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Author SHA1 Message Date
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
c0f22132aa tests: add resample test 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
d5d1ac6f56 defs: update 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
Jan Schmidt
5cc88fe610 win32: update win32 exports for new API 2016-03-25 01:13:54 +11:00
Jan Schmidt
fd2a14144a subparse: WebVTT parsing support 
WebVTT is a new subtitle format for HTML5 video. In this first
version of the parser the cue settings are parsed but only stored in
the internal parser state structure. Later on these settings could be
part of the GstBuffer metadata.

https://bugzilla.gnome.org/show_bug.cgi?id=629764
 2016-03-25 00:58:42 +11:00
Jan Schmidt
ecb8d2e023 typefind: Add a typefinder for WebVTT files 2016-03-25 00:58:41 +11:00
Jan Schmidt
468111ee49 typefind: Reduce URI typefinder from MAX to LIKELY 
Don't claim maximum likelihood for anything that starts
with text that looks like a uri, it's too broad.
 2016-03-25 00:58:41 +11:00
Jan Schmidt
fd92bdf894 decodebin2: Hold new buffering_post lock while posting msgs 
There's a small window between decodebin choosing a buffering level
to post and another thread choosing a different buffering level
where things can race. Close that window by holding a new lock
that's only for posting buffering messages - like what was done
in multiqueue.

https://bugzilla.gnome.org/show_bug.cgi?id=764020
 2016-03-24 15:01:15 +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
Jimmy Ohn
65f721b326 codec-utils: Add utilities for AAC and the AACHead header 
Add utilities about the channels and sample rate for AAC.

https://bugzilla.gnome.org/show_bug.cgi?id=749110
 2016-03-24 14:27:21 +02:00
Jimmy Ohn
090d0d1961 decodebin: Modify result of seekable in check_upstream_seekable function 
In check_upstream_seekable function, it returns FALSE value even though
we already declare about the seekable variable. So, This patch return
result of seekable in check_upstream_seekable function.

https://bugzilla.gnome.org/show_bug.cgi?id=763975
 2016-03-24 14:26:23 +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