search for: nsamplespersec

...fest 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_oggB->wfx.nAvgBytesPerSec = (unsigned long) (p_oggB->wfx.nSamplesPerSec * p_oggB->wfx.nBlockA...

...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 This structure has been superseded by the WAVEFORMATEX structure.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 data uses one channel and stereo data uses two c...

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

...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 with the body shown below is called. LPWAVEHDR lpHdr = (LPWAVEHDR) lParam; if(lpH...

...ed 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 (I didn't found any)? Than...

...s 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 *) msg.lParam;...

...d when I try to Get PCM Char buffer 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[..] = i...

...t; 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[f...

...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); In my recording callback function LRESULT RecordSound::OnS...

...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); In my recording callback function LRESULT Re...

...lized, I turn on VBR, the speak become very noisy immediately. 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 ph...

...now 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

...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); In my recording callback function LRESULT RecordSound::OnSo...

...t; 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[f...

...nce, DWORD_PTR dwParam1, DWORD_PTR dwParam2 ) { if ( uMsg == WOM_DONE ) { EnterCriticalSection ( &cs ); @@ -125,7 +125,7 @@ Set_WIN_Params ( FILE_T dummyFile , outFormat.nBlockAlign = (outFormat.wBitsPerSample + 7) / 8 * outFormat.nChannels; outFormat.nAvgBytesPerSec = outFormat.nSamplesPerSec * outFormat.nBlockAlign; - switch ( waveOutOpen ( &dev, deviceID, &outFormat, (DWORD)wave_callback, 0, CALLBACK_FUNCTION ) ) + switch ( waveOutOpen ( &dev, deviceID, &outFormat, (DWORD_PTR)wave_callback, 0, CALLBACK_FUNCTION ) ) { case MMSYSERR_ALLOCATED: return Box ( &quot...

...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) >>...

...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) >>...

...k 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); In my recording callback function...

...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) >>...

...k 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); In my recording callback function...