This adds code to specifically support legacy entries.
I kept this code in a separate commit so that it can be (relatively)
easily removed at some point in the future.
This simply exposes the formats that a device supports
via a simple protocol extension that will allow clients
to setup what a connected receiver supports format wise.
This has the advantage of allowing versioned updates in the future,
thus allowing us to be more user friendly going forward (as opposed
to just ignoring entries from old versions).
The primary motivation for this, however, is to allow variable length
storage in each entry which will be needed for upcoming work.
At present this commit will ignore any legacy entries but support
for reading and subsequently converting legacy entries will be added
shortly.
After the rework to the add pa_sink_input_new_data_set_sink() (and
the source equiv) calling with a NULL sink object will hit an assert.
This caused crashes with the esd protocol and there was the potential
(albeit unlikely) for a crash when creating a sink input without any
sinks available (module-always-sink mitigates this risk but it's still
a potential crasher).
This was added to ensure symmetry between playback and recording streams
code, but in reality this makes little sense practically speaking and thus
it is removed.
The previous logic in ade0a6f884
does not work with for input volumes.
This was discussed on the mailing list:
https://tango.0pointer.de/pipermail/pulseaudio-discuss/2011-May/010091.html
This approach can introduce a problem when setting the volumes
for sources. What follows is Tanu Kaskinen's analysis:
[quote]
I'll quote the log:
D: protocol-native.c: Client pavucontrol changes volume of source alsa_input.pci-0000_00_1b.0.analog-stereo.
D: alsa-source.c: Requested volume: 0: 45% 1: 45%
D: alsa-source.c: in dB: 0: -20.71 dB 1: -20.71 dB
D: alsa-source.c: Got hardware volume: 0: 45% 1: 45%
D: alsa-source.c: in dB: 0: -21.00 dB 1: -21.00 dB
D: alsa-source.c: Calculated software volume: 0: 101% 1: 101% (accurate-enough=no)
D: alsa-source.c: in dB: 0: 0.29 dB 1: 0.29 dB
D: source.c: Volume going up to 29273 at 270475970821
D: source.c: Volume change to 29273 at 270475970821 was written 34 usec late
D: alsa-source.c: Written HW volume did not match with the request: 0: 45% 1: 45% (request) != 0: 42% 1: 42%
D: alsa-source.c: in dB: 0: -21.00 dB 1: -21.00 dB (request) != 0: -22.50 dB 1: -22.50 dB
Looking at the last line, the requested volume seems to hit exactly the
right step (-21.00dB), but for some reason Alsa decides to choose
something else. I'm pretty sure that this happens because of rounding
errors. In the first phase we ask Alsa what dB value we should set, and
it returns -21.00 dB. The value is given as a long int, but we convert
that to pa_cvolume. Then when we set the volume, we convert the
pa_cvolume value back to a long integer. At this point I believe it gets
converted to -2101. This is not visible in the debug message for some
reason - the rounding algorithm must be different from what was used
with the pa_cvolume -> long conversion.
[/quote]
The commit after this contains a patch that addresses this issue.
This piggy backs onto the previous changes for protocol 22 and
thus does not bump the version. This and the previous commits should be
seen as mostly atomic. Apologies for any bisecting issues this causes
(although I would expect these to be minimal)
The "rm" basm constraint doesn't work with my version of gcc (4.5.2),
not even in a simple example. Since we usually only have 5 registers
available on i386, force it to be memory on that architecture.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Passing a NULL-terminated array of pa_format_info pointers is a bit
unwieldy for clients. Instead of this, let's pass in an array of
pointers and the number of elements in the array.
We were calculating new latency based on the latency set on the old
sink/source, rather than the actual latency requested by the client.
Over a series of moves, this will lead the latency being ~halved each
time, resulting in an eventual rewind flood from a latency that cannot
be handled.
We were using the block size in bytes instead of samples, which meant
preprocessing was broken. This fix makes a large-ish difference in the
quality of echo-cancellation with speex.
The smoother is paused on initialization and resumed when the sink
state is set to running. Otherwise, early latency estimates are
too low since there is some delay between module initialization and
entering the running state.
After the smoother is initially resumed, it is paused when the sink
state is not running. The previous behavior was to pause only when
the sink enters suspended state, however, this would lead to large
errors in latency estimates after the sink has been idle for some
time.
