Repair Waveoutopen Failed With Error Code 32 Tutorial

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Waveoutopen Failed With Error Code 32


Possible error values include the following. The documentation states: If wFormatTag is WAVE_FORMAT_PCM, nBlockAlign must equal (nChannels × wBitsPerSample) / 8 You are missing the / 8 portion: fmt.nBlockAlign = (fmt.nChannels * fmt.wBitsPerSample) / 8; share|improve this answer answered Please refer to our Privacy Policy or Contact Us for more details You seem to have CSS turned off. The streaming mechanism is maintained by creating and queuing one buffer while the other records.

I'll try it again.EDIT: vbam just freezes after loading the rom.EDIT: turns out, you have to install alsa-plugins-jack.EDIT: I even got flash to work, but vbam still freezes until I shut We appreciate your feedback. Last edited by SketchMan3 (July 2, 2012 8:37 pm) 20 July 2, 2012 8:44 pm munchluxe63 Offline BC, Canada SketchMan3 wrote:Have you tried BGB + Wine?Edit: Also, with recording to Audacity, Copy public bool Done { get { return !m_playing; } } protected bool m_playing = false; The Seconds property specifies the length of the audio in seconds. my site

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Can you post the WAVEFORMATEX data in here for my reference? In the case of the last block, the buffer is set to the size required to record for the remainder of the time needed. The GDI+ of audio, if you will.

Unlike WaveOut, the terms block and buffer are interchangeable when discussing the WaveIn class. You can also use the following flag instead of a device identifier: Value Meaning WAVE_MAPPERThe function selects a waveform-audio output device capable of playing the given format.   pwfx Pointer to Apr 1, 2010 at 2:24pm UTC Disch (13766) In WAVEFORMATEX cbSize is not the size of the structure, but the size of extra info in the file's "fmt" chunk. Otherwise, this is 0. */ + DWORD error; +} priv_t; - priv->handle_data = GlobalAlloc(GMEM_MOVEABLE | GMEM_SHARE, buf_len * num_buffers); - priv->ptr_data[0] = (HPSTR) GlobalLock(priv->handle_data); - priv->handle_wavheader = GlobalAlloc(GMEM_MOVEABLE | GMEM_SHARE, (DWORD)sizeof(WAVEHDR)

For more information, see Remarks. Createfile waveInGetNumDevs() : waveOutGetNumDevs(); + for (dev = -1; dev == WAVE_MAPPER || dev < dev_count; dev++) + { + if (recording) { - waveOutReset(priv->waveout); + priv->error = waveInGetDevCapsA(dev, &incaps, sizeof(incaps)); + IN NO EVENT SHALL THE COPYRIGHT OWNER OR ** CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, ** EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, ** PROCUREMENT OF Personal Open source Business Explore Sign up Sign in Pricing Blog Support Search GitHub This repository Watch 28 Star 230 Fork 60 erikd/libsndfile Code Issues 17 Pull requests 3 Projects

For example, for PCM data, an extra UINT is added to specify the number of bits per sample. Terms Privacy Opt Out Choices Advertise Get latest updates about Open Source Projects, Conferences and News. But on Windows Vista and newer (have tried Vista, 7, and 8.1), I get open error: The specified format is not supported or cannot be translated. Ticks and tempo are slightly variable when recording using LSDJ.

  • Mar 18, 2010 at 2:52pm UTC Disch (13766) .as in no sound.
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  • dwCallbackInstance User-instance data passed to the callback mechanism.
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  • The Waveform Audio Interface The Waveform Audio Interface provides several functions for controlling the hardware audio interface and provides the lowest level of control available to Windows Mobile developers.


I would recommend you just take the necessary data from the "fmt " chunk and discard the rest. SourceForge Browse Enterprise Blog Deals Help Create Log In or Join Solution Centers Go Parallel Resources Newsletters Cloud Storage Providers Business VoIP Providers Internet Speed Test Call Center Providers Thanks for Windows Error Codes can't determine length\n") ; } ; #endif snd_pcm_prepare (alsa_dev) ; break ; case -EBADFD : fprintf (stderr, "alsa_write_float: Bad PCM state.n") ; return 0 ; break ; case -ESTRPIPE : fprintf Copy public Wave.MMSYSERR Save(string fileName) { if (!m_inited) return Wave.MMSYSERR.ERROR; if (m_recording) Stop(); FileStream strm = null; BinaryWriter wrtr = null; try { if (File.Exists(fileName)) { FileInfo fi = new FileInfo(fileName);

Copy public Wave.MMSYSERR Play(uint curDevice, String fileName, IntPtr hwnd, int bufferSize, ushort volLeft, ushort volRight) { if (m_playing) return Wave.MMSYSERR.NOERROR; if (!File.Exists(fileName)) return Wave.MMSYSERR.ERROR; FileInfo fi = new FileInfo(fileName); if ((fi.Attributes Any idea how to fix that? VBAM, however, is awesome. It might be something as simple as the cbSize member being incorrect.

Please don't fill out this field. These methods are detailed in the comments of the code accompanying the P/Invoke Library sample. waveOutPause(): Pauses playback. his comment is here Welll there's always more details to know ..

For recording, this is the buffer from which we'll + * be getting the next samples. If the value specified by the uDeviceID parameter is a device identifier, it can vary from zero to one less than the number of devices present. Proof of turings halting problem Offline tool that can show dependencies between different metadata files Why was Vader surprised that Obi-Wan's body disappeared?

The lpData member of the first buffer points at + * data[buf_len*sample_size*0], the second buffer's lpData points + * data[buf_len*sample_size*1], etc.

If the buffer size is set to 0 then it will be made large enough to contain the entire wave file, otherwise, the audio being recorded will be streamed to buffers Syntax C++ Copy MMRESULT waveOutOpen(  LPHWAVEOUT     phwo,  UINT_PTR       uDeviceID,  LPWAVEFORMATEX pwfx,  DWORD_PTR      dwCallback,  DWORD_PTR      dwCallbackInstance,  DWORD          fdwOpen ); Parameters phwo Pointer to a buffer that receives a handle identifying the open It'll start with the RIFF header then you find the 'fmt ' chunk which has the samplerate and stuff then you have the 'data' chunk which has the PCM data. The m_curBlock member is the block that is currently being played - once again, this is not the same as the current buffer.

The Write method, on the other hand, does not write this extra data, as it has no knowledge of the required information. Calling waveOutUnprepareHeader would sort of kill the audio buffer. The tutorials and walkthroughs were very helpful in the development of my project. One thing I can think of is that the 'format' or 'type' (or whatever that member is called) is wrong.

If no buffers are ready for processing, this is the buffer + * that will be the next to become ready. + */ + unsigned current; - buf_len = sox_globals.bufsiz * What does "M.C." in "M.C. PLUS, the "info" chunk might not be immediately after the RIFF header like you appear to be assuming (there might be a comment chunk or something before it). Both the playback and recording samples also utilize a MessageWindow for receiving messages from the audio system when a block has finished.

To reduce memory use, the sample allows the user to specify a buffer size. waveInUnprepareHeader(): Releases a previously prepared WAVEHDR and data block. Unfortunately with my decision to drop SDL I lost my way of handling audio. The streaming mechanism is maintained by loading one buffer while the other plays.

Are any of the functions returning an error? If the buffers are saved or recording is started before they are saved, then the buffers will be freed at that time. I was wondering if any of you guys could tell me a Windows-specific way of handling audio. char[4] = "fmt ", The "fmt " characters specify that this is the section of the file describing the format specifically Int32 = 16, The size of the WAVEFORMATEX data to

Just as a test to see if you can get audio working. Copy public Wave.MMSYSERR Preload(uint curDevice, IntPtr hwnd, int maxRecordLength_ms, int bufferSize) { if (m_recording) return Wave.MMSYSERR.ERROR; if (m_inited) { Stop(); FreeWaveBuffers(); } WAVEINCAPS caps = new WAVEINCAPS(); waveInGetDevCaps(0, caps, caps.Size); if As for dumping the data, no need, i debugged the app and put a few watches on the variables. The constructor for this class takes an instance of WaveIn as its parameter.

Playback of audio is as simple as a single method call, while recording can be done with three. Use the handle to identify the device when calling other waveform-audio output functions. For playback, this is the buffer + * that will receive the next samples. waveOutReset(): Stops playing and empties the queue.