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Search found 496 results on 20 pages for 'wav'.

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  • Track status of Microsoft TTS output to wav file

    - by user325478
    I'm trying to track the status of my applications TTS output to a wav file. When speaking the text (to the speaker) the expected events (StartStream, Word, EndStream) are fired, however, no events are fired when outputing to a wave file. SpVoice vox = new SpVoice(); vox.Word += VoxWord; // Handle word processed event SpFileStream voxStream = new SpFileStream(); voxStream.Open(@"c:\test.wav", SpeechStreamFileMode.SSFMCreateForWrite, false); vox.AudioOutputStream = voxStream; vox.Speak("Hello World. Please track my status!", SpeechVoiceSpeakFlags.SVSFlagsAsync); Is it possible to asynchronously know the status of TTS output to wav?

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  • C# - .WAV Playback Randomly High Pitch

    - by Nate Shoffner
    For some reason, when a WAV file is played back using the snippet below, it randomly plays back screwy, like a high pitch noise. It doesn't happen all the time, just randomly. It seems to happen more often when it is played back more frequently. The WAV properties are below along with the code snippet I am using. WAV Properties: Bit Rate - 750kbps Audio Sample Size - 16 bit Channels - 1 (mono) Audio Sample Rate - 44kHz Audio Format - PCM Snippet: System.Media.SoundPlayer myPlayer = new System.Media.SoundPlayer(Captcha.Properties.Resources.sound1); myPlayer.Play(); Is this because of the way I am playing the file or the file itself? Thank you.

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  • Playing a .wav sound file in JPanel/JFrame using javax (Swing)

    - by JavaIceCream
    I need some code example on how I would use a filepath from a harddrive location to then play a .wav sound file when opened in swing GUI. I don't need it to show a play button, or pause or stop. I just want it to play when I select the 'Sound' option from my 'Files' in my window (I know how to do that already, no need to explain that). So basically, just how to play a .wav sound file from a filepath (i.e. c:/cake/thereisnone.wav) inside of a JFrame. And how can I easily apply methods to that sound file afterwards. Also, if anyone knows how to apply methods on a BufferedImage in a JFrame, that would be helpful too. Thank you very much everyone!

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  • AMR to WAV converter in JAVA

    - by sohilvassa
    Hello friends Is there amr to wav , wav to amr converter available written java? i need to do conversion in realtime.So i need JAVA API for server. Looking for any of this option open sourced or free or paid. Thanks

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  • How to open wav file with Lua

    - by Pete Webbo
    Hello, I am trying to do some wav processing using Lua, but have fallen a the first hurdle! I cannot find a function or library that will allow me to load a wav file and access the raw data. There is one library, but it onl allows playing of wavs, not access to the raw data. Are there any out there? Cheers, Pete.

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  • Wav analysis in python

    - by mudder
    I'm looking for a python library that will help me analyze the audio in wav files. At the very least I'm hoping to find some kind of interface that understands .wav format so that I don't have to :P at best I need a module with methods for reading wave form parameters like pitch, volume levels, etc

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  • Converting WAV to MP3 on Linux with low bitrates

    - by Olly
    I need to convert WAV files to MP3 files so they can be played on a website. I think that LAME would probably be the best tool. However the WAV files are low bitrate (around 8kbits recorded from a phone) and LAME's website states that it is the "best MP3 encoder at mid-high bitrates and at VBR". Is there is a better encoder for lower bitrates? If so can you define "better"?

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  • Playing a .WAV file in .NET

    - by Mori
    I'm trying to write a SAMPLER program, where each key has a different sound (a WAV file). Can someone explain to me or give me a link to an explanation where i can learn how to play the WAV files? If it matters, I'm working with Microsoft Visual C# and using WinForms.

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  • wav file manupalation

    - by kaushik
    I want get the details of the wave such as its frames into a array of integers. Using fname.getframes we can ge the properties of the frame and save in list or anything for writing into another wav or anything,but fname.getframes gives information not in integers some thing like a "/xt/x4/0w' etc.. But i want them in integer so that would be helpful for manupation and smoothening join of 2 wav files

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  • Need help manipulating WAV (RIFF) Files at a byte level

    - by Eric
    I'm writing an an application in C# that will record audio files (*.wav) and automatically tag and name them. Wave files are RIFF files (like AVI) which can contain meta data chunks in addition to the waveform data chunks. So now I'm trying to figure out how to read and write the RIFF meta data to and from recorded wave files. I'm using NAudio for recording the files, and asked on their forums as well on SO for way to read and write RIFF tags. While I received a number of good answers, none of the solutions allowed for reading and writing RIFF chunks as easily as I would like. But more importantly I have very little experience dealing with files at a byte level, and think this could be a good opportunity to learn. So now I want to try writing my own class(es) that can read in a RIFF file and allow meta data to be read, and written from the file. I've used streams in C#, but always with the entire stream at once. So now I'm little lost that I have to consider a file byte by byte. Specifically how would I go about removing or inserting bytes to and from the middle of a file? I've tried reading a file through a FileStream into a byte array (byte[]) as shown in the code below. System.IO.FileStream waveFileStream = System.IO.File.OpenRead(@"C:\sound.wav"); byte[] waveBytes = new byte[waveFileStream.Length]; waveFileStream.Read(waveBytes, 0, waveBytes.Length); And I could see through the Visual Studio debugger that the first four byte are the RIFF header of the file. But arrays are a pain to deal with when performing actions that change their size like inserting or removing values. So I was thinking I could then to the byte[] into a List like this. List<byte> list = waveBytes.ToList<byte>(); Which would make any manipulation of the file byte by byte a whole lot easier, but I'm worried I might be missing something like a class in the System.IO name-space that would make all this even easier. Am I on the right track, or is there a better way to do this? I should also mention that I'm not hugely concerned with performance, and would prefer not to deal with pointers or unsafe code blocks like this guy. If it helps at all here is a good article on the RIFF/WAV file format.

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  • How to extract semi-precise frequencies from a WAV file using Fourier Transforms

    - by Seisatsu
    Let us say that I have a WAV file. In this file, is a series of sine tones at precise 1 second intervals. I want to use the FFTW library to extract these tones in sequence. Is this particularly hard to do? How would I go about this? Also, what is the bast way to write tones of this kind into a WAV file? I assume I would only need a simple audio library for the output. My language of choice is C

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  • Create mp3 previews from wav and aiff files

    - by August Lilleaas
    I would like to create a program that makes mp3s of the first 30 seconds of an aiff or wav file. I would also like to be able to choose location and length, such as the audio between 2:12 and 2:42. Are there any tools that lets me do this? Shelling out is OK. The application will run on a linux server, so it would have to be a tool that works on linux. I don't mind doing it in two steps - i.e. a tool that first creates the cutout of the aiff/wav, then pass it to a mp3 encoder.

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  • How to stream a WAV file?

    - by jonasb
    I'm writing an app where I record audio and upload the audio file over the web. In order to speed up the upload I want to start uploading before I've finished recording. The file I'm creating is a WAV file. My plan was to use multiple data chunks. So instead of the normal encoding (RIFF, fmt , data) I’m using (RIFF, fmt , data, data, ..., data). The first issue is that the RIFF header wants the total length of the whole file, but that is of course not known when streaming the audio (I’m now using an arbitrary number). The other problem is that I'm not sure if it's valid since Audacity doesn't recognise the file, and Windows Media Player opens the file but plays only a very small part. I've been reading WAV specs but haven’t found an answer. Any suggestions?

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  • processing an audio wav file with C

    - by sa125
    Hi - I'm working on processing the amplitude of a wav file and scaling it by some decimal factor. I'm trying to wrap my head around how to read and re-write the file in a memory-efficient way while also trying to tackle the nuances of the language (I'm new to C). The file can be in either an 8- or 16-bit format. The way I thought of doing this is by first reading the header data into some pre-defined struct, and then processing the actual data in a loop where I'll read a chunk of data into a buffer, do whatever is needed to it, and then write it to the output. #include <stdio.h> #include <stdlib.h> typedef struct header { char chunk_id[4]; int chunk_size; char format[4]; char subchunk1_id[4]; int subchunk1_size; short int audio_format; short int num_channels; int sample_rate; int byte_rate; short int block_align; short int bits_per_sample; short int extra_param_size; char subchunk2_id[4]; int subchunk2_size; } header; typedef struct header* header_p; void scale_wav_file(char * input, float factor, int is_8bit) { FILE * infile = fopen(input, "rb"); FILE * outfile = fopen("outfile.wav", "wb"); int BUFSIZE = 4000, i, MAX_8BIT_AMP = 255, MAX_16BIT_AMP = 32678; // used for processing 8-bit file unsigned char inbuff8[BUFSIZE], outbuff8[BUFSIZE]; // used for processing 16-bit file short int inbuff16[BUFSIZE], outbuff16[BUFSIZE]; // header_p points to a header struct that contains the file's metadata fields header_p meta = (header_p)malloc(sizeof(header)); if (infile) { // read and write header data fread(meta, 1, sizeof(header), infile); fwrite(meta, 1, sizeof(meta), outfile); while (!feof(infile)) { if (is_8bit) { fread(inbuff8, 1, BUFSIZE, infile); } else { fread(inbuff16, 1, BUFSIZE, infile); } // scale amplitude for 8/16 bits for (i=0; i < BUFSIZE; ++i) { if (is_8bit) { outbuff8[i] = factor * inbuff8[i]; if ((int)outbuff8[i] > MAX_8BIT_AMP) { outbuff8[i] = MAX_8BIT_AMP; } } else { outbuff16[i] = factor * inbuff16[i]; if ((int)outbuff16[i] > MAX_16BIT_AMP) { outbuff16[i] = MAX_16BIT_AMP; } else if ((int)outbuff16[i] < -MAX_16BIT_AMP) { outbuff16[i] = -MAX_16BIT_AMP; } } } // write to output file for 8/16 bit if (is_8bit) { fwrite(outbuff8, 1, BUFSIZE, outfile); } else { fwrite(outbuff16, 1, BUFSIZE, outfile); } } } // cleanup if (infile) { fclose(infile); } if (outfile) { fclose(outfile); } if (meta) { free(meta); } } int main (int argc, char const *argv[]) { char infile[] = "file.wav"; float factor = 0.5; scale_wav_file(infile, factor, 0); return 0; } I'm getting differing file sizes at the end (by 1k or so, for a 40Mb file), and I suspect this is due to the fact that I'm writing an entire buffer to the output, even though the file may have terminated before filling the entire buffer size. Also, the output file is messed up - won't play or open - so I'm probably doing the whole thing wrong. Any tips on where I'm messing up will be great. Thanks!

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  • Trouble converting an MP3 file to a WAV file using Naudio

    - by WebDevHobo
    Naudio Library: http://naudio.codeplex.com/ I'm trying to convert an MP3 file to a WAV file, but I've run in to a small error. I know what's going wrong, but I don't really know how to go about fixing it. Here's the piece of code I'm running: private void button1_Click(object sender, EventArgs e) { using(Mp3FileReader reader = new Mp3FileReader(@"path\to\MP3")) { using(WaveFileWriter writer = new WaveFileWriter(@"C:\test.wav", new WaveFormat())) { int counter = 0; while(reader.Read(test, counter, test.Length + counter) != 0) { writer.WriteData(test, counter, test.Length + counter); counter += 512; } } } } reader.Read() goes into the Mp3FileReader class, and the method looks like this: public override int Read(byte[] sampleBuffer, int offset, int numBytes) { if (numBytes % waveFormat.BlockAlign != 0) //throw new ApplicationException("Must read complete blocks"); numBytes -= (numBytes % waveFormat.BlockAlign); return mp3Stream.Read(sampleBuffer, offset, numBytes); } mp3Stream is an object of the Stream class. The problem is: I'm getting an ArgumentException. MSDN says that this is because the sum of offset and numBytes is greater than the length of sampleBuffer. Documentation: http://msdn.microsoft.com/en-us/library/system.io.stream.read.aspx This happens because I increase the counter every time, but the size of the byte array test remains the same. What I've been wondering is: do I need to increase the size of the array dynamically, or do I need to find out the needed size at the beginning and set it right away? And also, instead of 512, the method in Mp3FileReader returns 365 the first time. Which is the size of a whole block. But I'm writing the full 512. I'm basically just using the read to check if I'm not at the end of the file yet. Do I need to catch the return value and do something with that, or am I good here?

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  • How to assemble a WAV file?

    - by David
    I'm doing an educational project in which 1) I record voice commands on a separate device and after appropriate processing etc... 2) I send 16-bit samples encapsulated in UDP packets over Ethernet to the PC. After receiving the packets and "extracting" data (samples) from them, I need to assemble the samples to a WAV file. Any example code? Any suggestions?

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  • How to assemble a WAV file, C# preferred?!

    - by David
    Hi guys! I'm doing an educational project in which 1) I record voice commands on a separate device and after appropriate processing etc... 2) I send 16-bit samples encapsulated in UDP packets over Ethernet to the PC. After receiving the packets and "extracting" data (samples) from them, I need to assemble the samples to a WAV file. Any example code? Any suggestions? Thank you in advance!

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  • fourier transform to transpose key of a wav file

    - by tbischel
    I want to write an app to transpose the key a wav file plays in (for fun, I know there are apps that already do this)... my main understanding of how this might be accomplished is to 1) chop the audio file into very small blocks (say 1/10 a second) 2) run an FFT on each block 3) phase shift the frequency space up or down depending on what key I want 4) use an inverse FFT to return each block to the time domain 5) glue all the blocks together But now I'm wondering if the transformed blocks would no longer be continuous when I try to glue them back together. Are there ideas how I should do this to guarantee continuity, or am I just worrying about nothing?

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  • Saving generated .wav file to server with PHP

    - by bionicOnion
    I am developing a website where users can compose their own music, and the site will generate a .wav file for their creation. This is working correctly (inasmuch as I can play it on the page). However, I would like to save this file to the server to be listened to/downloaded at a later time, and the saved version of the file can no longer be opened and played by the HTML audio tag. What, if anything, must I put into the file besides the file besides the raw data? Instead of setting the src attribute of the audio tag to the location of the file, will I actually need to open it and generate a URI?

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  • How to read/write high-resolution (24-bit, 8 channel) .wav files in Java?

    - by dB'
    I'm trying to write a Java application that manipulates high resolution .wav files. I'm having trouble importing the audio data, i.e. converting the .wav file into an array of doubles. When I use a standard approach an exception is thrown. AudioFileFormat as = AudioSystem.getAudioFileFormat(new File("orig.wav")); --> javax.sound.sampled.UnsupportedAudioFileException: file is not a supported file type Here's the file format info according to soxi: dB$ soxi orig.wav soxi WARN wav: wave header missing FmtExt chunk Input File : 'orig.wav' Channels : 8 Sample Rate : 96000 Precision : 24-bit Duration : 00:00:03.16 = 303526 samples ~ 237.13 CDDA sectors File Size : 9.71M Bit Rate : 24.6M Sample Encoding: 32-bit Floating Point PCM Can anyone suggest the simplest method for getting this audio into Java? I've tried using a few techniques. As stated above, I've experimented with the Java AudioSystem (on both Mac and Windows). I've also tried using Andrew Greensted's WavFile class, but this also fails (WavFileException: Compression Code 3 not supported). One workaround is to convert the audio to 16 bits using sox (with the -b 16 flag), but this is suboptimal since it increases the noise floor. Incidentally, I've noticed that the file CAN be read by libsndfile. Is my best bet to write a jni wrapper around libsndfile, or can you suggest something quicker? Note that I don't need to play the audio, I just need to analyze it, manipulate it, and then write it out to a new .wav file. * UPDATE * I solved this problem by modifying Andrew Greensted's WavFile class. His original version only read files encoded as integer values ("format code 1"); my files were encoded as floats ("format code 3"), and that's what was causing the problem. I'll post the modified version of Greensted's code when I get a chance. In the meantime, if anyone wants it, send me a message.

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