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1295 commits

Author SHA1 Message Date
Wim Taymans
19f7d9ca46 audio-resampler: estimate memory usage in auto mode 
Estimate the memory usage and use this to decide between full or
interpolated filter.
 2016-03-28 15:37:36 +02:00
Wim Taymans
984ee8a3f6 audio-resampler: small optimizations 2016-03-28 13:25:55 +02:00
Wim Taymans
cf9059f070 audio-resampler: improve non-interleaved flags 
Make it possible to have different interleaving on input and output
because we can quite trivially do that.
 2016-03-28 13:25:55 +02:00
Wim Taymans
33855f0fe1 audio-resampler: unroll some more loops 
Unroll some loops.
 2016-03-28 13:25:55 +02:00
Wim Taymans
90a41b81dc audio-resampler: keep precision 
Transpose and add before applying the cubic interpolation to avoid
overflows when using full precision.
 2016-03-28 13:25:55 +02:00
Wim Taymans
cc9d8594fe audio-resampler: small cleanups 2016-03-28 13:25:55 +02:00
Wim Taymans
e209c0d565 audio-resampler: optimize no resampling 
Switch to the faster nearest resample method when are doing no rate
conversion.
 2016-03-28 13:25:54 +02:00
Wim Taymans
f692d5e459 audio-resampler: add VARIABLE_RATE flag 
Add a VARIABLE rate flag that selects an interpolating filter.
Move some function setup code in the _new function.
 2016-03-28 13:25:54 +02:00
Wim Taymans
7bb149dcc1 audio-resampler: more neon optimizations 2016-03-28 13:25:54 +02:00
Wim Taymans
6dd5e5259f audio-resampler: avoid overflow in cubic interpolation 
Shift out an extra bit to have some more headroom when doing cubic
interpolation.
 2016-03-28 13:25:54 +02:00
Wim Taymans
61460fdfad audio-resampler: overread only 8 taps 
We only need 8 taps of zeroes as headroom for the SIMD optimized
functions.
 2016-03-28 13:25:54 +02:00
Wim Taymans
4772ebbddf audio-converter: use helper to check intermediate format 2016-03-28 13:25:54 +02:00
Wim Taymans
00e5a8bab8 audio-resampler: fix phase 2016-03-28 13:25:54 +02:00
Wim Taymans
9182ea17b5 audio-resampler: fix neon assembler 2016-03-28 13:25:53 +02:00
Wim Taymans
027165621b audio-resampler: avoid some format conversion 
Store the filter in the desired sample format so that we can simply do a
linear or cubic interpolation to get the new filter instead of having to
go through gdouble and then convert.
 2016-03-28 13:25:53 +02:00
Wim Taymans
2c33c2134c audio-resampler: fix neon linear float interpolation 2016-03-28 13:25:53 +02:00
Wim Taymans
d969a7a9d8 audio-resampler: reorder filter coefficients for more speed 
Reorder the filter coefficients to make it easier to use SIMD for
interpolation.
Fix orc flags a little.
Add specialized nearest resampling function.
 2016-03-28 13:25:53 +02:00
Wim Taymans
107f53ea0a audio-resampler: remove stereo optimizations 
The stereo optimizations don't give enough benefit.
Rename none to full to make it clear that we use a full filter instead
of an interpolated one
 2016-03-28 13:25:53 +02:00
Wim Taymans
b820074a49 audio-resample: remove neon double stubs 
NEON does not have double types.
 2016-03-28 13:25:53 +02:00
Wim Taymans
6f9237dfb5 audio-resampler: add more neon optimizations 2016-03-28 13:25:53 +02:00
Wim Taymans
307f360cca audio-resampler: add more neon optimizations 2016-03-28 13:25:53 +02:00
Wim Taymans
d5abdd83c9 audio-resampler: add neon optimizations 
Unroll some more loops in the fallback code that seems to work fine
for ARM.
Add some simple ARM optimizations taken from speex.
 2016-03-28 13:25:53 +02:00
Wim Taymans
25d81ffb55 audio-resampler: give better hints about the precision 
Give better hints to the compiler about the precision we expect from
the multiplications.
 2016-03-28 13:25:53 +02:00
Wim Taymans
ea497b509f audio-resample: small optimizations 
Remove some inline functions that are called in the slow path.
Unroll C fallback functions a little.
 2016-03-28 13:25:52 +02:00
Wim Taymans
167a415717 audio-resampler: Use n_phases when calculating taps offset 
Tweak linear interpolation oversampling.
Clear filter cache on rate changes when using a full filter.
 2016-03-28 13:25:52 +02:00
Wim Taymans
524ea147cc audio-resampler: improve filter construction 
Remove some unused variables from the inner product functions.
Make filter coefficients by interpolating if required.
Rename some fields.
Try hard to not recalculate filters when just chaging the rate.
Add more proprties to audioresample.
 2016-03-28 13:25:52 +02:00
Wim Taymans
0f3ff9177f audio-resampler: avoid overflow in fraction calculation 2016-03-28 13:25:52 +02:00
Wim Taymans
651ae201bc audio-resampler: increase precision 2016-03-28 13:25:52 +02:00
Wim Taymans
4cb52f1831 audio-resampler: add more optimizations 2016-03-28 13:25:52 +02:00
Wim Taymans
bdf194a09a audio-resample: fix taps conversion 
We do taps conversion in place so make sure we don't overwrite the
input with temporary data.
Optimize some more gint16 functions.
 2016-03-28 13:25:52 +02:00
Wim Taymans
f6e0481ab5 audio-resampler: Improve taps memory layout 
Rearrange the oversampled taps in memory to make it easier to use
SIMD instructions on them. this simplifies some sse code.
Add some more optimizations
 2016-03-28 13:25:52 +02:00
Wim Taymans
e9fc039bb1 audio-resampler: add cubic interpolation 2016-03-28 13:25:52 +02:00
Wim Taymans
58dcd0587d audio-resampler: add more functions 
Use some macros to generate more functions
 2016-03-28 13:25:51 +02:00
Wim Taymans
e02af5c534 audio-resampler: add linear interpolation method 
Make more functions into macros.
Add linear interpolation of filter coefficients.
 2016-03-28 13:25:51 +02:00
Wim Taymans
05d238def9 audio-resampler: add max-phase-error config 2016-03-28 13:25:51 +02:00
Wim Taymans
13e5b986cd audio-resampler: improve tap calculation 
Return the taps from make_taps, this makes it possible to not actually
have to cache the taps when we want to.
Fix overflow in phase calculation.
 2016-03-28 13:25:51 +02:00
Wim Taymans
6397db74cd audio-resampler: fix guint -> gint 2016-03-28 13:25:51 +02:00
Wim Taymans
45574ba4f4 audio-resampler: improve phase error 
Accept a phase error of maximum 10%, which turns out to be inaudible.
 2016-03-28 13:25:51 +02:00
Wim Taymans
b0b3350717 audio-resampler: improve phase calculation 
Also calculate the GCD with the current phase so that we can accurately
represent the current phase with the new resample rates.
 2016-03-28 13:25:51 +02:00
Wim Taymans
bbdb447b2b audio-resampler: fix history after buffer resize 
When we resize the temp buffer, move the history in its new place.
 2016-03-28 13:25:51 +02:00
Wim Taymans
ed747492ef audio-resampler: add reset function 
Add a function to reset the audio-resampler.
Use new function in audio-converter
Use the new functions in gstaudioresample and fixup drain functions.
 2016-03-28 13:25:51 +02:00
Wim Taymans
ea469ad9a8 audio-resampler: Small fixes 
Fix the phase.
Reset the new sample buffer with 0.
Move samples around when we change the filter size.
 2016-03-28 13:25:51 +02:00
Wim Taymans
a489f9ddb3 audio-resampler: Rework make_taps 
Make it return a pointer to the generated taps. That way we can later
decide to actually cache it or not.
 2016-03-28 13:25:51 +02:00
Wim Taymans
05eb109c0d audio-resampler: handle filter length changes 
Update the buffer with history samples when the filter length changes
because of an update of the parameters or sample rates.
 2016-03-28 13:25:51 +02:00
Wim Taymans
8dfb3ffb99 audio-resampler: fix samples_avail 
We only know the taps after we calculate them.
 2016-03-28 13:25:51 +02:00
Wim Taymans
c8fc9d88a7 audio-resampler: work on dynamically changing the samplerate 
Calculate the new phase for the new sample rate.
Fix some docs.
 2016-03-28 13:25:51 +02:00
Wim Taymans
4e48867097 audio-resampler: small cleanups 2016-03-28 13:25:51 +02:00
Wim Taymans
85c77659b9 audio-resampler: add fallback to mono function 
Remove stereo implementations. Implement fall back to mono functions
when the stereo function is missing.
 2016-03-28 13:25:50 +02:00
Wim Taymans
2555317a71 audio-resampler: add float stereo SSE function 2016-03-28 13:25:50 +02:00
Wim Taymans
e74c207433 audio-resampler: Fix compilation of intrinsics 
Only compile intrinsics when we are building for the selected
architecture.
Add sse4.1 optimized int32 resampler code.
 2016-03-28 13:25:50 +02:00
Wim Taymans
98bd349b88 audioconvert: only resample on supported formats 2016-03-28 13:25:50 +02:00
Wim Taymans
d348fbb9b9 audio-converter: make some optimized functions 
Make an optimized function that just calls the resampler when possible.
Optimize the resampler transform_size function a little.
 2016-03-28 13:25:50 +02:00
Wim Taymans
23531bdc93 audio-resampler: remove mirror function 
We don't need to mirror the input, just assume 0 samples.
Always move the processed samples to the start of the buffer.
Add some G_LIKELY
 2016-03-28 13:25:50 +02:00
Wim Taymans
6f685410b1 audio-resampler: also enable sse when sse2 is available 2016-03-28 13:25:50 +02:00
Wim Taymans
71871c5048 audio-resampler: optimizations 
Improve int16 resampling by using pmaddwd
Use intrinsics to scale and pack int16 samples
Align the coefficients so that we can use aligned loads
Add padding to taps and samples so that we don't have to use partial
loads for the remainder of the loops.
Remove copy_n, we can reuse the plain copy function with some new
parameters.
Align and pad the sample array.
 2016-03-28 13:25:50 +02:00
Wim Taymans
f55a67ca7c audio-resampler: make pluggable optimized functions 
Add support for x86 specialized functions and select them at runtime.
 2016-03-28 13:25:50 +02:00
Wim Taymans
819c4c26c7 audio-resampler: combine functions 2016-03-28 13:25:50 +02:00
Wim Taymans
de37491662 audio-converter: simplify API 
Remove the consumed/produced output fields from the resampler and
converter. Let the caler specify the right number of input/output
samples so we can be more optimal.
Use just one function to update the converter configuration.
Simplify some things internally.
Make it possible to use writable input as temp space in audioconvert.
 2016-03-28 13:25:50 +02:00
Wim Taymans
1d9a793545 audio-converter: more work on resampling 
- Fix the resampler in the audio converter
- fix memory leaks
 2016-03-28 13:13:59 +02:00
Wim Taymans
75d668e152 audio-converter: add resampler 
Add a resampler to the processing chain when needed.
port the audio resampler to the new audioconverter library
 2016-03-28 13:13:59 +02:00
Tim-Philipp Müller
f4fb623aba audiodecoder: avoid unnecessary gst_pad_has_current_caps() checks 
No need to do this for each input buffer, we have the input caps
stored somewhere already.

https://bugzilla.gnome.org/show_bug.cgi?id=763337
 2016-03-24 14:49:12 +02:00
Vineeth TM
44b70ca3a1 base: use new gst_element_class_add_static_pad_template() 
https://bugzilla.gnome.org/show_bug.cgi?id=763075
 2016-03-24 14:25:41 +02:00
Wim Taymans
c0ef1ea553 audio-channel-mixer: improve non-interleaved flags 
Make separate flags for non-interleaved input and output because the
channel mixer should be able to convert between the two layouts in the
future.
 2016-03-04 17:17:33 +01:00
Wim Taymans
221e661f07 audio-quantize: fix feedback dither 
Make sure we allocated enough extra space in the error buffer to
store the feedback error.
 2016-02-24 14:57:31 +01:00
Wim Taymans
3e8cf31a96 audio-converter: perform dithering on the current format 
Use the current (intermediate) format to decide how to set up dithering
instead of the input format.
 2016-02-24 14:57:31 +01:00
Tim-Philipp Müller
ddfe7a2808 win32: remove outdated build cruft 
This hasn't been touched for generations, doesn't work,
and is just causing confusion. We also don't want to
maintain these files manually.
 2016-02-20 10:05:17 +00:00
Wim Taymans
5cef3f31ad audio-converter: make a copy if we can't write in unpack 
If we don't have writable memory, make sure to make a copy of the input
samples into a temporary (writable) buffer, even if we are dealing with
a native intermediate format that we don't need to call the unpack
function for.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=761655
 2016-02-10 12:51:23 +01:00
HoonHee Lee
dfa2f49523 audio/videodecoder: Minor cleanup of last commit 
https://bugzilla.gnome.org/show_bug.cgi?id=761218
 2016-01-28 13:21:49 +01:00
HoonHee Lee
15df3c812b audio/videodecoder: use gst_pad_peer_query_caps to make output caps 
gst_pad_get_allowed_caps() will return NULL if the srcpad has no peer.
In that case, use gst_pad_peer_query_caps() with template caps as filter
to have negotiated output caps properly before forwarding GAP event.

https://bugzilla.gnome.org/show_bug.cgi?id=761218
 2016-01-28 11:34:22 +01:00
Wim Taymans
03566e5002 audio-converter: add reset function 2016-01-26 17:19:34 +01:00
Wim Taymans
2d971df593 audio-converter: handle NULL input 
Allow NULL as input to mean silence samples.
 2016-01-26 17:19:34 +01:00
Wim Taymans
6050509b65 audio-converter: improve _update_config 
Allow NULL config to keep the existing parameters.
Fix the docs.
 2016-01-26 17:19:34 +01:00
Wim Taymans
0f757bc23c audio-converter: audio-converter: make some optimized functions 
Make optimized functions for generic and passthrough conversion.
 2016-01-26 17:19:34 +01:00
Wim Taymans
cde091ae81 audio-quantize: add _reset function 
Add a reset function that clears any history.
 2016-01-26 16:45:44 +01:00
Wim Taymans
3674742957 audio-converter: ensure correct alignment of samples 
Make sure that the data we allocate for our temporary buffers is
properly aligned.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=760938
 2016-01-21 16:27:50 +01:00
Sebastian Dröge
761142e15a audioencoder: Add note to the documentation about various settings being reset before set_format() 
It's quite unexpected behaviour that various subclass settings are just
reset before set_format(). Unfortunately changing this now has the risk
of breaking existing code but we should reconsider this for 2.0.
 2016-01-16 11:05:13 +01:00
Wim Taymans
1b412a523d audio-channel-mixer: round before truncating 
Round the result before truncating for int channel mixing.
 2016-01-12 15:56:36 +01:00
Wim Taymans
ef3844cf6f audio-converter: Avoid conversion when possible 
When the input and output formats are the same and in a possible
intermediate format, avoid unpack and pack.
Never do passthrough channel mixing.
Only do dithering and noise shaping in S32 format
 2016-01-12 15:27:16 +01:00
Wim Taymans
4d47d43a13 audio-channel-mixer: add more formats 
Add support for float and int16 mixing
Remove in-place processing, this simplifies things as we won't be using it.
Don't do clipping for float audio formats
 2016-01-12 11:43:20 +01:00
Wim Taymans
8a8b12189e audio-converter: improve processing loop 
Process as many samples as we can from the input and return the number
of processed samples from the chain. This simplifies some code.
Fix the IN_WRITABLE handling, don't overwrite the flags.
 2016-01-12 11:37:17 +01:00
Wim Taymans
85afad72ec audio-converter: small API tweaks 
Pass flags in _converter_new() so that we can configure ourselves
differently depending on some options.
SOURCE_WRITABLE -> IN_WRITABLE because the array is called 'in'
 2016-01-08 17:34:50 +01:00
Wim Taymans
7f49b946cc audio-converter: prepare API for rate changes 
Use the update function to update the sample rates along with the config
once we implement resampling.
 2016-01-08 17:28:31 +01:00
Wim Taymans
980163457e audio-convert: simplify API 
Simplify the API, we don't need the consumed and produced output
arguments. The caller needs to use the _get_in_frames/get_out_frames API
to check how much input is needed and how much output will be produced.
 2016-01-08 17:19:58 +01:00
Sebastian Dröge
0da2709d0c audio/video: Use G_GNUC_INTERNAL for internal functions 2016-01-08 17:50:50 +02:00
Wim Taymans
40f4c5e352 audio: GstAudioChannelMix -> GstAudioChannelMixer 
Rename the GstAudioChannelMix object to GstAudioChannelMixer because it
looks better and to avoid a conflict with a library in -bad.
 2016-01-08 16:41:17 +01:00
Stefan Sauer
7bbfa39ada audioconvert: fix passthrough operation 
We did not take the sample size into account. Rearrange the tests to have more
conversion test and an extra test case for passthrough operations.

Fixes #759890
 2015-12-29 14:40:32 +01:00
Stefan Sauer
0bd3f818bb audio-converter: code cleanup 
Rename samples to num_samples, since we also have samples in chain, but that is
the data pointer. Always use gzize for num_samples. Make the log output a bit
more homogenous.
 2015-12-27 19:25:20 +01:00
Sebastian Dröge
3459bd6854 audio: Fix some documentation warnings 
Remove/rename function parameters and skip some functions that can't
be used by bindings as they are now.
 2015-12-26 09:43:56 +01:00
Wim Taymans
08734e7598 audio-converter: rework the main processing loop 
Rework the main processing loop. We now create an audio processing
chain from small core functions. This is very similar to how the
video-converter core works and allows us to statically calculate an
optimal allocation strategy for all possible combinations of operations.
Make sure we support non-interleaved data everywhere.
Add functions to calculate in and out frames and latency.
 2015-12-16 11:13:15 +01:00
Xavier Claessens
429860e51f base: Add g_autoptr() support to all types 
https://bugzilla.gnome.org/show_bug.cgi?id=754464
 2015-12-14 13:39:43 -05:00
Wim Taymans
f5a3f70571 audio: adapt API for non-interleaved formats 
Allow an array of sample blocks to be passed to the channel mix and
quantizer functions to support non-interleaved formats.
 2015-12-14 09:16:08 +01:00
Wim Taymans
aec17c63fd audio-converter: improve API for non-interleaved formats 
Make it possible to pass an array of sample blocks when dealing with
non-interleaved formats.
 2015-12-14 09:16:08 +01:00
Wim Taymans
5e55968546 audio-convert: improve converter API 
Improve the converter API to allow for an max input and output number of
samples and return the number of consumed/produced samples.
 2015-12-09 17:16:26 +01:00
Reynaldo H. Verdejo Pinochet
4ed7b0a0e6 Drop usage of deprecated g-ir-scanner --strip-prefix flag 2015-12-02 20:19:43 -08:00
Sebastian Dröge
2f3eb47a95 audiobasesrc: Post latency message on the bus after set_caps() 
The latency is only known once the caps are known, and might change
whenever the caps are changing.

https://bugzilla.gnome.org/show_bug.cgi?id=758911
 2015-12-01 19:58:25 +02:00
Michael Olbrich
43155807cd audiobasesink: Post latency message on the bus after set_caps() 
Any latency query before this will not get the correct latency so a new
latency query should be triggered once the audio sink know its own latency.

Without this the initial latency query from the pipeline arrives too early
sometimes and the resulting latency is too short.

https://bugzilla.gnome.org/show_bug.cgi?id=758911
 2015-12-01 19:58:25 +02:00
Luis de Bethencourt
df16e8dd5a audio-converter: remove unneeded check for unsigned < 0 
Commit ff6d1a2a25 changed sample's type from
gint to gsize (and renamed it to in_samples). gsize is an unsigned long,
which means it can never be a negative value and the check making sure that
in_samples is >= 0 is never going to be false. Removing it.

CID 1338689
 2015-11-12 14:18:30 +00:00
Vineeth TM
b61e1465b7 audio-quantize: Fix dither_buffer memory leak 
https://bugzilla.gnome.org/show_bug.cgi?id=757928
 2015-11-11 15:01:08 +01:00
Wim Taymans
ff6d1a2a25 audio-converter: add output size argument 
Make it possible to have a different number of output samples than input
samples when we, for example, want to add resampling later.
 2015-11-10 09:53:59 +01:00
Wim Taymans
30977cf1a5 audio-converter: require interleaved samples and no resampling 
We can't yet do resampling or anything other than interleaved audio.
 2015-11-06 18:00:41 +01:00