Greg Pomerantz wrote:
> +++ drivers/sound/i810_audio.c Mon Dec 17 23:15:19 2001
> @@ -609,6 +609,9 @@
>
> new_rate = ac97_set_dac_rate(codec, rate);
>
> + if ((dmabuf->fmt & I810_FMT_STEREO) == 0)
> + new_rate *= 2;
> +
> if(new_rate != rate) {
> dmabuf->rate = (new_rate * 48000)/clocking;
> }
> @@ -1687,6 +1690,12 @@
> if (dmabuf->enable & ADC_RUNNING) {
> stop_adc(state);
> }
> +
> + if (*(int *)arg == 0)
> + dmabuf->fmt &= ~I810_FMT_STEREO;
> + else
> + dmabuf->fmt |= I810_FMT_STEREO;
> +
> return put_user(1, (int *)arg);
>
> case SNDCTL_DSP_GETBLKSIZE:
OK, first off, Alan & Co. will shoot this one down because handling
stereo/mono conversion belongs in user space (the agreed upon way of
handling things evidently). Secondly, aside from that point, this is wrong
anyway. You are dividing the play rate in half to compensate for the fact
that the sound is being sent to two channels instead of one. So, now each
channel is playing at half speed and slightly out of phase and with much
reduced high frequency clarity. Instead, what you should do, is byte double
the stream (CPU consuming). You would need to take any input buffer and
copy every 16bit sample. A very simple (and non-optimized) stereo converter
might look like this:
short *input=&input_buffer, *output=&output_buffer;
int i,j;
for(i=0, j=0;i < input_size;i++, j+=2)
output[j] = output[j+1] = input[i];
--Doug Ledford <dledford@redhat.com> http://people.redhat.com/dledford Please check my web site for aic7xxx updates/answers before e-mailing me about problems
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This archive was generated by hypermail 2b29 : Sun Dec 23 2001 - 21:00:16 EST