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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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  • Testing for interface implementation in WCF/SOA

    - by rabidpebble
    I have a reporting service that implements a number of reports. Each report requires certain parameters. Groups of logically related parameters are placed in an interface, which the report then implements: [ServiceContract] [ServiceKnownType(typeof(ExampleReport))] public interface IService1 { [OperationContract] void Process(IReport report); } public interface IReport { string PrintedBy { get; set; } } public interface IApplicableDateRangeParameter { DateTime StartDate { get; set; } DateTime EndDate { get; set; } } [DataContract] public abstract class Report : IReport { [DataMember] public string PrintedBy { get; set; } } [DataContract] public class ExampleReport : Report, IApplicableDateRangeParameter { [DataMember] public DateTime StartDate { get; set; } [DataMember] public DateTime EndDate { get; set; } } The problem is that the WCF DataContractSerializer does not expose these interfaces in my client library, thus I can't write the generic report generating front-end that I plan to. Can WCF expose these interfaces, or is this a limitation of the serializer? If the latter case, then what is the canonical approach to this OO pattern? I've looked into NetDataContractSerializer but it doesn't seem to be an officially supported implementation (which means it's not an option in my project). Currently I've resigned myself to including the interfaces in a library that is common between the service and the client application, but this seems like an unnecessary extra dependency to me. Surely there is a more straightforward way to do this? I was under the impression that WCF was supposed to replace .NET remoting; checking if an object implements an interface seems to be one of the most basic features required of a remoting interface?

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  • Cast object to interface when created via reflection

    - by Al
    I'm trying some stuff out in Android and I'm stuck at when trying to cast a class in another .apk to my interface. I have the interface and various classes in other .apks that implement that interface. I find the other classes using PackageManager's query methods and use Application#createPackageContext() to get the classloader for that context. I then load the class, create a new instance and try to cast it to my interface, which I know it definitely implements. When I try to cast, it throws a class cast exception. I tried various things like loading the interface first, using Class#asSubclass, etc, none of which work. Class#getInterfaces() shows the interface is implemented. My code is below: PackageManager pm = getPackageManager(); List<ResolveInfo> lr = pm.queryIntentServices(new Intent("com.example.some.action"), 0); ArrayList<MyInterface> list = new ArrayList<MyInterface>(); for (ResolveInfo r : lr) { try { Context c = getApplication().createPackageContext(r.serviceInfo.packageName, Context.CONTEXT_IGNORE_SECURITY | Context.CONTEXT_INCLUDE_CODE); ClassLoader cl = c.getClassLoader(); String className = r.serviceInfo.name; if (className != null) { try { Class<?> cls = cl.loadClass(className); Object o = cls.newInstance(); if (o instanceof MyInterface) { //fails list.add((MyInterface) o); } } catch (ClassNotFoundException e) { // TODO Auto-generated catch block e.printStackTrace(); } // some exceptions removed for readability } } catch (NameNotFoundException e1) { e1.printStackTrace(); }

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  • .NET: Calling GetInterface method of Assembly obj with a generic interface argument

    - by Khnle
    I have the following interface: public interface PluginInterface<T> where T : MyData { List<T> GetTableData(); } In a separate assembly, I have a class that implements this interface. In fact, all classes that implement this interface are in separate assemblies. The reason is to architect my app as a plugin host, where plugin can be done in the future as long as they implement the above interface and the assembly DLLs are copied to the appropriate folder. My app discovers the plugins by first loading the assembly and performs the following: List<PluginInterface<MyData>> Plugins = new List<PluginInterface<MyData>>(); string FileName = ...;//name of the DLL file that contains classes that implement the interface Assembly Asm = Assembly.LoadFile(Filename); foreach (Type AsmType in Asm.GetTypes()) { //Type type = AsmType.GetInterface("PluginInterface", true); // Type type = AsmType.GetInterface("PluginInterface<T>", true); if (type != null) { PluginInterface<MyData> Plugin = (PluginInterface<MyData>)Activator.CreateInstance(AsmType); Plugins.Add(Plugin); } } The trouble is because neither line where I am getting the type as by doing Type type = ... seems to work, as both seems to be null. I have the feeling that the generic somehow contributes to the trouble. Do you know why?

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  • Playing NSF music in FMOD.net

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

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  • Playing NSF music in FMOD.net

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

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  • 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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  • Firefox 4 show tools menu in button interface

    - by chris.nullptr
    I like the extra vertical space provided by the latest firefox beta (4b9). However, the button interface doesn't give you access to the tools menu. I know that I can either temporarily enable the menubar by holding down the ALT key, but flipping a flag in about:config would be preferable. Does such a flag exist? Edit: I want to be able to access the tools menu from the new button interface like you can with Bookmarks, History, Options, and Help.

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  • Windows Server Shares web interface?

    - by Lazlow
    At work and home, I use Server 2003 with Windows Shares. We also have a couple of Netgear ReadyNAS appliances at work, managed via the web interface. Is there a built-in Web Interface for managing Shares, or a free one available - for use in Server 2003 (or 2008)? The reason I ask is that it would be good to manage the Shares via the Web, rather then RDP onto the Server each time - and would allow other users to treat it as a NAS.

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  • Using iptables to forward traffic destined for specific ip via specific interface

    - by shapeshifter
    I want to forward traffic destined for a specific ip from my internal network via a specific interface. I have two interfaces which are currently load balanced. I need all requests for a certain ip to go out via eth0 otherwise my external ip changes and sessions are dropped. eg. all requests from 10.1.1.1/24 to ip 11.22.33.44 on port 443 must go out via interface eth0. How can I do this with iptables?

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  • Conflicting ip routes with local table on attaching a virtual network interface

    - by user1071840
    I have an EC2 instance with these ip rules: $ sudo ip rule show 0: from all lookup local 32766: from all lookup main 32767: from all lookup default I can attach an elastic network interface to it with a private IP. Say the IP of my machine is 10.1.3.12 and the IP of the interface is 10.1.1.190. As soon as I attach the interface to my machine a new entry is added to the routing policy and local routing table: sudo ip rule show 0: from all lookup local 32765: from 10.1.1.190 lookup 10003 32766: from all lookup main 32767: from all lookup default $ sudo ip route show table local broadcast 10.1.1.0 dev eth3 proto kernel scope link src 10.1.1.190 local 10.1.1.190 dev eth3 proto kernel scope host src 10.1.1.190 broadcast 10.1.1.255 dev eth3 proto kernel scope link src 10.1.1.190 broadcast 10.1.3.0 dev eth0 proto kernel scope link src 10.1.3.12 local 10.1.3.12 dev eth0 proto kernel scope host src 10.1.3.12 broadcast 10.1.3.255 dev eth0 proto kernel scope link src 10.1.3.12 broadcast 127.0.0.0 dev lo proto kernel scope link src 127.0.0.1 local 127.0.0.0/8 dev lo proto kernel scope host src 127.0.0.1 local 127.0.0.1 dev lo proto kernel scope host src 127.0.0.1 broadcast 127.255.255.255 dev lo proto kernel scope link src 127.0.0.1 I can send traffic to this ENI directly from a host that can have the same IP as the host the ENI is attached to. This is where the problem starts. I ran tcpdump on the port in question and saw multiple SYNs going to the ENI with src '10.1.3.12' and destination '10.1.1.190' but didn't see even a single ACK. In my understanding if ACKs were being sent from the ENI they'd have destination as 10.1.3.12 i.e. the same as the local machine's IP and such packets will now be routed as local packets matching local routing policy: local 10.1.3.12 dev eth0 proto kernel scope host src 10.1.3.12 I'd like to send all the packets originating from 10.1.1.190 (my ENI) to go back on the same interface i.e. eth3 in this case. Contents of the nee table 10003 are: $ sudo ip route show table 10003 default via 10.1.1.1 dev eth3 I think I can do the following: I don't know if its possible but probably decrease the priority of local table so the packets match the table 10003. Use iptables to mangle these packets and update the local table route to include the mark information But I'm not sure if these are the right approaches.

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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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  • Custom flash mp3 player stopping in the middle of playing audio on windows nt ie6 system

    - by Charlotte Moller
    We have used a custom MP3 flash player for a lot of years on our website without any issues, but recently, a client of ours is reporting that the audio is playing for several seconds and then stopping. When they refresh the page or click play in the player again the audio plays fine. We are puzzled as to what could be causing this issue after this running successfully for our clients for so many years. The client system is Windows NT running IE6. Does anyone have any idea what could cause the audio to behave this way? Could audio drivers or the version of flash cause problems? We do not have flash programmers on our team so we are not even sure where to start looking within the flash code of the player. Any ideas?

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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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  • 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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  • 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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