Commit graph

1309 commits

Author SHA1 Message Date
Mikhail Fludkov
8d4f79b640 audiodecoder: fix invalid timestamps when PLC and delay
Elements inherited from GstAudioDecoder, supporting PLC and introducing
delay produce invalid timestamps. Good example is opusdec with in-band FEC
enabled. After receiving GAP event it delays the audio concealment until
the next buffer arrives. The next buffer will have DISCONT flag set which
will make GstAudioDecoder to reset it's internal state, thus forgetting
the timestamp of GAP event. As a result the concealed audio will have the
timestamp of the next buffer (with DISCONT flag) but not the timestamp
from the event.
2016-06-16 11:01:04 +01:00
Tim-Philipp Müller
d52a74f32e g-i: pass compiler env to g-ir-scanner
It's what introspection.mak does as well. Should
fix spurious build failures on gnome-continuous.
2016-05-24 00:44:21 +01:00
Kipp Cannon
f7a31a79f4 audio: Add const to segment parameter of gst_audio_buffer_clip()
e.g., allows this to be used with the reference retrieved by
gst_event_parse_segment().

https://bugzilla.gnome.org/show_bug.cgi?id=765663
2016-04-27 12:26:07 +03:00
Jan Schmidt
802eae296a Revert "audioringbuffer: start ringbuffer if needed upon commit"
This reverts commit 13ee94ef10.

Causes audio glitches at startup by starting to output segments
from the ringbuffer before it has been filled / fully prerolled.

https://bugzilla.gnome.org/show_bug.cgi?id=657076
2016-04-16 02:13:15 +10:00
Guillaume Desmottes
7c5dfd713c audioringbuffer: don't attempt to reorder position-less channels
As said in its doc GST_AUDIO_CHANNEL_POSITION_NONE is meant to be used
for "position-less channels, e.g. from a sound card that records 1024
channels; mutually exclusive with any other channel position".

But at the moment using such positions would raise a
'g_return_if_reached' warning as gst_audio_get_channel_reorder_map()
would reject it.

Fix this by preventing any attempt to reorder in such case as that's not
what we want anyway.

https://bugzilla.gnome.org/show_bug.cgi?id=763799
2016-04-12 14:48:30 -04:00
Guillaume Desmottes
1c56cfa144 audio: add debug output if channels mapping does not match
https://bugzilla.gnome.org/show_bug.cgi?id=763985
2016-04-12 14:48:30 -04:00
Guillaume Desmottes
3cb08304da gst-audio: add gst_audio_channel_positions_to_string()
We currently don't log much about channel positions making debugging
harder as it should be. This is the first step in my attempt to improve
this.

https://bugzilla.gnome.org/show_bug.cgi?id=763985
2016-04-12 14:48:30 -04:00
Fabrice Bellet
bfcd0737b7 audio: Fix a race with the audioringbuffer thread
There is a small window of time where the audio ringbuffer thread
can access the parent thread variable, before it's initialized
by the parent thread. The patch replaces this variable use by
g_thread_self().

https://bugzilla.gnome.org/show_bug.cgi?id=764865
2016-04-11 21:43:13 +10:00
Víctor Manuel Jáquez Leal
37c4915109 libs: audio: split allocation query caps and pad caps
Since the allocation query caps contains memory size and the pad's caps
contains the display size, an audio encoder or decoder might need to allocate
a different buffer size than the size negotiated in the caps.

This patch splits this logic distinction for audiodecoder and audioencoder.

Thus the user, if needs a different allocation caps, should set it through
gst_audio_{encoder,decoder}_set_allocation_cap() before calling the negotiate()
vmethod. Otherwise the allocation_caps will be the same as the caps in the
src pad.

https://bugzilla.gnome.org/show_bug.cgi?id=764421
2016-04-05 11:37:15 +02:00
Víctor Manuel Jáquez Leal
052fe11949 audioencoder: fix gtk-doc comment format 2016-04-04 17:12:16 +02:00
Alessandro Decina
74efde50ad audio-resampler: disable neon on arm64
Fix the build on arm64 by using HAVE_ARM_NEON instead of __ARM_NEON__.
2016-03-30 11:16:49 +11:00
Sebastian Dröge
0582d5a1bc audio-resampler: Use _mm_set_epi64x(0, x) instead of _mm_cvtsi64_si128(x) in more places 2016-03-29 12:11:48 +03:00
Sebastian Dröge
38a5a3614e resampler: Use _mm_set_epi64x(0, x) instead of _mm_cvtsi64_si128(x)
The latter is only available on x86-64 for some reason.
2016-03-29 10:15:07 +03:00
Edward Hervey
de2ded9557 audio: Fix distcheck
Don't forget to dist the needed files (which don't need to be installed)
2016-03-29 08:22:29 +02:00
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