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  • Media player control works only in IE (not FF/Chrome)

    - by user1323578
    I have media player control in my web page which that I able to get the media player only in Internet Explorer I can't able get in any other browser like Chrome or Firefox and code is media.aspx.cs: <%@ Page Language="C#" AutoEventWireup="true" CodeFile="Default.aspx.cs" Inherits="_Default" %> <%@ Register Assembly="Media-Player-ASP.NET-Control" Namespace="Media_Player_ASP.NET_Control" TagPrefix="cc1" %> <cc1:media_player_control id="Media_Player_Control1" runat="server" Height="314px" Width="518px" style="z-index: 100; left: 4px; position: absolute; top: 9px"> </cc1:media_player_control>

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  • Why is a FLAC encoded from a decoded MP3 bigger than the MP3?

    - by Ryan Thompson
    To be more precise than in the title, suppose I have a MP3 file that is 320 kbps. If I decompress it, then logically, all the data except for roughly 320 kilobits out of each second of audio should be redundant data, able to be compressed away. So, when I encode the decompressed file to FLAC, or any other lossless codec, why is it so much larger? On a related note, is it theoretically possible to losslessly recover the source mp3 audio from a decompressed wav? (I know the mp3 itself is lossy. I'm asking if it's possible to re-encode without any further loss.) EDIT: Let me clarify the related question, and the rationale behind it. Suppose I have a wav that was decompressed from an MP3 file (and assume I don't have the mp3 itself for some reason). If I don't want to lose any more quality, I can re-encode it with FLAC or any other lossless encoder and get a larger file just to maintain the same quality. Or, I can re-encode it to mp3 again and get the same size as the original but lose more data. Obviously, neither of these cases is ideal. I can either have the original size or the original quality, but not both (I mean the quality of the original mp3, not the original lossless source). My question is: Can we get both? Is it theoretically possible to recover the lossy compressed data from the lossy decompressed data, without losing even more? If it is possible, I could imagine a lossless compression algorithm that compresses the audio with FLAC. Then it also scans the audio for any signs of previous lossy compression, and if detected, recompresses it losslessly to the original lossy file. Then it keeps whichever file is smaller.

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  • Terrible noises from subwoofer of ACER Aspire 6930 with Realtek sound chip

    - by OneWorld
    After approximately 5-15 min of listening to music my subwoofer begins to make terrible noises. He's just "coughing". That began after 6 months I had this computer. Now I found out, that I can temporarily fix this problem by "restarting" the audio stream of the application that plays music. For example reloading last.fm page (reloads the flash file). Another way to reset the audio playback is switching the speaker configuration shown below in the screenshot. According to many posts on the internet like http://www.tomshardware.co.uk/forum/52918-20-acer-aspire-6935g-speaker-problem ACER support isn't any help Exchanging hardware doesn't fix the problem Even the later models have this problem Turning off the volume of the subwoofer is not an option to me. I still have warranty (I bought an extension of one year). I already tried about 15 versions of the Realtek driver with no success. I am not sure but MAYBE the problem did not occur on the original windows vista that was shipped with this computer. However, I removed the original windows for good reasons (english). What do you suggest me? Did anyone fix this problem? Maybe by writing a script which resets the audio streams every 5 minutes? Shall I take the effort to deal with the acer support until they give me another model? (I won't have a computer than for a longer time, will spend money on telephone hotlines (1,30 EUR / min)......) Here are additional infos, if they are any help: Windows 7 64 Bit (Original was Windows Vista Home Premium 32 Bit) All specs Audio driver version:

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  • How do I activate the F_LINE input in a transplanted HP chassis?

    - by admin
    I have an HP Pavilion Media Center PC chassis, vintage 2003 or so and I replaced the motherboard in it with a newer (vintage 2009) HP motherboard, M2N68-LA (Narra 5). I have scoured the internet trying to find pinouts for the motherboard to no avail. My question concerns the front panel audio, specifically Line In. The old chassis was built for AC97 but the new mobo is build for the newer HD audio standard. I figured out by comparison & experimentally how to connect the Mic & Headphone jacks to the HD audio header of the mobo by adding a manual switch to set the SENSE lines. Now all works fine for Mic & headphone. The old chassis also has a front panel Line In jack that the newer HP chassis does not have. However, the new mobo has a 4 pin white connector labeled F_LINE that I believe is a line input. Under Windows 7 I see the two Line Inputs in the mixer but I can't get one of them to become active. The 4 pin F_LINE connector uses the two middle pins for ground, and presumably the other two for left and right audio inputs. There are no pins for sensing on that connector. Can anyone tell me how to use that F_LINE input for the front panel, or how to activate it?

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  • Sound doesn't work anymore after replacing RAM

    - by thejh
    Hello, today, I replaced one old RAM module with two newer, bigger ones, but now, the sound doesn't seem to work anymore. Already ran alsaconf and it didn't help. Output of lspci for the audio device: 00:07.0 Audio device: nVidia Corporation MCP67 High Definition Audio (rev a1) Subsystem: Giga-byte Technology Device a002 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 (500ns min, 1250ns max) Interrupt: pin A routed to IRQ 21 Region 0: Memory at f5100000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Masking: 00000000 Pending: 00000000 Capabilities: [6c] HyperTransport: MSI Mapping Enable+ Fixed+ Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel The audio device is onboard and has six configurable outputs, two or so are also capable of being an input (if I remember it correctly), but I don't know how to control it under linux. Does somebody know how/whether replacing the RAM could be related to my problem and/or how to fix it?

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  • How to boost playback volume in real time on media recorded with a very low volume.

    - by L Marksman
    I have never heard a satisfactory answer to this often misunderstood question, let me explain. Lets say I have a sound card and earphones/speakers that can play back audio loud enough in most cases. This is great but the problem is that you always find people who do not know how to record audio, from Youtube video's to music. So now you end up with a audio playback that only uses 10% or less of the capacity of your sound hardware, in vista/win 7 you will see this frequently in the mixer with the volume pushed up to max but the green sound level only goes up a millimeter or two. I am looking for (preferably free) software or a method to boost the sound level of any audio from any source in real time to use more of my hardware capacity similar to what VLC media player can do. Oh and please, do not tell me it is impossible. I am not trying to boost the volume past what my hardware is capable of, I am just trying to use my hardware's full capacity. Also please do not tell met to buy new hardware, I know I can use hardware amplification, I don't want to (like many others) spend money on a simple little problem like this. Thanks!

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  • Oracle User Productivity Kit Translation

    - by ultan o'broin
    Oracle's customers just love the User Productivity Kit (UPK). I hear only great things about it from our international customers at the Oracle Usability Advisory Board meetings too. The UPK is the perfect solution for enterprise applications training needs (I previously reviewed a fine book about UPK btw). One question I am often asked is how source content created using the UPK can be translated into another language. I spoke with Peter Maravelias, Principal Product Strategy Manager for UPK about this recently. UPK is already optimized for easy source-target translation already. There is even a solution for re-recording demos. Here's what you can do to get your source content into another language: Use UPK's ability to automatically translate events and actions. UPK comes with XML templates that allow you to accomplish this in 21 languages with a simple publishing action switch. These templates even deal with the tricky business of using gender-based translations. Spanish localization template sample Japanese localization template sample Use the Import and Export localization features to export additional custom content in a format like XLIFF, easily handled by translation tools. You could also export and import in Word format. Re-record the sound (audio) files that go with the recordings, one per screen. UPK's granular approach to the sound files means that timing isn't an option. Retiming demos isn't required. A tip here with sound files and XLFF-exported custom content is to facilitate translation context by avoiding explicit references to actions going on in the screen recordings. A text based storyboard with screenshots accompanying the sound files should also be provided to the translators. Provide a glossary of terms too. Use the re-record option in UPK to record any demo from a translated application. This will allow all the translated UI labels to be automatically captured. You may be required to resize any action events here due to text expansion issues. Of course, you will need translated data in the translated application too, so plan for this in advance. However, source-target language skills aren't required for the re-recording. The UPK Player itself, of course, is also available from Oracle along with content and doc in 21 languages. The Developer and Setup is also translated in a smaller number of languages. Check the Oracle UPK website for latest details. UPK is a super solution for global enterprise applications training deployments allowing source content to be translated into multiple languages easily. See this post on the UPK blog for more insight too!

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  • Beat detection and FFT

    - by Quincy
    So I am working on a platformer game which includes music with beat detection. I am currently using a simple if the energy that is stored in the history buffer is smaller then the current energy there is a beat. The problem with this is that ofcourse if you use songs like rock songs where you have a pretty steady amplitude this isn't going to work. So I looked further and found algorithms splitting the sound into multiple bands using FFT. I then found this : http://en.literateprograms.org/Cooley-Tukey_FFT_algorithm_(C) The only problem I'm having is that I am quite new to audio and I have no idea how to use that to split the signal up into multiple signals. So my question is : How do you use a FFT to split a signal into multiple bands ? Also for the guys interested, this is my algorithm in c# : // C = threshold, N = size of history buffer / 1024 public void PlaceBeatMarkers(float C, int N) { List<float> instantEnergyList = new List<float>(); short[] samples = soundData.Samples; float timePerSample = 1 / (float)soundData.SampleRate; int sampleIndex = 0; int nextSamples = 1024; // Calculate instant energy for every 1024 samples. while (sampleIndex + nextSamples < samples.Length) { float instantEnergy = 0; for (int i = 0; i < nextSamples; i++) { instantEnergy += Math.Abs((float)samples[sampleIndex + i]); } instantEnergy /= nextSamples; instantEnergyList.Add(instantEnergy); if(sampleIndex + nextSamples >= samples.Length) nextSamples = samples.Length - sampleIndex - 1; sampleIndex += nextSamples; } int index = N; int numInBuffer = index; float historyBuffer = 0; //Fill the history buffer with n * instant energy for (int i = 0; i < index; i++) { historyBuffer += instantEnergyList[i]; } // If instantEnergy / samples in buffer < instantEnergy for the next sample then add beatmarker. while (index + 1 < instantEnergyList.Count) { if(instantEnergyList[index + 1] > (historyBuffer / numInBuffer) * C) beatMarkers.Add((index + 1) * 1024 * timePerSample); historyBuffer -= instantEnergyList[index - numInBuffer]; historyBuffer += instantEnergyList[index + 1]; index++; } }

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  • Flash Player error logs on Mac OS X

    - by paleozogt
    I'm on Mac OS X 10.5.8 running Flash Player 10,0,32,18. Flash Player is dumping giant amounts of error logging into the system log (stuff like "bit length overflow" and "code 0 bits 6-7"). Here's a tiny sampling: Oct 14 13:09:41 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:09:41 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 6 bits 6->7 Oct 14 13:09:41 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:09:41 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 5 bits 6->7 Oct 14 13:09:55 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:09:55 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 6 bits 6->7 Oct 14 13:09:55 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:09:55 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 5 bits 6->7 Oct 14 13:09:55 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:09:55 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 0 bits 6->7 Oct 14 13:10:06 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:10:06 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 4 bits 6->7 Oct 14 13:10:06 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:10:06 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 12 bits 6->7 Oct 14 13:10:20 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:10:20 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 6 bits 6->7 Oct 14 13:10:20 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:10:20 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 5 bits 6->7 Oct 14 13:10:21 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:10:21 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 0 bits 6->7 Oct 14 13:10:21 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:10:21 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 12 bits 6->7 Oct 14 13:10:31 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:10:31 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 7 bits 6->7 Oct 14 13:10:31 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:10:31 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 12 bits 6->7 Oct 14 13:11:06 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:11:06 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 3 bits 6->7 Oct 14 13:11:06 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:11:06 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 4 bits 6->7 Oct 14 13:11:06 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:11:06 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 3 bits 7->6 Oct 14 13:11:06 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 4 bits 5->6 Oct 14 13:11:07 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:11:07 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 3 bits 6->7 Oct 14 13:11:07 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:11:07 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 3 bits 6->7 Oct 14 13:11:15 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:11:15 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 3 bits 6->7 Oct 14 13:11:26 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Oct 14 13:11:26 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 7 bits 6->7 Oct 14 13:11:26 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 0 bits 4->5 Oct 14 13:11:26 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 14 bits 4->5 Oct 14 13:11:26 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 10 bits 5->4 Oct 14 13:11:26 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: code 4 bits 5->4 Oct 14 13:11:26 thorst-2 [0x0-0x58058].com.adobe.flash-10.0[2416]: bit length overflow Any ideas on what this may be about?

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  • Android: Voice Recording and saving audio

    - by user1320912
    I am working on application that will record the voice of the user and save the file on the SD card and then allow the user to listen to the audio again. I am able to allow the user to record his voice using the RecognizerIntent, but I cant figure out how to save the audio file and allow the user to hear the audio. I would appreciate it if someone could help me out. I have displayed my code below: // Setting up the onClickListener for Audio Button attachVoice = (Button) findViewById(R.id.AttachVoice_questionandanswer); attachVoice.setOnClickListener(new OnClickListener() { public void onClick(View v) { Intent voiceIntent = new Intent(RecognizerIntent.ACTION_RECOGNIZE_SPEECH); voiceIntent.putExtra(RecognizerIntent.EXTRA_LANGUAGE_MODEL, RecognizerIntent.LANGUAGE_MODEL_FREE_FORM); voiceIntent.putExtra(RecognizerIntent.EXTRA_PROMPT, "Please Speak"); startActivityForResult(voiceIntent, VOICE_REQUEST); } }); protected void onActivityResult(int requestCode, int resultCode, Intent data) { if(requestCode == VOICE_REQUEST && resultCode == RESULT_OK){ }

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  • Ubuntu 12.04 - No sound - HELP!

    - by Bruno Tacca
    I'm panicking... my sound stopped working after I tried to set-up my notebook speakers, plus two headphone jacks... My idea was to multichannel the sound to 3 channels, built-in speakers, and sound-card 2 headphone jacks. After a couple efforts I did it with 2 channels, speakers and 1 headphone jack, but the other wasn't working. After more tries and tries, sound stop working. I just want my sound back... crying like a baby on the floor. And, if possible, but not necessary, a simple guide to active the 3 channels. xD I will post the diagnosis according to https://help.ubuntu.com/community/SoundTroubleshootingProcedure STEP 1 Did it, still no sound. STEP 2 Did it, still no sound. STEP 3 and #STEP 4 (I removed the log cause there is a limit of characters to be posted.) The log can be found here: https://answers.launchpad.net/ubuntu/+source/alsa-driver/+question/238653 STEP 5 Rebooted, still no sound. STEP 6 Did it. In the Output Devices tab, nothing is muted. I play a music with the Rhythmbox Music Player, I don't hear anything but in the pavucontrol I can see in the Built-in Audio Analog Stereo a sound bar shaking... but, no sound. STEP 7 In alsamixer, AlsaMixer v1.0.25 Card: HDA Intel PCH Chip: Creative CA0132 information View: F3:[Playback] F4: Capture F5: All Item: Headphone [dB gain: 25.00, 25.00] Then, I have 5 columns Headphone, Speaker, PCM, S/PDIF, S/PDIF Default PCM A little weird when I try to mute the Headphone and the Speaker, here what happens: Starting both unmutted, mutting headphone cause speaker being mutted automaticaly. Starting both unmutted, mutting speaker cause headphone being mutted automaticaly. Starting both mutted, possible to unmute both separately. STEP 8 I cannot hear sound on both (headphone and/or speaker). STEP 9 Dual boot... Restarted, windows was with sound at max volume. Restarted again, still no sound at ubuntu. I heard something when ubuntu started, a little noise, then silence again. The sound icon always start mutted, after unmutting, I have no sound. STEP 10 I dont have this command in my ubuntu. STEP 11 Tried at STEP 8, no sound. There are no problem with jumpers or hardware, cause I have sound working on windows. STEP 12 No way to open my alienware and loss the warranty x.X" STEP 13 I think it's loaded, judging my the logs STEP 14 Alienware M17xR4, the hardware is listed in the logs above, at STEP 4. There are two headphone hacks, one with just an headphone printed above, and the other with an headset (with mic) printed, there is a mic jack too, and a spdif (optical) too. STEP 15 I dont want to enable S/PDIF STEP 16 I never used the HDMI output, yet... Thanks in advance. I hope I listed all the information you need.

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  • Testing background audio in the simulator

    - by Cactuar
    I'm experimenting with the new background audio service in iPhone OS 4.0 but I can't get it to work in the simulator. According to this page: iPhone Application Programming Guide: Executing Code in the Background it seems that all I have to do is add the a UIBackgroundModes key with an array containing audio to my Info.plist file and the audio my application plays should automatically continue when I switch to another app. I have done this but the audio still pauses as I switch to another app, when I switch back it continues where it left off. This is the code I'm using to play the sound: NSURL *url = [NSURL fileURLWithPath:[NSString stringWithFormat:@"%@/audio.mp3", [[NSBundle mainBundle] resourcePath]]]; NSError *error; audioPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:url error:&error]; audioPlayer.numberOfLoops = -1; if (audioPlayer == nil) NSLog(@"%@", [error userInfo]); else [audioPlayer play]; Has anyone gotten this to work? Could it be that it would work on an actual device and it's just a problem with the simulator? I'm a bit hesitant to install 4.0 on my phone since I've heard it's still very buggy. Wish I had another device to use only for development.

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  • Capture Flash Audio in 4.7 Edge?

    - by emcmanus
    Is there a way to capture plugin (Flash) audio before it gets to the sound card? I'd like to record plugin audio, hopefully without actually playing the sound. Capturing audio at the device level is an absolute last resort, as the application would pick up all system audio rather than just the Webkit plugin. I'm aware of the recent switch back from QTMultimedia; is this possible with phonon? I spent the night looking for some way to access the phonon graph via QWebFrame (or any of the QtWebkit widgets) -- and didn't turn up much. I also started digging through QTWebkit, particularly NPAPI, without success. For reference, I'm using the edge version of 4.7 (6aa50af000f85cc4497749fcf0860c8ed244a60e) This seems to be a fairly challenging problem. Any hints would be greatly appreciated.

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  • Audio queue start failed

    - by mobapps99
    Hi , i'm developing a project which has both audio streaming and playing audio from file. For audio streaming i'm using AudioStreamer and for playing from file i'm using avaudioplayer. Both streaming and playing works perfectly as long as the app is not interrupted by a phone call or sms. But when a call/sms comes after dismissing the call when i try to restart streaming i'm getting the error "Audio queue start failed" . This happens only when i have used avaudioplayer at least once and after that used streaming. When the avaudioplayer obeject is not created , in this scenario the there is no problem with resuming streaming after dismissing the call. My guess is that some thing is wrong with audioqueue. Help is very much appreciated.......

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  • Making a DVD video with a still image and PCM 16bit audio with ffmpeg

    - by João
    I'm trying to make a small video with a still image and a sound file playing in the background to pass it to dvdauthor and create a DVD. The command I'm using is this: ffmpeg -loop_input -i image.jpg -qscale 2 -i song.flac -aspect 4:3 -target pal-dvd -acodec pcm_s16le -shortest output.mpg However, the resulting video file doesn't have sound at all (testing it on VLC Player). I don't know if I can't combine "-acodec pcm_s16le" with "-target pal-dvd" to override the later, or if there is something else wrong with the command. If I try without the "-acodec pcm_s16le" parameter the video and audio works, I can even create a DVD ISO with it. However, the audio stays as AC3. I wanted to include with the video the lossless audio, not a compressed one. I suppose the DVD standart allows to have PCM audio in it, am I right?

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  • How to get the default audio format of a TTS Engine

    - by Itslava
    In Microsoft TTS 5.1 or newer. The SpVoice.AudioOutputStream property says: The AudioOutputStream property gets and sets the current audio stream object used by the voice. Setting the voice's AudioOutputStream property may cause its audio output format to be automatically changed to match the text-to-speech (TTS) engine's preferred audio output format. If the voice's AllowAudioOutputFormatChangesOnNextSet property is True, the format change takes place; if False, the format remains unchanged. In order to set the AudioOutputStream property of a voice to a specific format, its AllowOutputFormatChangesOnNextSet should be False. It means a engine's always has a preferred audio output format. So, how can i get it.. i have not found any interface to get that attribute.

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  • Calculate new position of player

    - by user1439111
    Edit: I will summerize my question since it is very long (Thanks Len for pointing it out) What I'm trying to find out is to get a new position of a player after an X amount of time. The following variables are known: - Speed - Length between the 2 points - Source position (X, Y) - Destination position (X, Y) How can I calculate a position between the source and destion with these variables given? For example: source: 0, 0 destination: 10, 0 speed: 1 so after 1 second the players position would be 1, 0 The code below works but it's quite long so I'm looking for something shorter/more logical ====================================================================== I'm having a hard time figuring out how to calculate a new position of a player ingame. This code is server sided used to track a player(It's a emulator so I don't have access to the clients code). The collision detection of the server works fine I'm using bresenham's line algorithm and a raycast to determine at which point a collision happens. Once I deteremined the collision I calculate the length of the path the player is about to walk and also the total time. I would like to know the new position of a player each second. This is the code I'm currently using. It's in C++ but I am porting the server to C# and I haven't written the code in C# yet. // Difference between the source X - destination X //and source y - destionation Y float xDiff, yDiff; xDiff = xDes - xSrc; yDiff = yDes - ySrc; float walkingLength = 0.00F; float NewX = xDiff * xDiff; float NewY = yDiff * yDiff; walkingLength = NewX + NewY; walkingLength = sqrt(walkingLength); const float PI = 3.14159265F; float Angle = 0.00F; if(xDes >= xSrc && yDes >= ySrc) { Angle = atanf((yDiff / xDiff)); Angle = Angle * 180 / PI; } else if(xDes < xSrc && yDes >= ySrc) { Angle = atanf((-xDiff / yDiff)); Angle = Angle * 180 / PI; Angle += 90.00F; } else if(xDes < xSrc && yDes < ySrc) { Angle = atanf((yDiff / xDiff)); Angle = Angle * 180 / PI; Angle += 180.00F; } else if(xDes >= xSrc && yDes < ySrc) { Angle = atanf((xDiff / -yDiff)); Angle = Angle * 180 / PI; Angle += 270.00F; } float WalkingTime = (float)walkingLength / (float)speed; bool Done = false; float i = 0; while(i < walkingLength) { if(Done == true) { break; } if(WalkingTime >= 1000) { Sleep(1000); i += speed; WalkTime -= 1000; } else { Sleep(WalkTime); i += speed * WalkTime; WalkTime -= 1000; Done = true; } if(Angle >= 0 && Angle < 90) { float xNew = cosf(Angle * PI / 180) * i; float yNew = sinf(Angle * PI / 180) * i; float NewCharacterX = xSrc + xNew; float NewCharacterY = ySrc + yNew; } I have cut the last part of the loop since it's just 3 other else if statements with 3 other angle conditions and the only change is the sin and cos. The given speed parameter is the speed/second. The code above works but as you can see it's quite long so I'm looking for a new way to calculate this. btw, don't mind the while loop to calculate each new position I'm going to use a timer in C# Thank you very much

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  • Access MP3 audio data independently of ID3 tags?

    - by kyl191
    Hi, this is a 2 part question. First off, is it possible to access the audio data in an MP3 independently of the ID3 tags, and secondly, is there any way to do so using available libraries? I recently consolidated my music collection from 3 computers and ended up with songs which had changed ID3 tags, but the audio data itself was unmodified. Running a search for duplicate files failed because the file changed with the ID3 tag change, but I think it should be possible to identify duplicate files if I just run a deduplication using the audio data for comparison. I know that it's possible to seek to a particular position past the ID3 header in the file, and directly read the data, but was wondering if there's a library that would expose the audio data so I could just extract the data, run a checksum on it, and store the computed result somewhere, then look for identical checksums. (Also, I'd probably have to use some kind of library when you take into account variable length headers.)

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  • Simple thruster like behaviour when rotating sprite

    - by ensamgud
    I'm prototyping some 2D game concepts with XNA and have added some basic keyboard inputs to control a triangle sprite. When I press key up the sprite accelerates in it's current facing direction, when I release the key it brakes down. For rotation, when I press left/right keys I rotate the sprite. Currently the sprite immedately changes direction when I rotate it. What I want is for it to keep moving in the same direction when I rotate, until I hit key up, adding thrust in whatever direction the sprite is pointing at. This would simulate thrusters on a classic space shooter like Asteroids. I'm adding an image to describe the behaviour I'm after and some code samples of how I'm doing things at the moment. This is my player struct, holding information of the sprite. public struct PlayerData { public Vector2 Position; // where to draw the sprite public Vector2 Direction; // travel direction of sprite public float Angle; // rotation of sprite public float Velocity; public float Acceleration; public float Decelleration; public float RotationAcceleration; public float RotationDecceleration; public float TopSpeed; public float Scale; } This is how I'm currently handling thrusting / braking (when pressing/releasing key up) (simplified, removed some bounds checking etc): player.Velocity += player.Acceleration * 0.1f; player.Velocity -= player.Acceleration * 0.1f; And when I rotate the sprite left and right: player.Angle -= player.RotationAcceleration * 0.1f; player.Angle += player.RotationAcceleration * 0.1f; This runs in the update loop, keeps the direction updated and updates the position: Vector2 up = new Vector2(0f, -1f); Matrix rotMatrix = Matrix.CreateRotationZ(player.Angle); player.Direction = Vector2.Transform(up, rotMatrix); player.Direction *= player.Velocity; player.Position += player.Direction; I am following along various beginner tutorials and haven't found any describing this, but I have tried some on my own without success. Do I need to change my velocity and acceleration fields to Vectors instead of floats to accomplish this type of movement? I realise my Angle and the Direction vector is currently tied together and I need to disconnect these somehow to be able to rotate freely without changing the direction of the movement, but I can't quite figure out how to do this while keeping the acceleration/decceleration functional. Would appreciate an explanation rather than pure code samples. Thanks,

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  • video is not playing on ipad(device) when i use a separate audio track with it

    - by sujith1406
    In my application i need to play a video(silent ) together with another audio.i am using mpmovieplayercontroller for video and avaudioplayer for audio.the problem is on device (for ipad) the video is not playing .it is working perfect on ipad and iphone simulator .also on iphone .i am using ipad (os 3.2 ) installed.why is this so?? this is the code i am using NSString *trackname=[dict objectForKey:@"AudioFile"]; NSLog(@"track--->%@",trackname); NSString *newAudioFile = [[NSBundle mainBundle] pathForResource:trackname ofType:@"mp4"]; player = [[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:newAudioFile] error:NULL]; if(error) { NSLog(@"%@",&error); } [player prepareToPlay]; [player play]; player.numberOfLoops=0; slider.maximumValue=player.duration; slider.minimumValue=0; [playpausebtn setBackgroundImage:[UIImage imageNamed:@"pausebutton.png"] forState:UIControlStateNormal]; timer = [NSTimer scheduledTimerWithTimeInterval:0.5 target:self selector:@selector(updateSlider) userInfo:nil repeats:YES]; NSString *videoFile = [[NSBundle mainBundle] pathForResource:@"video-track" ofType:@"mp4"]; moviePlayer=[[MPMoviePlayerController alloc]initWithContentURL:[NSURL fileURLWithPath:videoFile]]; [[moviePlayer view] setFrame: [videoView bounds]]; // frame must match parent view [videoView addSubview: [moviePlayer view]]; [videoView setBackgroundColor:[UIColor blackColor]]; moviePlayer.repeatMode=MPMovieRepeatModeOne; moviePlayer.controlStyle=MPMovieControlStyleNone; [moviePlayer play];

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  • How can I add a character and enemies to a game that uses Parallax Scrolling? [on hold]

    - by Homer_Simpson
    I use the following code to create Parallax Scrolling: http://www.david-gouveia.com/portfolio/2d-camera-with-parallax-scrolling-in-xna/ Parallax Scrolling is working but I don't know how to add the player and the enemies. I tried to add a player class to the existing code, but if the player moves, then the camera isn't pointing at the player. The player leaves the camera viewport after a few seconds. I use the following code(as described in the tutorial), but it's not working: // Updates my camera to lock on the character _camera.LookAt(player.Playerposition); What can I do so that the player is always the center of the camera? How should I add the character and the enemies to the game? Should I create a layer for the character and the enemies? For example: new Layer(_camera) { Parallax = new Vector2(0.9f, 1.0f) } At the moment, I don't use a layer for the player and I don't have implemented the enemies because I don't know how to do that. My player class: public class Player { Texture2D Playertex; public Vector2 Playerposition = new Vector2(400, 240); private Game1 game1; public Player(Game1 game) { game1 = game; } public void Load(ContentManager content) { Playertex = content.Load<Texture2D>("8bitmario"); TouchPanel.EnabledGestures = GestureType.HorizontalDrag; } public void Update(GameTime gameTime) { while (TouchPanel.IsGestureAvailable) { GestureSample gs = TouchPanel.ReadGesture(); switch (gs.GestureType) { case GestureType.HorizontalDrag: Playerposition.X += 3f; break; } } } public void Render(SpriteBatch batch) { batch.Draw(Playertex, new Vector2(Playerposition.X - Playertex.Width / 2, Playerposition.Y - Playertex.Height / 2), Color.White); } } In Game1, I update the player and camera class: protected override void Update(GameTime gameTime) { // Updates my character's position player.Update(gameTime); // Updates my camera to lock on the character _camera.LookAt(player.Playerposition); base.Update(gameTime); } protected override void Draw(GameTime gameTime) { GraphicsDevice.Clear(Color.CornflowerBlue); foreach (Layer layer in _layers) layer.Draw(spriteBatch); spriteBatch.Begin(SpriteSortMode.Deferred, null, null, null, null, null, _camera.GetViewMatrix(new Vector2(0.0f, 0.0f))); player.Render(spriteBatch); spriteBatch.End(); base.Draw(gameTime); }

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  • Ubuntu 11.04 VM shows a black screen in VMware Player

    - by Roel Veldhuizen
    I have a Ubuntu Server 11.04 64 bit VM running on VMware Player 3.1.4 that only shows a black screen. No matter what I try, the screen remains black. The VM has worked the first time. When I reset the machine, it shows the VMware loader and a flickering _ for about a second. Then the screen turns black again. VM settings: Memory: 512MB Processors: 1 HD: 20GB CD: auto detect Floppy: auto detect Network adapter: NAT USB controller: present soundcard: auto detect printer: present display: auto detect I just created a fresh VM and the same happens, so it seems that the problem is consistent.

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  • MP3 player no longer syncing

    - by zildjohn01
    I recently installed third-party drivers for the (Sony) PS3 controller on my friend's PC (Windows XP). I found out a few days later that his MP3 player (also Sony) is no longer recognized by Windows. He gets the "connect device" sound, and about 250ms later, the "disconnect device" sound. I figured the controller driver took over the Walkman's device ID, so I went through the registry and C:\Windows\inf removing all references to Sony's VID (054C), but I haven't had any luck. What would you do in this situation?

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