Configure CMake to require the installed backend libraries. This should
help to find build system regressions.
On TravisCI with Linux this requires the ALSA, PulseAudio, PortAudio, OSS
and JACK backend.
On TravisCI with Android cross compile Linux this requires the OpenSL
backend.
On TravisCI with MacOSX this requires the CoreAudio backend.
ON AppVeyor with Windows this requires the WinMM, DSound and MMDevAPI
backend.
Install Ubuntu development packages for PulseAudio, PortAudio, ALSA and
JACK to enable the building of most Linux backends on TravisCI.
Intall Ubuntu development packages for Qt5 to enable `alsoft-config`.
This extends the base ALC_SOFT_loopback extension with support for B-Format.
When ALC_FORMAT_CHANNELS_SOFT is set to ALC_BFORMAT3D_SOFT, then additional
attributes must be specified. ALC_AMBISONIC_LAYOUT_SOFT must be set to
ALC_ACN_SOFT or ALC_FUMA_SOFT for the desired channel layout,
ALC_AMBISONIC_SCALING_SOFT must be set to ALC_N3D_SOFT, ALC_SN3D_SOFT, or
ALC_FUMA_SOFT for the desired channel scaling/normalization scheme, and
ALC_AMBISONIC_ORDER_SOFT must be set to an integer value greater than 0 for the
ambisonic order (maximum allowed is implementation-dependent).
Note that the number of channels required for ALC_BFORMAT3D_SOFT is dependent
on the ambisonic order. The number of channels can be calculated by:
num_channels = (order+1) * (order+1); /* or pow(order+1, 2); */
In addition, a new alcIsAmbisonicFormatSupportedSOFT function allows apps to
determine which layout/scaling/order combinations are supported by the loopback
device. For example,
alcIsAmbisonicFormatSupported(device, ALC_ACN_SOFT, ALC_SN3D_SOFT, 2) will
check if 2nd order AmbiX (ACN layout and SN3D scaling) rendering is supported
for ALC_BFORMAT3D_SOFT output.
This has a couple behavioral changes. First and biggest is that querying
AL_BUFFERS_PROCESSED from a source will always return all buffers processed
when in an AL_STOPPED state. Previously all buffers would be set as processed
when first becoming stopped, but newly queued buffers would *not* be indicated
as processed. That old behavior was not compliant with the spec, which
unequivocally states "On a source in the AL_STOPPED state, all buffers are
processed."
Secondly, querying AL_BUFFER on an AL_STREAMING source will now always return
0. Previously it would return the current "active" buffer in the queue, but
there's no basis for that in the spec.
This avoids using seq_cst for loading the source state when either inside the
mixer, or otherwise protected from inconsistencies with async updates. It also
fixes potential race conditions with getting the source offset just as a source
stops.
This change allows pair-wise panning to mostly go through the normal ambisonic
panning methods, with one special-case. First, a term is added to the stereo
decoder matrix's X coefficient so that a centered sound is reduced by -3dB on
each output channel. Panning in front creates a similar gain response to the
typical
L = sqrt(1-pan)
R = sqrt(pan)
for pan = [0,1]. Panning behind the listener can reduce (up to) an additional
-10dB, creating a audible difference between front and back sounds as if
simulating head obstruction.
Secondly, as a special-case, the source positions are warped when calculating
the ambisonic coefficients so that full left panning is reached at -30 degrees
and full right at +30 degrees. This is to retain the expected 60-degree stereo
width. This warping does not apply to B-Format buffer input, although it
otherwise has the same gain responses.
The default number of auxiliary effect slots is now 64. This can still be
raised by the config file without a hard maximum, but incurs processing cost
for each effect slot generated by the app.
The default number of source sends is now actually 2, as per the EFX docs.
However, it can be raised up to 16 via ALC_MAX_AUXILIARY_SENDS attribute
requests, rather than the previous 4.
