<div class="gmail_quote">On Wed, Aug 17, 2011 at 1:51 PM, Arun Raghavan <<a href="mailto:arun.raghavan@collabora.co.uk">arun.raghavan@collabora.co.uk</a>> wrote: <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"> <div class="im">On Wed, 2011-08-17 at 13:34 +0530, kuldeep ghan wrote: > Hi, > > I am working on a setup which has Linux Kernel running and I have > pulseaudio as Sound server. ASoc Alsa driver snd_pcm_mmap function > returns me > Virtual Memory mapped to physical memory which is strictly 32 Bit > Aligned. </div>Out of curiosity, what platform is this? </blockquote><div>I am using OMAP4 platform. </div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"> <div class="im"> > When I am trying to Mix two streams with gst-launch, pa_mix function > tries to write 16Bit mixed samples to this physical memory resulting > in Bus error. > > As a work around I am trying following code which is trying to write > two Mixed 16Bit samples write making it a 32Bit write. > src/pulsecore/sample-util.c > > // *((int16_t*) data) = (int16_t) sum; > if(j==0) { > temp[0]=(int16_t) sum; > j=1; > } else { > temp[1]=(int16_t) sum; > j=0; > sum = ((temp[1] << 16) | (0x0000FFFF & > temp[0])); > > data = (uint8_t*) data - sizeof(int16_t); > *((int32_t*) data) = (int32_t) sum; > data = (uint8_t*) data + sizeof(int16_t); > } > > With this I am able to mix a gst-launch stream and already loaded > sample from pulseaudio played with play-sample from pacmd. > But not able to mix two streams with gst-launch. </div>What is the error you see when playing 2 streams? Do two instances of paplay work fine together? </blockquote><div>The error I see is "Unhandled fault: external abort on non-linefetch (0x1818) at 0x40013000" </div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"> BTW, Another potential workaround (if your hardware supports it), is to force the default sample spec to use S32NE samples (you can edit the default value in /etc/pulse/daemon.conf). The downside is obviously that you're transferring 2x the amount of data that you'd have done with S16LE samples and the format conversion cost for just about every client. </blockquote><div> I have tried using S32NE as default format but it Causes under-runs/over-runs. As I am new to ALSA subsystem and PulseAudio I am yet to figure out how to get those under-runs/over-runs solved? </div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"> Unless there's more hardware out there with this sort of limitation (I don't believe there is), I wouldn't be too keen on adding this sort of workaround upstream. Regards, Arun _______________________________________________ pulseaudio-discuss mailing list <a href="mailto:pulseaudio-discuss@lists.freedesktop.org">pulseaudio-discuss@lists.freedesktop.org</a> <a href="http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss" target="_blank">http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss</a> </blockquote></div> Thanks, Kuldeep