I am having a devil of a time getting tthe alsa sb16 driver to stay
synced even in half duplex.
I have a VoIP app that is half duplex and it woeks fine using the oss
free drivers in the kernel tree. I simply close the file discriptor on
/dev/dsp, and re-open it, setting the parameters each, mono/16 bit/8Khz,
every time you change from reading /dev/dsp, ie transmit, and writing
to /dev/dsp, ie receiving.
But, the microphone audio seems to be getting buffered ALL THE TIME and
even closing and re-openning does not clear the buffer and start from
scratch.
Here the code that is run on every Tx.Rx change over AFTER the audiofd
is closed.
int soundInput(void){
int arg, status;
long frag_size = 0x7FFF000B;
audiofd = open(SoundFileIn, O_RDWR);
if(audiofd < 0){
perror(" opening audio input ");
exit(1);
}
if (ioctl(audiofd, SNDCTL_DSP_SETFRAGMENT, &frag_size) == -1) {
perror("SNDCTL_DSP_SETFRAGMENT ioctl failed");
}
arg = SIZE;
status = ioctl(audiofd, SOUND_PCM_WRITE_BITS, &arg);
if(status == -1)
perror("SOUND_PCM_WRITE_BITS ioctl failed");
if(arg != SIZE)
perror("unable to set sample size");
arg = CHANNELS;
status = ioctl(audiofd, SOUND_PCM_WRITE_CHANNELS, &arg);
if(status == -1)
perror("SOUND_PCM_WRITE_CHANNELS ioctl failed");
if(arg != CHANNELS)
perror("unable to set number of channels");
arg = RATE;
status = ioctl(audiofd, SOUND_PCM_WRITE_RATE, &arg);
if(status == -1)
perror("SOUND_PCM_WRITE_RATE ioctl failed");
arg = AFMT_S16_LE;
status = ioctl(audiofd, SOUND_PCM_SETFMT, &arg);
if(status == -1)
perror(" SOUND_PCM_SETFMT ioctl failed");
ioctl(audiofd, SOUND_PCM_SYNC, 0);
}
--
Jeff, wd4nmq
[EMAIL PROTECTED]
http://mywebpages.comcast.net/wd4nmq
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