search for: wbitspersampl

...rt 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_oggB->wfx.nAvgBytesPerSec = (unsigned long) (p_oggB->wfx.nSamplesPerSec * p_oggB->wfx.nBlockAlign); p_oggB->wfx.cbSize = 0; Could someone send me source code,...

...a member in another structure along with the additional information. 2. PCMWAVEFORMAT structure The PCMWAVEFORMAT structure describes the data format for PCM waveform-audio data. This structure has been superseded by the WAVEFORMATEX structure. typedef struct { WAVEFORMAT wf; WORD wBitsPerSample; } PCMWAVEFORMAT; Members wf A WAVEFORMAT structure containing general information about the format of the data. wBitsPerSample Number of bits per sample. 3. WAVEFORMATEX structure The WAVEFORMATEX structure defines the format of waveform-audio data. Only format information common to all...

Hi, I've changed the m_waveFormatEx.wBitsPerSample for both record buffer and playSound buffer, changed my array from char to short and I'm still getting the same constant background noise. I notice that when I talk into the mike, I will get a slightly louder noise, can't hear any clear speech... Any idea what's wrong? Regards, Carine...

...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 :- When the buffer(size = 2000 bytes) gets filled with sound data a function wit...

...eliang@yahoo.com.sg; speex-dev@xiph.org Sent: Thursday, 2 November 2006 2:04:39 PM Subject: RE: [Speex-dev] Integrating speex with VideoNet application: Constantbackground noise DIV { MARGIN:0px;} At first glance it looks like you are opening the output device in 8-bit mode (m_waveFormatEx.wBitsPerSample =8;). You will need to use 16-bit mode if you expect to write an array of signed short integers to the audio buffer. 8-bit PCM is an unsigned mode, no longer used much these days. Also, a good debugging trick is to include something like this in your output sample processing loop:...

...C 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); speex_encoder_ctl(enc_state, SPEEX_SET_QUALITY, &quality);...

...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 pcm stream in realtime...

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

...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: { WAVEHDR *Hdr = (WAVEHDR *...

...fer > > Play PCM char buffer using WaveOut > > > > The end-result is very distorted sound. > > > > > > I use the following PCM format: > > > > waveFormat.nChannels = 1; > > waveFormat.nSamplesPerSec = 8000; > > waveFormat.wBitsPerSample = 8; > > > > Once sound is aquired by the mic, I send it to Speex in a Char buffer. > It looks something like this: > > > > Encode(char* inBuffer, DWORD bufferLength) > > ... > > float* input = new float[frameSize]; > > ZeroMemory(input, fram...

...trivial matter to you and other experts in the field, but believe me, there are a lot of programmers, whose posts I saw on the web, who tried to use WaveIn/WaveOut and Speex and failed. As I understand it, WaveIn just buffers audio data according to the bit rate specified. So, if we use waveFormat.wBitsPerSample = 8, then each Char of the audio buffer describes an audio sample using 8 bits. On the output side, WaveOut expects each Char to describe an 8-bit audio sample as well. If we deal with 16-bit samples, then two Chars would describe a sample, and so on. My question is on how to deal with Speex outp...

...ffer from the microphone Encode Char buffer using Speex Decode Speex Char buffer into PCM char buffer Play PCM char buffer using WaveOut The end-result is very distorted sound. I use the following PCM format: waveFormat.nChannels = 1; waveFormat.nSamplesPerSec = 8000; waveFormat.wBitsPerSample = 8; Once sound is aquired by the mic, I send it to Speex in a Char buffer. It looks something like this: Encode(char* inBuffer, DWORD bufferLength) ... float* input = new float[frameSize]; ZeroMemory(input, frameSize); ... for( ) { input[..] = inBuffer[..]; } ... speex_encode...

...trivial matter to you and other experts in the field, but believe me, there are a lot of programmers, whose posts I saw on the web, who tried to use WaveIn/WaveOut and Speex and failed. As I understand it, WaveIn just buffers audio data according to the bit rate specified. So, if we use waveFormat.wBitsPerSample = 8, then each Char of the audio buffer describes an audio sample using 8 bits. On the output side, WaveOut expects each Char to describe an 8-bit audio sample as well. If we deal with 16-bit samples, then two Chars would describe a sample, and so on. My question is on how to deal with Speex outp...

...Message----- From: Carine Liang [mailto:carineliang@yahoo.com.sg] Sent: Sunday, November 05, 2006 4:51 PM To: John Miles; speex-dev@xiph.org Subject: Re: [Speex-dev] Integrating speex with VideoNet application: Constantbackground noise Hi, I've changed the m_waveFormatEx.wBitsPerSample for both record buffer and playSound buffer, changed my array from char to short and I'm still getting the same constant background noise. I notice that when I talk into the mike, I will get a slightly louder noise, can't hear any clear speech... Any idea what's wrong? Reg...

...o.com.sg; speex-dev@xiph.org Sent: Thursday, 2 November 2006 2:04:39 PM Subject: RE: [Speex-dev] Integrating speex with VideoNet application: Constantbackground noise DIV { MARGIN:0px;} At first glance it looks like you are opening the output device in 8-bit mode (m_waveFormatEx.wBitsPerSample =8;). You will need to use 16-bit mode if you expect to write an array of signed short integers to the audio buffer.. 8-bit PCM is an unsigned mode, no longer used much these days. Also, a good debugging trick is to include something like this in your output sample processing...

...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

...s 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; > > > You want 16 bits per sample, and 16000 bytes per second... -SteveK

...C 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); speex_encoder_ctl(enc_state, SPEEX_SET_QUALITY, &quality);...

...sage----- From: Carine Liang [mailto:carineliang@yahoo.com.sg] Sent: Sunday, November 05, 2006 4:51 PM To: John Miles; speex-dev@xiph.org Subject: Re: [Speex-dev] Integrating speex with VideoNet application: Constantbackground noise Hi, I've changed the m_waveFormatEx.wBitsPerSample for both record buffer and playSound buffer, changed my array from char to short and I'm still getting the same constant background noise. I notice that when I talk into the mike, I will get a slightly louder noise, can't hear any clear speech... Any idea what's wrong? Regards,...

...fer > > Play PCM char buffer using WaveOut > > > > The end-result is very distorted sound. > > > > > > I use the following PCM format: > > > > waveFormat.nChannels = 1; > > waveFormat.nSamplesPerSec = 8000; > > waveFormat.wBitsPerSample = 8; > > > > Once sound is aquired by the mic, I send it to Speex in a Char buffer. > It looks something like this: > > > > Encode(char* inBuffer, DWORD bufferLength) > > ... > > float* input = new float[frameSize]; > > ZeroMemory(input, fram...