remixing-produce-lfe controls upmixing, and remixing-consume-lfe
controls downmixing. The motivation is that a user might want to
synthesize LFE while playing stereo audio on his/her 5.1 speakers,
but at the same time follow the industry recommendation to omit
the LFE channel when producting a stereo downmix (e.g. for headphones)
from 5.1 content. Or the other way round.
Fixes: #753.
The resampler framework just forwards the request to the lfe filter.
There are no resampler impl that can rewind yet, so just reset the
resampler impl instead of properly rewinding yet.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Add a user defined parameter lfe-crossover-freq for the lfe-filter,
to pass this parameter to the lfe-filter, we need to change the
pa_resampler_new() API as well.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
When enable-lfe-remixing is set, an LFE channel is present in the
resampler's destination channel map but not in the source channel map,
we insert a low-pass filter instead of just averaging the channels.
Other channels will get a high-pass filter.
In this patch, the crossover frequency is hardcoded to 120Hz (to be fixed
in later patches).
Note that in current state the LFE filter is
- not very optimised
- not rewind friendly (rewinding can cause audible artifacts)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Added ID and names for the resampler presets and also updated the working sample rate deduction to take the new resampler into account. The initial libsoxr backend version does not variable rate resampling, so it is disabled in this case.
FSF addresses used in PA sources are no longer valid and rpmlint
generates numerous warnings during packaging because of this.
This patch changes all FSF addresses to FSF web page according to
the GPL how-to: https://www.gnu.org/licenses/gpl-howto.en.html
Done automatically by sed-ing through sources.
some resampler implementations (e.g. libsamplerate and ffmpeg) do not consume
the entire input buffer; the impl_resample() function now has a return value
returning the number of frames in the input buffer not processed
these frames must be saved in appropriate buffer and presented together with
new input data
also change the parameter names from in_samples, out_samples to in_n_frames,
out_n_frames, respectively (n_frames = samples / channels)
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
is to allocate all audio memory blocks from a per-process memory pool which is
available as read-only SHM segment to other local processes. Then, instead of
writing the actual audio data to the socket just write references to this
shared memory pool.
To work optimally all memory blocks should now be of type PA_MEMBLOCK_POOL or
PA_MEMBLOCK_POOL_EXTERNAL. The function pa_memblock_new() now generates memory
blocks of this type by default.
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1266 fefdeb5f-60dc-0310-8127-8f9354f1896f
