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  • Audio tag doesn't work in FF-Ubuntu 12.04

    - by Nyx
    Does anyone know why this code... <audio width="0" height="0" autoplay="autoplay" loop="loop" preload="none"> <source src="images/musica/Intro.ogg" type="audio/ogg" /> <source src="images/musica/Intro.mp3" type="audio/mpeg" /> </audio> ...works fine in FF17-WinXP and not in FF17-Ubuntu 12.04? I think something is wrong with MIME types but everything looks normal. After searching on the web for days I couldn't find a good answer. Thanks

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  • How to play multiple audio sources simultaneously in Silverlight

    - by Shurup
    I want to play simultaneous multiply audio sources in Silverlight. So I've created a prototype in Silverlight 4 that should play a two mp3 files containing the same ticks sound with an intervall 1 second. So these files must be sounded as one sound if they will be played together with any whole second offsets (0 and 1, 0 and 2, 1 and 1 seconds, etc.) I my prototype I use two MediaElement (me and me2) objects. DateTime startTime; private void Play_Clicked(object sender, RoutedEventArgs e) { me.SetSource(new FileStream(file1), FileMode.Open))); me2.SetSource(new FileStream(file2), FileMode.Open))); var timer = new DispatcherTimer { Interval = TimeSpan.FromMilliseconds(1) }; timer.Tick += RefreshData; timer.Start(); } First file should be played at 00:00 sec. and the second in 00:02 second. void RefreshData(object sender, EventArgs e) { if(me.CurrentState != MediaElementState.Playing) { startTime = DateTime.Now; me.Play(); return; } var elapsed = DateTime.Now - startTime; if(me2.CurrentState != MediaElementState.Playing && elapsed >= TimeSpan.FromSeconds(2)) { me2.Play(); ((DispatcherTimer)sender).Stop(); } } The tracks played every time different and not simultaneous as they should (as one sound). Addition: I've tested a code from the Bobby's answer. private void Play_Clicked(object sender, RoutedEventArgs e) { me.SetSource(new FileStream(file1), FileMode.Open))); me2.SetSource(new FileStream(file2), FileMode.Open))); // This code plays well enough. // me.Play(); // me2.Play(); // But adding the 2 second offset using the timer, // they play no simultaneous. var timer = new DispatcherTimer { Interval = TimeSpan.FromSeconds(2) }; timer.Tick += (source, arg) => { me2.Play(); ((DispatcherTimer)source).Stop(); }; timer.Start(); } Is it possible to play them together using only one MediaElement or any implementation of MediaStreamSource that can play multiply sources?

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  • High level vs. low level programming. Do I really have to choose?

    - by EpsilonVector
    Every once in a while I'm asked in interviews which I like the best- low level or high level. It seems to me that the implicit message is that they are both a specialty and they want to know which direction I'm heading. The trouble is, I seem to like both. Low level is extremely challenging and often requires a great deal of esoteric knowledge. High level is where all the sexy things happen: applications that people use directly, results that can be easily demonstrated (showed off) in a way that is accessible to everybody, and you get to work with really advanced tools and interact with new technologies. I would really love to do both, even if it means alternating between them (I doubt there are jobs that will let me do both simultaneously), but I'm guessing that the industry rewards specialists more than generalists. Will it really be problematic career wise if I never choose one over the other? Is it practical to alternate between the two in the sense that if I were to leave a job doing one of them, I should experience no "friction" trying to get a job doing the other (assuming I'm reasonably in the loop)? Are there career opportunities where you get to do both? Do I really have to choose one over the other?

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  • How can I stream audio signals from various devices/computers to my home server?

    - by Breakthrough
    I currently have a headless home server set up (running Ubuntu 12.04 server edition) running a simple Apache HTTP server. The server is near an audio receiver, which controls a set of indoor and outdoor speakers in my home. Recently, my father purchased a Bluetooth adapter, which our various laptops and cellphones can connect to, outputting the music to the speakers. I was hoping to find a solution that worked over Wi-Fi, namely because it won't cost anything (I already have a server with an audio card), and it doesn't depend on Bluetooth. Is there any cross-platform (preferably free and open-source) solution that I can use which will allow me to stream audio to my home server, over my home network, from a wide variety of devices (laptops running Windows/Linux or cellphones running Android/BB/iOS)? I need something that works at least with Windows and Android. Also, just to clairfy, I want something that simply allows devices to connect to my server and output an audio signal without any action on the server end (since it's a server hidden away near my receiver). Any subsequent connection attempt should be dropped, so only one device can be in control of the stereo at once.

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  • Help with dynamic range compression function (audio)

    - by MusiGenesis
    I am writing a C# function for doing dynamic range compression (an audio effect that basically squashes transient peaks and amplifies everything else to produce an overall louder sound). I have written a function that does this (I think): public static void Compress(ref short[] input, double thresholdDb, double ratio) { double maxDb = thresholdDb - (thresholdDb / ratio); double maxGain = Math.Pow(10, -maxDb / 20.0); for (int i = 0; i < input.Length; i += 2) { // convert sample values to ABS gain and store original signs int signL = input[i] < 0 ? -1 : 1; double valL = (double)input[i] / 32768.0; if (valL < 0.0) { valL = -valL; } int signR = input[i + 1] < 0 ? -1 : 1; double valR = (double)input[i + 1] / 32768.0; if (valR < 0.0) { valR = -valR; } // calculate mono value and compress double val = (valL + valR) * 0.5; double posDb = -Math.Log10(val) * 20.0; if (posDb < thresholdDb) { posDb = thresholdDb - ((thresholdDb - posDb) / ratio); } // measure L and R sample values relative to mono value double multL = valL / val; double multR = valR / val; // convert compressed db value to gain and amplify val = Math.Pow(10, -posDb / 20.0); val = val / maxGain; // re-calculate L and R gain values relative to compressed/amplified // mono value valL = val * multL; valR = val * multR; double lim = 1.5; // determined by experimentation, with the goal // being that the lines below should never (or rarely) be hit if (valL > lim) { valL = lim; } if (valR > lim) { valR = lim; } double maxval = 32000.0 / lim; // convert gain values back to sample values input[i] = (short)(valL * maxval); input[i] *= (short)signL; input[i + 1] = (short)(valR * maxval); input[i + 1] *= (short)signR; } } and I am calling it with threshold values between 10.0 db and 30.0 db and ratios between 1.5 and 4.0. This function definitely produces a louder overall sound, but with an unacceptable level of distortion, even at low threshold values and low ratios. Can anybody see anything wrong with this function? Am I handling the stereo aspect correctly (the function assumes stereo input)? As I (dimly) understand things, I don't want to compress the two channels separately, so my code is attempting to compress a "virtual" mono sample value and then apply the same degree of compression to the L and R sample value separately. Not sure I'm doing it right, however. I think part of the problem may the "hard knee" of my function, which kicks in the compression abruptly when the threshold is crossed. I think I may need to use a "soft knee" like this: Can anybody suggest a modification to my function to produce the soft knee curve?

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  • Why does 3.5 mm audio out work through headphones but not through external speakers?

    - by Rickster
    I have a computer that has a 3.5 mm audio jack on the front of it. The computer itself has no speakers, so this is the only way to hear sound. If I plug headphones into it, the audio properly plays through the headphones, and if I plug in external speakers it used to play through them as well. Just today I turned on my computer and the audio no longer plays through the speakers, but if I plug in the headphones instead it works. The speakers aren't broken, as both the speakers and headphones work in my iPod and play music. I thought that 3.5 mm jacks could not send data back to the computer, and the computer had no way of differentiating between different devices plugged into the jack. If this is true, how is it that the computer plays audio through the headphones but not through speakers plugged into the same 3.5 mm jack, and both devices are functional? Or is my knowledge on 3.5 mm jacks incorrect? I don't believe drivers are important, as the same driver runs the 3.5 mm jack for all devices, but if necessary I can provide additional information. Any ideas would be appreciated. Thanks!

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  • Troubleshooting High-CPU Utilization for SQL Server

    - by Susantha Bathige
    The objective of this FAQ is to outline the basic steps in troubleshooting high CPU utilization on  a server hosting a SQL Server instance. The first and the most common step if you suspect high CPU utilization (or are alerted for it) is to login to the physical server and check the Windows Task Manager. The Performance tab will show the high utilization as shown below: Next, we need to determine which process is responsible for the high CPU consumption. The Processes tab of the Task Manager will show this information: Note that to see all processes you should select Show processes from all user. In this case, SQL Server (sqlserver.exe) is consuming 99% of the CPU (a normal benchmark for max CPU utilization is about 50-60%). Next we examine the scheduler data. Scheduler is a component of SQLOS which evenly distributes load amongst CPUs. The query below returns the important columns for CPU troubleshooting. Note – if your server is under severe stress and you are unable to login to SSMS, you can use another machine’s SSMS to login to the server through DAC – Dedicated Administrator Connection (see http://msdn.microsoft.com/en-us/library/ms189595.aspx for details on using DAC) SELECT scheduler_id ,cpu_id ,status ,runnable_tasks_count ,active_workers_count ,load_factor ,yield_count FROM sys.dm_os_schedulers WHERE scheduler_id See below for the BOL definitions for the above columns. scheduler_id – ID of the scheduler. All schedulers that are used to run regular queries have ID numbers less than 1048576. Those schedulers that have IDs greater than or equal to 1048576 are used internally by SQL Server, such as the dedicated administrator connection scheduler. cpu_id – ID of the CPU with which this scheduler is associated. status – Indicates the status of the scheduler. runnable_tasks_count – Number of workers, with tasks assigned to them that are waiting to be scheduled on the runnable queue. active_workers_count – Number of workers that are active. An active worker is never preemptive, must have an associated task, and is either running, runnable, or suspended. current_tasks_count - Number of current tasks that are associated with this scheduler. load_factor – Internal value that indicates the perceived load on this scheduler. yield_count – Internal value that is used to indicate progress on this scheduler.                                                                 Now to interpret the above data. There are four schedulers and each assigned to a different CPU. All the CPUs are ready to accept user queries as they all are ONLINE. There are 294 active tasks in the output as per the current_tasks_count column. This count indicates how many activities currently associated with the schedulers. When a  task is complete, this number is decremented. The 294 is quite a high figure and indicates all four schedulers are extremely busy. When a task is enqueued, the load_factor  value is incremented. This value is used to determine whether a new task should be put on this scheduler or another scheduler. The new task will be allocated to less loaded scheduler by SQLOS. The very high value of this column indicates all the schedulers have a high load. There are 268 runnable tasks which mean all these tasks are assigned a worker and waiting to be scheduled on the runnable queue.   The next step is  to identify which queries are demanding a lot of CPU time. The below query is useful for this purpose (note, in its current form,  it only shows the top 10 records). SELECT TOP 10 st.text  ,st.dbid  ,st.objectid  ,qs.total_worker_time  ,qs.last_worker_time  ,qp.query_plan FROM sys.dm_exec_query_stats qs CROSS APPLY sys.dm_exec_sql_text(qs.sql_handle) st CROSS APPLY sys.dm_exec_query_plan(qs.plan_handle) qp ORDER BY qs.total_worker_time DESC This query as total_worker_time as the measure of CPU load and is in descending order of the  total_worker_time to show the most expensive queries and their plans at the top:      Note the BOL definitions for the important columns: total_worker_time - Total amount of CPU time, in microseconds, that was consumed by executions of this plan since it was compiled. last_worker_time - CPU time, in microseconds, that was consumed the last time the plan was executed.   I re-ran the same query again after few seconds and was returned the below output. After few seconds the SP dbo.TestProc1 is shown in fourth place and once again the last_worker_time is the highest. This means the procedure TestProc1 consumes a CPU time continuously each time it executes.      In this case, the primary cause for high CPU utilization was a stored procedure. You can view the execution plan by clicking on query_plan column to investigate why this is causing a high CPU load. I have used SQL Server 2008 (SP1) to test all the queries used in this article.

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  • How can I stop Ubuntu from playing audio from 2 interfaces at the same time?

    - by Solignis
    Hi there, I just loaded Ubuntu 10.10 Maverick on my home machine. The machine is running a Core2Duo E6750 on an MSI motherboard with an Nvidia GTX260-OC Graphics card. The problem I am having as stated in the title is for some reason Ubuntu is playing audio through my headphone coming out from the computer and it is also playing the audio at the exact same time through the HDMI connection coming out of the graphics card, it has a plug to allow this. What is going on, I have never seen this before. Most importantly of all can it be fixed so that I can sepertate the 2 interfaces, the one is a standard PC audio IO and the HDMI one is connected through the mobo's internal SPDIF. More information can be provided if required.

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  • Default audio device gives an error on WINDOWS 7 (x64) when triing to run VLC from CMD (VideoLAN, VL

    - by Ole Jak
    I use WINDOWS 7 (x64) (Russian) I want to stream life audio from my default audio capture device (microphone) When I set up VLM settings using visual enviroment instruments - VLM settings it all works fine. But when I export created settings/configuration *.vlm file and try to inport it into VLM it gives me nothing I opened that .vlm there is some text... so now I try to run VLC with default settings like this: vlc -i dshow:// --dshow-adev= :sout=#transcode{acodec=mp3,ab=128,channels=2,samplerate=44100}:std{access=http,mux=raw,dst=127.0.0.1:8084} but it dies giving me errors...=( So what shall I do to do live MP3 streaming from my default audio input device using VLC in non UI mode?

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  • Can I reprogram a microphone input to be used as an audio output? (on XP)

    - by qftme
    I have a five year old Sony Vaio laptop (vgn-fw31m) that has had impact damage to the audio-output mini-jack for about the last year or so. In a recent discussion with my brother, we wondered whether it would be possible to write a program that would enable windows to use the microphone mini-jack input as the audio-output? As I currently use this laptop for work I am not keen to risk pulling it apart in order to replace the components comprising the audio-out. I therefore 'hope' that a programming solution exists. I would really appreciate any advice on this and eagerly await your response. Kind regards, qftme :)

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  • Help calling def from class.

    - by wtzolt
    Hello, Noob question... class msgbox: def __init__(self, lbl_msg = '', dlg_title = ''): self.wTree = gtk.glade.XML('msgbox.glade') self.wTree.get_widget('dialog1').set_title(dlg_title) self.wTree.get_widget('label1').set_text(lbl_msg) self.wTree.signal_autoconnect( {'on_okbutton1_clicked':self.done} ) def done(self,w): self.wTree.get_widget('dialog1').destroy() class Fun(object): wTree = None def __init__(self): self.wTree = gtk.glade.XML( "main.glade" ) self.wTree.signal_autoconnect( {'on_buttonOne' : self.one,} ) gtk.main() @yieldsleep def one(self, widget, data=None): self.msg = msgbox('Please wait...','') yield 500 self.msg = msgbox().done() # <----------------??? self.msg = msgbox('Done!','') With this i get an error: messageBox().done() TypeError: done() takes exactly 2 arguments (1 given) How can i make the dialog box with "please wait" to close before the second dialog box with "done" appears?? Thank you.

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  • Order by text and then by number

    - by Chaim Chaikin
    I have data like: Audio 1 File 10 Audio 2 Audio 3 File 11 Audio 13 Audio 22 File 20 Test 22 Audio 10 File 1 File 2 I need it order first by the text (i.e. Audio, File, Test) and then by number (1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22 etc.) The problem is that sorting it returns something like this: Audio 1 Audio 10 Audio 13 Audio 2 Audio 22 Audio 3 File 1 File 10 File 11 File 2 File 20 Test 22 While the result I want is: Audio 1 Audio 2 Audio 3 Audio 10 Audio 13 Audio 22 File 1 File 2 File 10 File 11 File 20 Test 22 If they were just numbers (i.e. without the audio, file, test) then I could just sort numerically. However, how can I sort here first by text and then by number.

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  • Java style FOR loop in a clojure interpeter ?

    - by Kevin
    I have a basic interpreter in clojure. Now i need to implement for (initialisation; finish-test; loop-update) { statements } inside my interpreter. I will attach my interpreter code I got so far. Any help is appreciated. Interpreter (declare interpret make-env) ;; (def do-trace false) ;; ;; simple utilities (def third ; return third item in a list (fn [a-list] (second (rest a-list)))) (def fourth ; return fourth item in a list (fn [a-list] (third (rest a-list)))) (def run ; make it easy to test the interpreter (fn [e] (println "Processing: " e) (println "=> " (interpret e (make-env))))) ;; for the environment (def make-env (fn [] '())) (def add-var (fn [env var val] (cons (list var val) env))) (def lookup-var (fn [env var] (cond (empty? env) 'error (= (first (first env)) var) (second (first env)) :else (lookup-var (rest env) var)))) ;; -- define numbers (def is-number? (fn [expn] (number? expn))) (def interpret-number (fn [expn env] expn)) ;; -- define symbols (def is-symbol? (fn [expn] (symbol? expn))) (def interpret-symbol (fn [expn env] (lookup-var env expn))) ;; -- define boolean (def is-boolean? (fn [expn] (or (= expn 'true) (= expn 'false)))) (def interpret-boolean (fn [expn env] expn)) ;; -- define functions (def is-function? (fn [expn] (and (list? expn) (= 3 (count expn)) (= 'lambda (first expn))))) (def interpret-function (fn [expn env] expn)) ;; -- define addition (def is-plus? (fn [expn] (and (list? expn) (= 3 (count expn)) (= '+ (first expn))))) (def interpret-plus (fn [expn env] (+ (interpret (second expn) env) (interpret (third expn) env)))) ;; -- define subtraction (def is-minus? (fn [expn] (and (list? expn) (= 3 (count expn)) (= '- (first expn))))) (def interpret-minus (fn [expn env] (- (interpret (second expn) env) (interpret (third expn) env)))) ;; -- define multiplication (def is-times? (fn [expn] (and (list? expn) (= 3 (count expn)) (= '* (first expn))))) (def interpret-times (fn [expn env] (* (interpret (second expn) env) (interpret (third expn) env)))) ;; -- define division (def is-divides? (fn [expn] (and (list? expn) (= 3 (count expn)) (= '/ (first expn))))) (def interpret-divides (fn [expn env] (/ (interpret (second expn) env) (interpret (third expn) env)))) ;; -- define equals test (def is-equals? (fn [expn] (and (list? expn) (= 3 (count expn)) (= '= (first expn))))) (def interpret-equals (fn [expn env] (= (interpret (second expn) env) (interpret (third expn) env)))) ;; -- define greater-than test (def is-greater-than? (fn [expn] (and (list? expn) (= 3 (count expn)) (= '> (first expn))))) (def interpret-greater-than (fn [expn env] (> (interpret (second expn) env) (interpret (third expn) env)))) ;; -- define not (def is-not? (fn [expn] (and (list? expn) (= 2 (count expn)) (= 'not (first expn))))) (def interpret-not (fn [expn env] (not (interpret (second expn) env)))) ;; -- define or (def is-or? (fn [expn] (and (list? expn) (= 3 (count expn)) (= 'or (first expn))))) (def interpret-or (fn [expn env] (or (interpret (second expn) env) (interpret (third expn) env)))) ;; -- define and (def is-and? (fn [expn] (and (list? expn) (= 3 (count expn)) (= 'and (first expn))))) (def interpret-and (fn [expn env] (and (interpret (second expn) env) (interpret (third expn) env)))) ;; -- define with (def is-with? (fn [expn] (and (list? expn) (= 3 (count expn)) (= 'with (first expn))))) (def interpret-with (fn [expn env] (interpret (third expn) (add-var env (first (second expn)) (interpret (second (second expn)) env))))) ;; -- define if (def is-if? (fn [expn] (and (list? expn) (= 4 (count expn)) (= 'if (first expn))))) (def interpret-if (fn [expn env] (cond (interpret (second expn) env) (interpret (third expn) env) :else (interpret (fourth expn) env)))) ;; -- define function-application (def is-function-application? (fn [expn env] (and (list? expn) (= 2 (count expn)) (is-function? (interpret (first expn) env))))) (def interpret-function-application (fn [expn env] (let [function (interpret (first expn) env)] (interpret (third function) (add-var env (first (second function)) (interpret (second expn) env)))))) ;; the interpreter itself (def interpret (fn [expn env] (cond do-trace (println "Interpret is processing: " expn)) (cond ; basic values (is-number? expn) (interpret-number expn env) (is-symbol? expn) (interpret-symbol expn env) (is-boolean? expn) (interpret-boolean expn env) (is-function? expn) (interpret-function expn env) ; built-in functions (is-plus? expn) (interpret-plus expn env) (is-minus? expn) (interpret-minus expn env) (is-times? expn) (interpret-times expn env) (is-divides? expn) (interpret-divides expn env) (is-equals? expn) (interpret-equals expn env) (is-greater-than? expn) (interpret-greater-than expn env) (is-not? expn) (interpret-not expn env) (is-or? expn) (interpret-or expn env) (is-and? expn) (interpret-and expn env) ; special syntax (is-with? expn) (interpret-with expn env) (is-if? expn) (interpret-if expn env) ; functions (is-function-application? expn env) (interpret-function-application expn env) :else 'error)))

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  • Python - List and Loop in one def

    - by Dunwitch
    I'm trying to get the def wfsc_pod1 and wfsc_ip into the same def. I'm not quite sure how to approach the problem. I want wfsc_pod1 to display all the information for name, subnet and gateway. Then wfsc_ip shows the ip addresses below it. I also get a None value when I run it as it. Not sure why. Anything more pythonic is more appreciated. class OutageAddress: subnet = ["255.255.255.0", "255.255.255.1"] # Gateway order is matched with names gateway = ["192.168.1.1", "192.168.1.2", "192.168.1.3", "192.168.1.4", "192.168.1.5", "192.168.1.6", "192.168.1.7", "192.168.1.8", "192.168.1.9"] name = ["LOC1", "LOC2", "LOC3", "LOC4", "LOC5", "LOC6", "LOC7", "LOC8", "LOC9"] def wfsc_pod1(self): wfsc_1 = "%s\t %s\t %s\t" % (network.name[0],network.subnet[0],network.gateway[0]) return wfsc_1 def wfsc_ip(self): for ip in range(100,110): ip = "192.168.1."+str(ip) print ip network = OutageAddress() print network.wfsc_pod1() print network.wfsc_ip()

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  • KDE: How can I select audio output device for mplayer?

    - by grimripper
    I recently installed Kubuntu 13.10 64-bit, and I'm having a problem with selecting audio output device. In Phonon, when I select audio device preference order and press Apply, Amarok and Dragon will immediately switch to the preferred device. VLC and SMplayer are not affected. VLC has its own setting for selecting the output device, but SMplayer remains a problem. It always plays audio on internal audio, and I can't change output to HDMI. How can I select HDMI for SMplayer's audio output device? I don't know if it matters, but when I select HDMI audio in Phonon and click Test, the test sound plays on the internal audio output as well. In the hardware settings tab, the front left and front right test buttons play audio on HDMI. Also, volume up/down buttons affect HDMI volume when SMplayer is focused. This would make sense if I could get SMplayer to play audio over HDMI, but it would be better if the volume keys affected SMplayer's own volume, or the "mplayer2: audio stream" which appears in volume control while mplayer is playing. EDIT: I've recompiled mplayer with alsa support, and can now select the audio output device from SMplayer's settings. Didn't affect the issue with Phonon of course, but it's a suitable workaround.

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  • AVAudioRecorder - Continue recording to file after user stops recording by leaving the application a

    - by Tegeril
    Can this be done? And if not, how far down towards Core Audio do I need to go (what method of recording should I be using instead)? I've noticed the behavior of AVAudioRecorder is to overwrite a file if it finds one at the path provided when you request that it record again, so I know that's not going to work. I'm also curious about file format restriction with this idea. Can you effectively resume an AAC or IMA4 encoding (the length of the files I want to record make WAV and probably even Apple Lossless prohibitive)? Thanks.

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  • How to get audio driver for compaq c700 ?

    - by Leena
    Hi, Initially i have audio driver and its works fine.But some times speaker was clear.So one of my friend installed some audio driver,after that totally disabled the volume. For that reason, i also tried to get audio driver and installed many times.Now i don't know many drivers .inf in my laptop.from device manager i have deleted the audio driver's,below i have attached the screen shot yours kind reference. Please help me to get audio drivers.First, i need to remove the unwanted drivers .inf files from laptop then i have to install the new audio driver. Experts,please suggest me to get audio driver without reinstall the OS. Details: Compaq c700 (i don't know model number) windows xp sp2 p/n : KT188PA#ACJ I appreciate your help.

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  • How to get audio driver for compaq c700 ?

    - by Leena
    Hi, Initially i have audio driver and its works fine.But some times speaker was clear.So one of my friend installed some audio driver,after that totally disabled the volume. For that reason, i also tried to get audio driver and installed many times.Now i don't know many drivers .inf in my laptop.from device manager i have deleted the audio driver's,below i have attached the screen shot yours kind reference. Please help me to get audio drivers.First, i need to remove the unwanted drivers .inf files from laptop then i have to install the new audio driver. Experts,please suggest me to get audio driver without reinstall the OS. Details: Compaq c700 (i don't know model number) windows xp sp2 p/n : KT188PA#ACJ I appreciate your help.

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  • How to get audio driver for compaq c700 ?

    - by Leena
    Hi, Initially i have audio driver and its works fine.But some times speaker was clear.So one of my friend installed some audio driver,after that totally disabled the volume. For that reason, i also tried to get audio driver and installed many times.Now i don't know many drivers .inf in my laptop.from device manager i have deleted the audio driver's,below i have attached the screen shot yours kind reference. Please help me to get audio drivers.First, i need to remove the unwanted drivers .inf files from laptop then i have to install the new audio driver. Experts,please suggest me to get audio driver without reinstall the OS. Details: Compaq c700 (i don't know model number) windows xp sp2 p/n : KT188PA#ACJ I appreciate your help.

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  • Streaming Audio over UDP to Android

    - by Mr. Pig
    Is it possible to have Android (perhaps via MediaPlayer or a different existing class) accept media streams over UDP? I've successfully had MediaPlayer connect to an HTTP stream (as well as static files hosted on an HTTP server) but I'm wondering how one would go about accepting a stream from a UDP source. I've seen this and suppose a solution similar to that (where I download the stream via an independent UDP socket and then move the data to a MemoryBuffer that I then pass to MediaPlayer) is an option but I'm curious if a method already exists in the SDK, and if it does not, what other options do I have? Thanks

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  • Audio recording and playback in Silverlight

    - by Ramesh
    I have a Silverlight 4 application that records user's voice through the mic. Now, as soon as the recording is completed, I need to play the recorded voice back to the user before posting it to the server. Is it at all possible to play it back to the user without getting into format conversions etc? Any ideas are welcome. Thanks!

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  • Android - Getting audio to play through earpiece

    - by Donal Rafferty
    I currently have code that reads a recording in from the devices mic using the AudioRecord class and then playing it back out using the AudioTrack class. My problem is that when I play it out it plays vis the speaker phone. I want it to play out via the ear piece on the device. Here is my code: public class LoopProg extends Activity { boolean isRecording; //currently not used AudioManager am; int count = 0; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.main); am = (AudioManager) getSystemService(Context.AUDIO_SERVICE); am.setMicrophoneMute(true); while(count <= 1000000){ Record record = new Record(); record.run(); count ++; Log.d("COUNT", "Count is : " + count); } } public class Record extends Thread { static final int bufferSize = 200000; final short[] buffer = new short[bufferSize]; short[] readBuffer = new short[bufferSize]; public void run() { isRecording = true; android.os.Process.setThreadPriority (android.os.Process.THREAD_PRIORITY_URGENT_AUDIO); int buffersize = AudioRecord.getMinBufferSize(11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT); AudioRecord arec = new AudioRecord(MediaRecorder.AudioSource.MIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize); AudioTrack atrack = new AudioTrack(AudioManager.STREAM_MUSIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize, AudioTrack.MODE_STREAM); am.setRouting(AudioManager.MODE_NORMAL,1, AudioManager.STREAM_MUSIC); int ok = am.getRouting(AudioManager.ROUTE_EARPIECE); Log.d("ROUTING", "getRouting = " + ok); setVolumeControlStream(AudioManager.STREAM_VOICE_CALL); //am.setSpeakerphoneOn(true); Log.d("SPEAKERPHONE", "Is speakerphone on? : " + am.isSpeakerphoneOn()); am.setSpeakerphoneOn(false); Log.d("SPEAKERPHONE", "Is speakerphone on? : " + am.isSpeakerphoneOn()); atrack.setPlaybackRate(11025); byte[] buffer = new byte[buffersize]; arec.startRecording(); atrack.play(); while(isRecording) { arec.read(buffer, 0, buffersize); atrack.write(buffer, 0, buffer.length); } arec.stop(); atrack.stop(); isRecording = false; } } } As you can see if the code I have tried using the AudioManager class and its methods including the deprecated setRouting method and nothing works, the setSpeatPoneOn method seems to have no effect at all, neither does the routing method. Has anyone got any ideas on how to get it to play via the earpiece instead of the spaker phone?

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  • Core-audio - constructing an AudioBufferList struct (Q about c struct definition)

    - by mustISignUp
    The definition of AudioBufferList looks weird to me… i guess my C is not so good struct AudioBufferList { UInt32 mNumberBuffers; AudioBuffer mBuffers[kVariableLengthArray]; }; typedef struct AudioBufferList AudioBufferList; Why AudioBuffer mBuffers[kVariableLengthArray]; and not AudioBuffer *mBuffers; ? kVariableLengthArray appears to be == 1. Eh? I think i have it working but would appreciate it if anyone could set me straight.

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  • Finding out estimated duration of a stream using Core Audio

    - by Reflog
    I am streaming a MP3 over network using custom feeding code, not AVAudioPlayer (which only works with URLs) using APIs like AudioFileStreamOpen and etc. Is there any way to estimate a length of the stream? I know that I can get a 'elapsed' property using: if(AudioQueueGetCurrentTime(queue.audioQueue, NULL, &t, &b) < 0) return 0; return t.mSampleTime / dataFormat.mSampleRate; But what about total duration to create a progress bar? Is that possible?

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