctly pulling apart info from the file header & fmt chunks, isolating the data chunk, and creating a new ByteArray to store the data chunk. It takes in an uncompressed WAVE file with a format tag of 1. The WAVE file is embedded into my SWF with the following Flex embed tag: [Embed(source="some_sound.wav", mimeType="application/octet-stream")] public var sound_class:Class; public var wave:WaveFile = new WaveFile(new sound_class()); After the data chunk is separated, the class attempts to make a Sound object that can stream the samples from the data chunk. I'm having issues with the streaming process, probably because I'm not good at math and don't really know what's happening with the bits/bytes, etc. Here are the two documents I'm using as a reference for the WAVE file format: http://www.lightlink.com/tjweber/StripWav/Canon.html https://ccrma.stanford.edu/courses/422/projects/WaveFormat/ Right now, the file IS playing back! In real time, even! But...the sound is really distorted. What's going on?

    Read the article

  • (Android SDk 2.1) Getting error when I use setAudioSource and setVideoSource

    - by Rainfer
    I got the follow error when I run setAudioSource and setVideoSource. 03-16 10:26:25.302: ERROR/audio_input(52): unsupported parameter: x-pvmf/media-input-node/cap-config-interface;valtype=key_specific_value 03-16 10:26:25.302: ERROR/audio_input(52): VerifyAndSetParameter failed 03-16 10:26:25.302: ERROR/CameraInput(52): Unsupported parameter(x-pvmf/media-input-node/cap-config-interface;valtype=key_specific_value) 03-16 10:26:25.302: ERROR/CameraInput(52): VerifiyAndSetParameter failed on parameter #0 This error happen on both emulator and the device. (I am using Google nexus one) I have set the CAMERA and RECORD_AUDIO user permission already. I spent many days but I still cannot figure out what is the cause of this runtime error.

    Read the article

  • Detect and record a sound with python

    - by Jean-Pierre
    I'm using this program to record a sound in python: import pyaudio import wave import sys chunk = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 44100 RECORD_SECONDS = 5 WAVE_OUTPUT_FILENAME = "output.wav" p = pyaudio.PyAudio() stream = p.open(format = FORMAT, channels = CHANNELS, rate = RATE, input = True, frames_per_buffer = chunk) print "* recording" all = [] for i in range(0, RATE / chunk * RECORD_SECONDS): data = stream.read(chunk) all.append(data) print "* done recording" stream.close() p.terminate() write data to WAVE file data = ''.join(all) wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb') wf.setnchannels(CHANNELS) wf.setsampwidth(p.get_sample_size(FORMAT)) wf.setframerate(RATE) wf.writeframes(data) wf.close() I want to change the program to start recording when sound is detected by the sound card input. Probably should compare the input sound level in Chunk, but how do this?

    Read the article

  • Playing sounds in iPhone SDK?

    - by seanny94
    Does anyone have a snippet that uses the AudioToolBox framework that can be used to play a short sound? I would be grateful if you shared it with me and the rest of the community. Everywhere else I have looked doesn't seem to be too clear with their code. Thanks!

    Read the article

< Previous Page | 57 58 59 60 61 62 63 64 65 66 67 68  | Next Page >