The primary reason for this change is to allow time graphs that do not
go through the origin and hence smoothing starting from the origin is
not desired. This change will allow passing time data into the smoother
while paused and then abruptly use that data without smoothing using the
'quick fixup' flag when resuming.
Primary use case is allowing recording time graphs where the data
recorded originates from a time before the stream was created. The
resulting graft will be shifted and should not be smoothened to go
through the origin.
Hi All,
This patch fixes the solaris audio device source and sink, and fixes some
portability issues that break the build on solaris. Questions and comments
welcomed.
I've tested this patch only with OpenSolaris Express snv 103. Eventually I
hope to be able to test a few older releases and older hardware (though it
is hard to say whether there is much interest in those).
This is my first brush with pulseaudio and so I read the wiki docs and
some of the source code but I'm still unsure of a few things. In
particular I'm wondering about rewind processing, corking and what (if
anything) the module needs for those. I'm also unclear on the implications
of thread_info.buffer_size, .fragment_size and .max_request, and whether
my code is correct or not.
This patch disables link map/library versioning unless ld is GNU ld.
Another approach for solaris would be to use that linker's -M option, but
I couldn't make that work (due to undefined mainloop, browse and simple
symbols when linking pacat. I can post the errors if anyone is intested.)
Thanks,
Finn Thain
In theory the callback called after reading headers could free our whole object, so we should not
take it upon ourselves to free the headers after the call to the callback.
Erich Boleyn <erich@uruk.org> wrote:
> Using RTP for multi-room music streaming, updated to Pulse 0.9.14 from
> 0.9.9, RTP reception new crashes with a segfault on all machines at
> the first "Updating sample rate" log message.
>
> Source of the segfault appears to be null pointer for
> "impl_update_rates" function in resampler routine, perhaps
> uninitialized resamplers in general?
A fresh look after work made the resampler initialization code pop out.
The problem is in the sink connection being made from
"module-rtp-recv.c", the "PA_SINK_INPUT_VARIABLE_RATE" flag should be
passed into "pa_sink_input_new", but is not there. Made the change and
tested it, fixes the problem. Checked and head-of- tree off of the
pulseaudio.org source browsing link does not have this fix either.
One-liner patch attached.
This allows us to reconnect upon disconnection but this has thus far proved unreliable.
We no longer close the socket. We leave this to the module thread to do the closing.
We can also flush the remote buffer now.
Refs #69
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/coling@2503 fefdeb5f-60dc-0310-8127-8f9354f1896f
* Store the mainloop, hostname and port internally on construction
* This should allow use to easily reconnect if disconnected although this has thus far proved unreliable.
The changes look like more than they are due to moving a function around.
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/coling@2502 fefdeb5f-60dc-0310-8127-8f9354f1896f
* Change the encode_sample routine to simply return normal memchunks allocated from the mempool.
* unref the memchunks returned from encode_sample when we are done with them.
* Create an encoded 'silence' sample and play this at all times to prevent hangup and to 'hog' the airtunes device
This now works and can be used as a regular sink albeit with a constant latency of about 8 seconds :s
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/coling@2485 fefdeb5f-60dc-0310-8127-8f9354f1896f
Only clear IO related stuff if this free() was triggered deliberatly (i.e. not by server side disconnect)
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/coling@2411 fefdeb5f-60dc-0310-8127-8f9354f1896f
Change port to be 16 bits
Do not free stuff on closure as this happens further up the stack.
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/coling@2410 fefdeb5f-60dc-0310-8127-8f9354f1896f
Store the core* rather than just the mainloop as we can reuse the mempool without passing it in as an argument.
const'ify and deconst'ify some vars
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/coling@2404 fefdeb5f-60dc-0310-8127-8f9354f1896f
This does not seem to fix the pool full messages so I'll have to try and suss that out.
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/coling@2400 fefdeb5f-60dc-0310-8127-8f9354f1896f
Convert the iochannel to an fd and do not call a pa_iochannel_cb_t callback but rather trigger the callback on connection and pass the fd.
Change pa_raop_client_send_sample to pa_raop_client_encode_sample and work with memchunks.
Fix a subtle size bug in the bit writer that techincally isn't triggered in normal operation.
Clean up the _free function to actually free stuff.
Do the actual ALAC encoding.
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/coling@2394 fefdeb5f-60dc-0310-8127-8f9354f1896f
Fix the length type for send_sample (restrict to 16bit value)
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/coling@2374 fefdeb5f-60dc-0310-8127-8f9354f1896f
This makes things fully asyncronous.
Some of the continuation headerlist stuff could be moved to headerlist for neatness, but this is OK for now.
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/coling@2373 fefdeb5f-60dc-0310-8127-8f9354f1896f
Properly handle the sequence of events that establish a connection.
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/coling@2369 fefdeb5f-60dc-0310-8127-8f9354f1896f
It still requires a socket client and callback system to be added before it will be functional.
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/coling@2366 fefdeb5f-60dc-0310-8127-8f9354f1896f
