Sapi 5 1 Deutsch Youtube
Both versions of your plug-in fail to work in my configuration. When starting playback, I get a YASAPI error message that says 'AUDCLNTEUNSUPPORTEDFORMAT'.
Then when I click ok, to acknowledge the error message, Winamp crashes leaving 'winamp.exe' running in the background.I think this happens because I'm using another plug-in just ahead of yours in my processing chain. This other plug-in (Matrix Mixer) up-samples the decoded mp3 stereo input signal (16bits@44.1KHz) to 24bits@96KHz (to match my soundcard driver format settings) and up-mixes the stereo channels to 6 channels (to match my soundcard driver channel settings for my 5.1 speaker system).If I'm right, which part or parts of the input signal is/are not compatible with your plug-in?I can let Windows do the up-sampling and up-mixing (to match my soundcard driver settings), but it didn't sound as good as when doing the processing in Winamp. Of course I can change my soundcard driver settings to match the source material, but I'd rather use the full capabilities of my soundcard and speakers. The only WASAPI plug-in I've found that worked for me (and provided the up-sampling and up-mixing that I want) is the abandoned 'Maiko' output plug-in. I had to stop using Maiko because the last version released causes problems with the 'Jump to File Extra' plug-in.So I'm back to using Matrix Mixer and the official DirectSound output plug-in to get the processing that I want (what I was using before finding Maiko), but this combo only supports shared mode.
Maiko's exclusive mode sounded noticeably better and I was hoping your plug-in would too. Got up to 4.9x buffer and Bjork stopped popping during playback. Even though I use FLAC, I'm not really an audiophile.
So I am assuming the audio is not defected in any other obvious manner.EDIT: Are there any plans to have the volume slider in Winamp work with YASAPI? I noticed that the outasio plugin also does not have volume control. Is this just a limitation with these sorts of plugins?I increased my buffer to 5x as it was not playing at the default buffer size. It is working now - ploughing through my mp3 collections. No issues to report yet and using the sse2 version.
@all:Many thanks for the valuable feedback!Most likely the distortion and popping results from ring buffer underflow. That's why you could resolve this by simply enlarging the buffer. But there is not only the size which may lead to to the buffer drying out. The new version lets you configure some of the possiblities:You have the following relation between the parameters regarding buffer sizes:minimum size of the buffer shared with the device (provided by WASAPI).
Both versions of your plug-in fail to work in my configuration. When starting playback, I get a YASAPI error message that says 'AUDCLNTEUNSUPPORTEDFORMAT'.This is just a 1:1 report of an error resulting from a call to the method of the interface. It means that your audio device doesn't support the format (number of channels, sample frequency, bits per sample) you're going to play.For convenience, here are the links to the new 0.2.0 version:. Home and Documentation:.
Sourceforge Project:. Download. Both versions of your plug-in fail to work in my configuration. When starting playback, I get a YASAPI error message that says 'AUDCLNTEUNSUPPORTEDFORMAT'.This is just a 1:1 report of an error resulting from a call to the method of the interface. It means that your audio device doesn't support the format (number of channels, sample frequency, bits per sample) you're going to play.I get exactly the Same error Message, and the Maiko Wasapi plugin plays just fine. (using your inffsox too)My Soundcard is configured as 5.1 and my MP3's of course are Stereo,so in some way the error message would make sense,but for the stereo upmixing there is a function from Creative called CMSS3Dand even from Dolby called Dolby Digital Live or Dolby DTS.and even the maiko wasapi plugin provides such a functionality (I personly Like this one the most)I really like your inffsox plugin, it is working like a charm (except MP3 ID3 support )but this plugin does not work for me.
This is just a 1:1 report of an error resulting from a call to the method of the interface. It means that your audio device doesn't support the format (number of channels, sample frequency, bits per sample) you're going to play.How does your plug-in determine the audio device to use? I have more than one installed, but the one selected for Windows and in Winamp's DirectSound output plug-in (i.e. My soundcard) plays the up-sampled and up-mixed mp3 files just fine. I'm running Windows 7 and those links you provided are a bit over my head.
Maybe I should try shared mode, but that would defeat my purpose.I will wait for ravermeister's results before trying your latest version. Cleaning up after crashes is a bit of a hassle. I was under the impression that the Windows control panel sound utility displays the formats (bit depth & sampling rate) that the audio device's driver supports and that WASAPI would up-sample or down-sample the source to match the format selected. If it doesn't that would explain why the sound quality was different from the up/down sampling done by the various Winamp plug-ins that provide it.Windows also has a channel up-mixing feature (called speaker-fill, I think) if the audio driver supports it, but all this does is mirror the stereo channels to the other channels.Matrix mixer (and Maiko) allows true automatic or manual matrix mixing of the stereo channels to the output channels (6 in 5.1 or 8 in 7.1 speaker setups).
The stereo channels can be combined (in phase or either channel inverted to enhance or remove the identical sounds in each channel) at equal or various output levels and each combination directed to a particular output channel. Delays can be added to each channel as well.
Bottom line, the stereo source can be morphed into a semblance of true multichannel surround sound where each output channel can provide something different. The effect can be extended by adding frequency equalizing or other DPS effects (with other plug-ins or soundcard utilities) before or after the mixing.Movie soundtracks provide all this on their own, but it's fun playing sound engineer with stereo sources (making subtle or dramatic changes). That's the proof of what, exactly?Are you saying that your plug-in only accepts audio that is a PCM, 2 channel, 16bit, 44.1KHz signal? If not, how are the audio transforms to be accomplished? I don't understand what I'm supposed to do to have your plug-in work for me.
Am I supposed to remove everything in the processing chain except for the input decoders and your output plug-in?I'm looking for more than a bit perfect reproduction of the source file. If that is what your plug-in is designed to do, then that is prefectly fine for those who only want that. I guess I misunderstood the intent and it's just not suitable for me. So I'll stay with what I have and keep looking for another way to achive WASAPI exclusive mode support for up-sampled and up-mixed source files. Do we have to increase the size of the other buffers in line too (ring buffer, etc)?I dont't know because I have one PC to test and it works.However, we should imagine what's going on there:.
On one side there is WA delivering packets of sound at a rate which in mean represents the sample rate. The same is true for the WASAP side: it reads packets of another size and at another rate which also represents in mean the sample rate. We can conclude that in mean the ring buffer is empty. It just compensates for the different packet sizes. But there is still some probability that the buffer becomes empty, and we here glitches or pops. The mean number of frames in the buffer should never become zero.
You may achieve this by increasing the 'start playing.' The 'start playing.' Option means that when WASAPI takes the first packet from the buffer the number of frames in the buffer not becomes zero immediately (instead it approximately is the mean number of frames in the buffer).I increased the 'start playing.'
Parameter to 2 and it works (which means nothing because for me it's working anyway). I dont't know because I have one PC to test and it works.However, we should imagine what's going on there:. On one side there is WA delivering packets of sound at a rate which in mean represents the sample rate. The same is true for the WASAP side: it reads packets of another size and at another rate which also represents in mean the sample rate. We can conclude that in mean the ring buffer is empty. It just compensates for the different packet sizes.
But there is still some probability that the buffer becomes empty, and we here glitches or pops. The mean number of frames in the buffer should never become zero. You may achieve this by increasing the 'start playing.' The 'start playing.' Option means that when WASAPI takes the first packet from the buffer the number of frames in the buffer not becomes zero immediately (instead it approximately is the mean number of frames in the buffer).I increased the 'start playing.' Parameter to 2 and it works (which means nothing because for me it's working anyway).Thank you.
I will increase the 'start playing' parameter and see what this changes. Thanks for the prompt reply.
Got this error when trying to play a certain album of MP3s:Any idea what's up? No noticeable issues when YASAPI is disabled.When you get this error YASAPI is indeed enabled (no chance you get this when YASAPI is disabled).You get this under the same circumstances you get AUDCLNTEUNSUPPORTEDFORMAT. Both result from a call to the method of the interface:. when you get SFALSE you are most likely in shared mode, and.
when you get AUDCLNTEUNSUPPORTEDFORMAT you are most likely in exclusive mode.You should get rid of the SFALSE in shared mode when the format under Control Panel - Sound (in Windows) is configured to mach the format of your audio (especially the sample frequency). You should get rid of the SFALSE in shared mode when the format under Control Panel - Sound (in Windows) is configured to mach the format of your audio (especially the sample frequency).I always make sure the Windows Sound utility's format setting matches what is being output by Winamp. But as I said before, the format that is output does not match the format of the source file.
Does your plug-in read (and pass on) the format of the source file or the format that is passed to it by other plug-ins ahead of it in the processing chain? Ok, but the format is 24 bit, 96000 Hz, 6 channels which the Windows Sound configuration tool is setup to expect and it's working fine with the plug-ins I'm currently using (but they only support shared mode).I don't want to annoy you, but logically your plug-in should work for me, but it doesn't.
I'm just trying to understand why the Winamp DirectSound output plug-in accepts and passes on the signal to the WASAPI in an acceptable way and your output plug-in doesn't seem to.Is it possible the 24 bits are being truncated to 16? I'm trying to figure out what may work without trying all the possible permutations.
I'm afraid of what multiple crashes may do to my system. Is this coming from WA? If yes, how you are doing this?The configuration dialog for the DirectSound output plug-in has a status page that shows the output format it is using.
I use Matrix Mixer to do the up-sampling and up-mixing of the decoded input source and pass the resulting signal directly to a selectable output plug-in (in this case DirectSound, which is the default).A DSP plug-in (if used) would process the signal after the appropriate input plug-in and before Matrix Mixer. I don't use the Winamp equalizer and I'm not sure if it's processing is done before or after a DSP plug-in. I assume WASAPI would apply a soundcard's or mobo soundchip's equalizing and/or DSP signal processing utilities (I don't use them either) to the signal after it leaves Winamp.I sometimes use Stereo Tool , a DSP plug-in, to 'enhance' some of my lower quality mp3 files I've downloaded over the years. I don't need it for files I've ripped myself. What's new?. Option for treating mono as stereo (default switched on).
New default options for buffer configuration. Some fixes.Please note that the new default options for buffer configuration are tested with internet radio. With the old default options internet radio was prone to glitches and pops even on my side because of huge fluctuations of the stream. With the new options it seems that these fluctuations are buffered sufficiently.
Home and Documentation:. Sourceforge Project:. Download.
Sapi 5.1 Text-to-speech Engine
Speech SynthesisWhat is SAPI?The Speech API (SAPI) is an application programming interface developed by Microsoft to allow the use of speech synthesis within Windows applications. The SAPI provides a high-level interface between an application and speech engines. Text-To-Speech software synthesizes text strings and files into spoken audio using synthetic voices.Where can I get SAPI 4?To use SAPI 4 voices, download and install the redistributable Microsoft file.Also, you may download and install the Microsoft; the Speech Control Panel will add an icon to your Control Panel to enable you to list the compatible text-to-speech engines installed on your system and customize their settings for your use.Where can I get SAPI 5?Windows XP (and later versions) comes with SAPI 5. Speech EnginesHow do I know what Text-To-Speech voices have been installed on the computer?You can view all the voices available on the computer by following Control Panel - Speech - Speech Properties - Text To Speech - Voice selection.Windows 10 has the new voices Microsoft Mark Mobile and Microsoft Zira Mobile, but they are not available in text-to-speech software. Is it possible to unlock the new voices?By default, the Microsoft mobile voice is locked for using in text-to-speech software via SAPI 5.
You can unlock it with a simple registry tweak. Download the, extract the file for your language and for your version of the operating system ('mobilex86.reg' for 32-bit and 'mobilex64.reg' for 64-bit), click the right mouse button on the file's name and choose the context menu item 'Merge'.
The Microsoft mobile voice will appear in the list of the available voices in Balabolka.My computer is running the 64-bit version of Windows. I have installed the 64-bit Runtime package for and the English voices. But the list of available voices is still empty. Where is a problem?Balabolka is the 32-bit application.
You need to install the 32-bit Runtime package for Microsoft Speech Platform also.What is Google Text-To-Speech?In Google Translate you can find a 'Listen' button that converts text to speech. After pressing of this button a browser starts to download MP3 file. The service supports converting to speech texts not longer than 100 symbols. Balabolka allows to divide big text on small parts, create an audio file for each part and merge them together (WAV, MP3 and OGG formats are supported). SAPI TagsHow do I change a voice during reading aloud?Use the XML tags.
It is recommended to write the opening and closing tags inside the same paragraph. For example: Hello, how are you?I am good.I'm getting the error message ' OLE error 80045042'. What does that mean?This error means: 'The XML parser failed due to bad syntax.'
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Sapi 5 Voices
You must verify the syntax of the XML tags inside the text. Or, some part of the text looks like the start of an XML tag, and it is confusing SAPI. If you don't want to use XML tags, remove the symbols ' ' from the text (or replace them by words 'less than' and 'greater than'). Software UsageI open a DjVu file, but the program doesn't show any text.
What is wrong?DjVu format was designed to store scanned documents. A DjVu file contains images of pages for books, magazines, etc. Also, DjVu can contain an OCR text layer. Balabolka can extract data from a text layer of DjVu only. If such layer is not available, the only way to get text is to use a text recognition system (for example, FineReader).How can I remove dashes in the beginning of paragraphs?You should use the main menu item 'Edit Replace'. Type ^p— in the Find what box and ^p in the Replace with box. The program allows to use most of special codes from Microsoft Word:^p Paragraph mark ^t Tab character ^ nnn ASCII character (where nnn is the character code) ^0 nnn ANSI character (where 0 is zero and nnn is the character code) ^U nnnnn Unicode character (where nnnnn is the character code) ^+ Em dash ( — ) ^= En dash ( – ) ^^ Caret character ^s Nonbreaking space ^?
Any character ( in the Find what box only) ^# Any digit ( in the Find what box only) ^$ Any letter ( in the Find what box only) ^c Microsoft Windows Clipboard contents ( in the Replace with box only) ^& Contents of the Find what box ( in the Replace with box only)I am not satisfied, how Balabolka extracts text from PDF files. Can I use the other way for processing of PDF?The process of text extracting from PDF files is complicated enough, because PDF files do not contain plain text.
You may use an external command-line utility for text extracting: for example, the program pdftotext.exe from the project. Copy pdftotext.exe to the subfolder 'utils' in the folder of Balabolka, choose the main menu item 'Options Text Import', the tab 'Custom Text Import', and click the Add button. Define the command for using of pdftotext.exe:%BFolder%utilspdftotext.exe -q -nopgbrk -enc UTF-8%Input%%Output%Define the name of the converter (for example, 'Xpdf Converter'), file extension ('PDF') and output encoding ('UTF-8'). After activating of the option Use instead of default extracting method for this file type the program will be able to use the external utility for text extracting from PDF files. Audio FilesHow can every line of a text file be converted to a separate audio file?You may add two empty lines after every line with text in the document. Open the file in Balabolka, choose the main menu item 'Edit Replace', type ^p in the Find what box and ^p^p^p in the Replace with box, click the Replace all button. Choose the main menu item 'File Split and Convert to Audio Files'; choose the split method 'by two empty lines in succession' and click the Split and Convert button.Can I use the first line of text as the name of the audio file (for example, '01 Chapter One.mp3', '02 Chapter Two.mp3')?Type%FirstLine% in the 'Base Output Filename' box in the window 'Split and Convert to Audio Files'.
The application will replace this variable by the first line of each text part. To change the position of the sequence number inside the filename, use the%Number% variable. General Software QuestionsWhat is Spritz-Reader?is a a new speed-reading technology.
It allows to speed reading rates anywhere from 100 to 1000 words per minute. The time consuming part of usual reading lies mainly in the actual eye movements from word to word and sentence to sentence. Spritz positions words in a spot on a display where you can recognize the word, without moving your eyes.What is the IFilter interface?The IFilter interface works with documents. It provides filters to extract information from proprietary file formats. Full-text search engines call the methods of the IFilter interface. IFilters are available for Adobe PDF, WordPerfect and many other popular file formats.
You can get IFilter installers from respective vendors. Is a good place to get started. Note that because Balabolka is the 32-bit application you may need to install the 32-bit version of IFilters.Can I use media files generated with Balabolka in YouTube videos?Balabolka is freeware, so you may create audio files for free. But if you use the commercial voices, you need to contact the developers of the voice and purchase the license for commercial use/audio broadcast. The audio distribution license will allow you to use speech in YouTube videos.For example, the information from the web-site: 'Cepstral Personal voices are for personal use only and are NOT licensed for audio distribution. This means the audio you create is for your use only and cannot be shared with others or used in videos, presentations, or webpages.
If you are interested in an audio distribution license, please contact sales.' Is it possible to create a video file in Balabolka (containing speech as audio data and synchronized text as video data)?No, Balabolka doesn't have such option. But you can get the similar result in another way.My application allows to create files in and formats. Use the main menu item 'Options Audio Files' in Balabolka. On the tab 'LRC, SRT' you may define settings for subtitle files. The application will convert text to an audio file (for example, FILE.MP3) and create a subtitle file for this audio file (for example, FILE.SRT). You can play an audio file in three ways:.
in video player (for example, or ). in audio player (for example, with ). in karaoke software (for example, ). You can get a video file by using of the software.
It is a command-line tool that creates video files and converts video formats. Download the of the command file: it converts an audio file INPUT.WAV and a subtitle file INPUT.SRT to OUTPUT.MP4 with rendered text (all files must be at the same folder where FFMPEG.EXE is placed). It is a basic example: read the to get more information about the FFmpeg parameters.I am creating a 3D character, and it is necessary to generate a lip animation synchronized with audio generated by a text-to-speech engine.
Can you help me?When a speech engine reads aloud, it generates information about visemes. A viseme is the basic visual unit of speech that represents the position of the mouth and face when pronouncing a phoneme.
SAPI 5 supports the list of. The console application of Balabolka contains the that allows to generate the output text file with visemes.
The application will create the audio file and then read it aloud to get visemes and their timecodes.