search for: wformattag

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...

...iesen ------ 1. WAVEFORMAT structure The WAVEFORMAT structure describes the format of waveform-audio data. Only format information common to all waveform-audio data formats is included in this 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 This structure has been superseded by the WAVEFORMATEX structure.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 d...

...tand the warning when I try to break it down. First, *legacy* WAVE file -- what makes this a "legacy" WAVE file? Second, format type 1 What is that exactly? In the source code I see "format type" is referred to as "format tag", since the variable name is "wFormatTag". Is this what's referred to in this reference http://soundfile.sapp.org/doc/WaveFormat/ as AudioFormat There it says 1 for PCM Is there something about PCM format that implies either 8- or 16-bits-per-second? I'm new at this and trying to understand it. Can someone explai...

...| +-------------+-+ +-| 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.wBitsPerSampl...

...dows. I have posted 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 record...

Hi (again), I started to work on an ACM (Audio Compression Manager) for FLAC. I currently only have the canvas. The encoding/decoding is not done yet. So it's completely useless. But since the canvas is the harder thing to start, here it is. (just in case I don't have time to continue) http://mukoli.free.fr/flac-ACM.v0.1.0.zip

Hi, Is there a defined format tag for Opus that should be used with the WAVEFORMATEX structure? See wFormatTag (http://msdn.microsoft.com/en-us/library/windows/desktop/dd390970.aspx). /John

Hi Ingo, >It is just a WAVEFORMATEX with a GUID in it. For the ACM system it is just a format with the id >WAVE_FORMAT_EXTENSIBLE and since ACM drivers are not registered by wFormatTag it should be working. >For testing purposes I've written a Vorbis ACM driver, that uses a WAVEFORMATEXTENSIBLE and I experienced >a problem when opening files with Media Player. It spend quite some time at 100 % CPU doing nothing (from the >users perspective). However the problem might...

...a-nerd.com> To: flac-dev at xiph.org Subject: Re: [flac-dev] warning that legacy WAVE file has format type 1 but bits-per-sample is 24 Date: Sat, 27 May 2017 08:40:24 +1000 Mark H. David wrote: > I'm new at this and trying to understand it. Can someone explain?\ You are correct. With a wFormatTag of 1, the bist per sample should be either 8 or 16 (according to the Microsoft specs). > Also, a related but more practical question: is there *any* way to > create a 24- or 32-bits-per-sample WAVE file without getting this > warning when flac tries to encode it? Yes there is. To avoid...

...the microphone. #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(&sp...

...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_enc...

...ately. Anyone 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

...und 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:- > > 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...

...the microphone. #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(&spe...

...er_destroy(dec_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 e...

...ks ok) Here 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_W...

>> Is there a defined format tag for Opus that should be used with the >> WAVEFORMATEX structure? See wFormatTag >> (http://msdn.microsoft.com/en-us/library/windows/desktop/dd390970.aspx). > > There isn't one. Putting Opus into RIFF is complicated by the fact that > it requires both variable-duration and variable-size frames, though I > suppose something could be done along the lines tha...

...e this in lame. And so you can put as >>many data as you want in the WAV header, ie metadata. So it could be done >> > ! > > I thought this, since ACM has problems with WAVEFORMATEXTENSIBLE. The > annoying thing is, that WAVEFORMATEX requires a registration of a global > wFormatTag value for FLAC with Microsoft. > > I'm just interested, because I'd like to enable FLAC in my Ogg > demultiplexer. (However, I won't write a single line of code until April.) > > Regards, > > Ingo > > > _______________________________________________ &...

Hi, >That's what I wanted to say a month ago. Mr. Blum, I'm really sorry for misunderstanding you. My mistake. I choose an ACM codec because it's the only option that works on every Windows Platform, and since the DirectShow has a generic wrapper that allows it to use ACM codecs, it appeared a good idea. I didn't note that the file format was going to be a problem. But I think

...0; + //unsigned int period_time = 20000; snd_pcm_uframes_t buffer_size; snd_pcm_uframes_t period_size; int flags; @@ -753,9 +753,9 @@ static DWORD wodOpen(WORD wDevID, LPWAVE ALSA_getFormat(wwo->format.Format.wFormatTag)); dir=0; - EXIT_ON_ERROR( snd_pcm_hw_params_set_buffer_time_near(pcm, hw_params, &buffer_time, &dir), MMSYSERR_INVALPARAM, "unable to set buffer time"); + //EXIT_ON_ERROR( snd_pcm_hw_params_set_buffer_time_near(pcm, hw_params, &buffer_time, &dir), MMSYSERR_IN...