search for: wave_format_pcm

...structure. This structure has been superseded by the WAVEFORMATEX structure. typedef struct { WORD wFormatTag; WORD nChannels; DWORD nSamplesPerSec; DWORD nAvgBytesPerSec; WORD nBlockAlign; } WAVEFORMAT; Members wFormatTag Format type. The following type is defined: WAVE_FORMAT_PCM Waveform-audio data is PCM. nChannels Number of channels in the waveform-audio data. Mono data uses one channel and stereo data uses two channels. nSamplesPerSec Sample rate, in samples per second. nAvgBytesPerSec Required average data transfer rate, in bytes per second. For example...

...+-------------+-+ +-| buffer 3 | +-------------+ First i fill all buffest with ov_read. Then I start to play firts buffer. When i stops I'll start to play second buffer and fill firts buffer with ov_read. Buffer setting are: p_oggB->wfx.wFormatTag = WAVE_FORMAT_PCM; p_oggB->wfx.nChannels = oggstruct.vi->channels; p_oggB->wfx.nSamplesPerSec = (unsigned long) oggstruct.vi->rate; p_oggB->wfx.wBitsPerSample = 16; p_oggB->wfx.nBlockAlign = (oggstruct.vi->channels * p_oggB->wfx.wBitsPerSample) / 8; p_...

...my problem before but didn't get a solution. I am doing an VOIP project in which i am recording sound and streaming it to the peer. I wanted to encode and decode wav files that brought me to this site. I am recording sound in the following format:- m_WaveFormatEx.wFormatTag = WAVE_FORMAT_PCM; m_WaveFormatEx.nChannels = 1; m_WaveFormatEx.wBitsPerSample = 8; m_WaveFormatEx.cbSize = 0; m_WaveFormatEx.nSamplesPerSec = 8000; m_WaveFormatEx.nBlockAlign = 1; m_WaveFormatEx.nAvgBytesPerSec = 8000; The recording is as follows :- Whe...

...The flac encoder does give a warning during the encoding process: WARNING: legacy WAVE file has format type 1 but bits-per-sample=24 Looking at the specs for WAV files, it appears that for 24-bit files should have their format code set to WAVE_FORMAT_EXTENSIBLE, but these files are still set to WAVE_FORMAT_PCM. Could this be the source of the problem? If not, do you have any other ideas? I can provide example files if wanted. Thanks, Justin

Hi Guys, I've just faced a wrong warning trying to encode a 24 bits WAV file if(wFormatTag == 1) { if(bps != 8 && bps != 16) { if(bps == 24 || bps == 32) { /* let these slide with a warning since they're unambiguous */ flac__utils_printf(stderr, 1, "%s: WARNING: legacy WAVE file has

...one. #define FRAME_SIZE 160 The application is coded in MFC C++. The record buffer is set to be the same as the FRAME_SIZE such that my callback function is called everytime 160 bytes of data is recorded. Here's what I did in the initialisation (called only once). m_waveFormatEx.wFormatTag =WAVE_FORMAT_PCM; m_waveFormatEx.nChannels =1; m_waveFormatEx.wBitsPerSample =8; m_waveFormatEx.cbSize =0; m_waveFormatEx.nSamplesPerSec = 8000; m_waveFormatEx.nBlockAlign =1; m_waveFormatEx.nAvgBytesPerSec = 8000; int quality =8; speex_bits_init(&bits); enc_state = speex_encoder_init(&speex_nb_mode); spe...

...m the DirectX SDK documentation: ... The particular format of a sound can be described by a WAVEFORMATEX or WAVEFORMATEXTENSIBLE structure. The wFormatTag member of this structure contains a unique identifier assigned by Microsoft Corporation. The only tags valid with DirectSound are the following: WAVE_FORMAT_PCM. This tag indicates PCM, an uncompressed format. WAVE_FORMAT_EXTENSIBLE. Multichannel audio format; valid only with WDM drivers. ... >another question. Does FLAC need the metadata at the beginning of a file. Assume you open a FLAC file in >VirtualDub, for example. When you cut out the first...

...FRAME_SIZE 160 The application is coded in MFC C++. The record buffer is set to be the same as the FRAME_SIZE such that my callback function is called everytime 160 bytes of data is recorded. Here's what I did in the initialisation (called only once). m_waveFormatEx.wFormatTag =WAVE_FORMAT_PCM; m_waveFormatEx.nChannels =1; m_waveFormatEx.wBitsPerSample =8; m_waveFormatEx.cbSize =0; m_waveFormatEx.nSamplesPerSec = 8000; m_waveFormatEx.nBlockAlign =1; m_waveFormatEx.nAvgBytesPerSec = 8000; int quality =8; speex_bits_init(&bits); enc_state = speex_encoder_init(&speex...

...e has any idea? Also, DTX is still not working. I have gone through the manual but not much info on this area. The following is my setting from sound card using mm_system sdk. WAVEFORMATEX *wfx = (WAVEFORMATEX *)input; wfx->nSamplesPerSec = 8000; wfx->nChannels = 1; wfx->wFormatTag = WAVE_FORMAT_PCM; wfx->wBitsPerSample = 16; wfx->nBlockAlign = 2; wfx->nAvgBytesPerSec = 16000; Thanks in advance, YueWeng __________________________________ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo

...o the peer. I wanted to >encode and decode > wav files that brought me to this site. > > I am recording sound in the following format:- > > I don't know much about windows "Wave" stuff, but this doesn't look right: > m_WaveFormatEx.wFormatTag = WAVE_FORMAT_PCM; > m_WaveFormatEx.nChannels = 1; > m_WaveFormatEx.wBitsPerSample = 8; > m_WaveFormatEx.cbSize = 0; > m_WaveFormatEx.nSamplesPerSec = 8000; > m_WaveFormatEx.nBlockAlign = 1; > m_WaveFormatEx.nAvgBytesPerSec = 8000; > > >...

...one. #define FRAME_SIZE 160 The application is coded in MFC C++. The record buffer is set to be the same as the FRAME_SIZE such that my callback function is called everytime 160 bytes of data is recorded. Here's what I did in the initialisation (called only once). m_waveFormatEx.wFormatTag =WAVE_FORMAT_PCM; m_waveFormatEx.nChannels =1; m_waveFormatEx.wBitsPerSample =8; m_waveFormatEx.cbSize =0; m_waveFormatEx.nSamplesPerSec = 8000; m_waveFormatEx.nBlockAlign =1; m_waveFormatEx.nAvgBytesPerSec = 8000; int quality =8; speex_bits_init(&bits); enc_state = speex_encoder_init(&speex_nb_mode); spee...

...c_state); return writing_pos; } The functions are called this way: //Encoding save_length = audio_encode(buffer,Hdr,Hdr->dwBytesRecorded,8); //... //Decode again: save_length = audio_decode(Hdr,buffer,save_length); Here are the waveform options: waveform.nChannels = 1; waveform.wFormatTag = WAVE_FORMAT_PCM; waveform.wBitsPerSample = 16; waveform.nSamplesPerSec = 8000; waveform.nBlockAlign = (waveform.nChannels*waveform.wBitsPerSample)/8; waveform.nAvgBytesPerSec = waveform.nSamplesPerSec*waveform.nBlockAlign; waveform.cbSize = 0; What am I doing wrong? Are there any examples for encoding a waveform...

...The flac encoder does give a warning during the encoding process: WARNING: legacy WAVE file has format type 1 but bits-per-sample=24 Looking at the specs for WAV files, it appears that for 24-bit files should have their format code set to WAVE_FORMAT_EXTENSIBLE, but these files are still set to WAVE_FORMAT_PCM. Could this be the source of the problem? If not, do you have any other ideas? I can provide example files if wanted. Thanks, Justin

...is an example of my code. I get message from input device then encode with opus then decode it back to output wave header and play. Doing it i hear only noise in my headphones. So how can i use Opus in this way? Thanks in advance WAVEFORMATEX waveform; waveform.cbSize = 0; waveform.wFormatTag = WAVE_FORMAT_PCM; waveform.nChannels = 2; waveform.wBitsPerSample = 8; waveform.nSamplesPerSec = 48000; waveform.nBlockAlign = waveform.nChannels * (waveform.wBitsPerSample >> 3); waveform.nAvgBytesPerSec = waveform.nSamplesPerSec * waveform.nBlockAlign; #define BUFSIZE 512 ... case MM_WIM_DATA:...

Hi all, I recently managed to install "multimedia German dictionary", which I get with the local newspaper , under wine and it works OK. The problem it is when I try to use the most important feature for me - the pronunciation. When I try to listen to any word , no sound is played and wine outputs the following to the console: wodOpen unimplemented format: WAVE_FORMAT_ADPCM I rewieved

Hi, I am using Speex to compress speech samples used for dictation. I am thrilled by the compression ratios Speex achieves with almost no loss in quality. However, I have come across a few files which Speex (I use the Windows variant) refuses to compress. I have put a sample file at http://leggewie.biz/test.wav. The error message I receive is "Unsupported WAVE file fmt chunk, not

...160 The application is coded in MFC C++. The record buffer is set to be the same as the FRAME_SIZE such that my callback function is called everytime 160 bytes of data is recorded. Here's what I did in the initialisation (called only once). m_waveFormatEx.wFormatTag =WAVE_FORMAT_PCM; m_waveFormatEx.nChannels =1; m_waveFormatEx.wBitsPerSample =8; m_waveFormatEx.cbSize =0; m_waveFormatEx.nSamplesPerSec = 8000; m_waveFormatEx.nBlockAlign =1; m_waveFormatEx.nAvgBytesPerSec = 8000; int quality =8; speex_bits_init(&bits); enc_state = speex_encoder_init(...

...er (3): ws2_32: Don't post completion packet if receiving fails with error immediately. iphlpapi: Implement if_nametoindex. msvcp140: Implement _To_wide. Zebediah Figura (4): secur32/tests: Zero-initialize contexts. msacm32: Don't write cbSize if the format is WAVE_FORMAT_PCM. winemp3.acm: Fix error in acmStreamSize(). msacm32: Rewrite PCM conversion functions. Zhenbo Li (1): ole32: Add stub for OleGetIconOfFile. -- Michael Stefaniuc mstefani at winehq.org

...160 The application is coded in MFC C++. The record buffer is set to be the same as the FRAME_SIZE such that my callback function is called everytime 160 bytes of data is recorded. Here's what I did in the initialisation (called only once). m_waveFormatEx.wFormatTag =WAVE_FORMAT_PCM; m_waveFormatEx.nChannels =1; m_waveFormatEx.wBitsPerSample =8; m_waveFormatEx.cbSize =0; m_waveFormatEx.nSamplesPerSec = 8000; m_waveFormatEx.nBlockAlign =1; m_waveFormatEx.nAvgBytesPerSec = 8000; int quality =8; speex_bits_init(&bits); enc_state = speex_encoder_init(...

Hi all, Currently I'm looking for users of the --keep-foreign-metadata feature of FLAC. There has been some improvement of this feature in FLAC 1.4.0. Since 2007 there has been a warning in FLAC that --keep-foreign-metadata is a new feature. I think removal of this warning is long overdue, but there are still some issues surrounding it. So, if there are users of this feature on the mailing