Commit graph

1483 commits

Author SHA1 Message Date
Stefan Sauer
5fd4e71da3 audioaggregator: reduce variable scope
This is a non-functional change that makes the code more alike to the previous
check. I should be more obvious when we drop a buffer.
2017-10-02 17:01:36 +02:00
Stefan Sauer
17848d7c0c audioaggregator: update comment for aggregate
Replace collect-pads left-over. Remove first paragraph, we're not doing this.
Remove 3), 4) since this is not per pad.
2017-10-02 17:01:36 +02:00
Thibault Saunier
c610076d85 meson: Add mssing GstBase-1.0 include in the gir generation 2017-09-28 21:56:29 -03:00
Sebastian Dröge
bf68e74403 audio: Add stream align API for getting timestamp at discont and number of samples since discont
https://bugzilla.gnome.org/show_bug.cgi?id=787560
2017-09-28 14:06:24 +03:00
Sebastian Dröge
d2fd740388 audio: Add reverse playback support to GstAudioStreamAlign
https://bugzilla.gnome.org/show_bug.cgi?id=787560
2017-09-28 14:06:07 +03:00
Sebastian Dröge
ec1e20ffe5 audio: Add helper object for audio discontinuity detection and sample alignment
This is the same code that is in decklinkaudiosrc, audioringbuffer,
audiomixer and various other places. Have it once instead of copying it
everywhere.

https://bugzilla.gnome.org/show_bug.cgi?id=787560
2017-09-28 14:06:05 +03:00
Mathieu Duponchelle
4196e67ff1 [API]: GST_AUDIO_CONVERTER_OPT_MIX_MATRIX
Taken from audiomixmatrix, credits to Vivia Nikolaidou

https://bugzilla.gnome.org/show_bug.cgi?id=785471
2017-09-22 16:19:59 +02:00
Mathieu Duponchelle
877d6faeea [API]: gst_audio_channel_mixer_new_with_matrix
+ Refactor previous constructor to call on that new constructor

+ Reimplement is_passthrough to strictly check whether the matrix
  is an identity matrix, comparing channel-masks was incorrect:
  the mixer may be remixing from a list of positions to the same
  list of positions, but ordered differently, and reciprocally,
  the mixer may be remixing from a list of positions to another
  list of positions identically ordered

+ Remove unused tmp field, must have been a refactoring leftover

https://bugzilla.gnome.org/show_bug.cgi?id=785471
2017-09-22 16:19:58 +02:00
Stefan Sauer
ceee4db9d0 audioaggregator: fix typo in comment 2017-09-17 21:06:48 -07:00
Tim-Philipp Müller
ece4a096a1 libs: add some more missing GST_EXPORT 2017-08-10 10:27:23 +01:00
Mark Nauwelaerts
00fa39befa audioencoder: also adjust sample count upon discont to avoid ts overflow
Only adjusting the base_ts might lead to a negative ts and as such integer
overflow into a huge timestamp which then propagates into the granulepos
and so on.  Instead, resync to incoming buffer timestamp using both base_ts
and sample count rather than only base_ts.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=785948
2017-08-09 09:32:55 +02:00
Stefan Sauer
4080b844c0 audioaggregator: use local var
Instead of the self->priv-> deref use the local var we created already.
2017-07-26 09:43:52 +02:00
Tim-Philipp Müller
d95e62be82 audio: mark symbols explicitly for export with GST_EXPORT 2017-07-18 12:46:57 +01:00
Stefan Sauer
35a4149caf aggregator: code cleanups
Fix comment typos, some copy'n'paste in logging. Add more doc comments.
2017-07-13 22:02:17 +02:00
Olivier Crête
930c3cea40 audioaggregate: Don't hold object locks across calls to aggregate_one
https://bugzilla.gnome.org/show_bug.cgi?id=782878
2017-05-21 18:44:53 +02:00
Olivier Crête
55ab23cc30 audioaggregator: Use downstream allocator and params if available
https://bugzilla.gnome.org/show_bug.cgi?id=746529
2017-05-21 13:24:29 +02:00
Olivier Crête
8a302f1815 aggregator: Remove unused GST_FLOW_NOT_HANDLED 2017-05-20 16:21:17 +02:00
Matthew Waters
57673e608e aggregator: add simple support for caps handling
Modelled off the videoaggregator caps handling as that seems the most
mature aggregtor-using implementation that has caps handling there is.

https://bugzilla.gnome.org/show_bug.cgi?id=776931
2017-05-20 16:21:17 +02:00
Olivier Crête
26819ba296 aggregator: Delay clipping to output thread
This is required because the synchronized events like caps or segments
may only be processed on the output thread.

https://bugzilla.gnome.org/show_bug.cgi?id=781673
2017-05-20 16:21:16 +02:00
Olivier Crête
1e7aaf775f aggregator: Simplify clip function
The return value was ignored anyway

https://bugzilla.gnome.org/show_bug.cgi?id=781673
2017-05-20 16:21:16 +02:00
Sebastian Dröge
2a8784e218 audioclock: Sink the reference in the constructor
This is now needed as GstClock does not do that internally anymore,
because that broke bindings.

And mark the function correctly as (transfer full), which it already was
before.

https://bugzilla.gnome.org/show_bug.cgi?id=743062
2017-05-17 10:40:52 +03:00
Nicolas Dufresne
1ceb40cd1e audioringbuffer: Accept MPEG 1 layer 3 version 2.5
https://bugzilla.gnome.org/show_bug.cgi?id=781929
2017-05-16 15:37:16 -04:00
Tim-Philipp Müller
d7dd381f78 audio: mark symbols explicitly for export with GST_EXPORT 2017-05-16 15:22:26 +01:00
Tim-Philipp Müller
60e9629860 g-i: no need to load registry in g-i scanner 2017-05-04 23:55:20 +01:00
Sebastian Dröge
a3656ba29b libs: Check if meta transform_func is NULL before using it
https://bugzilla.gnome.org/show_bug.cgi?id=782050
2017-05-02 14:31:14 +03:00
Tim-Philipp Müller
5185cce994 audio: resampler: fix typos in docs 2017-04-13 16:40:02 +01:00
Rico Tzschichholz
07a427b403 meson: Pass --c-include accordingly to GIR builds 2017-04-13 12:23:32 +01:00
Sebastian Dröge
8b468c3c36 audio: Generate audiobasesink/src and audiocdsrc GLib enums automatically
And ensure that GstAudioBaseSrcSlaveMethod's re-timestamp stays
re-timestamp and doesn't become retimestamp.
2017-04-09 11:49:50 +03:00
Thibault Saunier
099ac9faf2 docs: Convert gtkdoc comments to markdown
Modernizing the documentation, making it simpler to read an
modify and allowing us to possibly switch to hotdoc in the
future.
2017-03-10 18:19:17 -03:00
Jan Schmidt
5903e2dfbb audioringbuffer: Also add FLAC to debug strings.
Oops, also add FLAC to the debug strings array.

https://bugzilla.gnome.org/show_bug.cgi?id=777655
2017-02-04 14:46:42 +11:00
Jan Schmidt
04c14de1d1 audioringbuffer: Prevent overflow of debug names array
Add new audio types to the list of strings used for debug
so we don't index past the end of that array.

https://bugzilla.gnome.org/show_bug.cgi?id=777655
2017-02-04 14:42:33 +11:00
Sebastian Dröge
f939f05304 audio-resampler: Fix integer overflow in clamping code
https://bugzilla.gnome.org/show_bug.cgi?id=777921
2017-01-30 12:35:04 +02:00
Tim-Philipp Müller
83e84d5acf audio: add since markers to docs for new enums
https://bugzilla.gnome.org/show_bug.cgi?id=777655
2017-01-23 20:04:54 +00:00
Vincent Penquerc'h
6134dab3bb audio: add FLAC to GstAudioRingBufferFormatType
https://bugzilla.gnome.org/show_bug.cgi?id=777655
2017-01-23 13:47:39 -05:00
Olivier Crete
c46607095e audioringbuffer: Also support raw AAC
Support raw AAC streams without the ADTS header

https://bugzilla.gnome.org/show_bug.cgi?id=777655
2017-01-23 13:45:50 -05:00
Evan Nemerson
98064ed9bf audioringbuffer: add set_callback_full() for g-i
https://bugzilla.gnome.org/show_bug.cgi?id=678301
2016-12-22 15:34:58 +00:00
Thibault Saunier
8bbf67c37d audio: Fix introspection annotation
In gst_audio_check_valid_channel_positions the mask
is an out parameter.

And minor conversion from a print to a GST_ERROR.
2016-12-16 11:27:31 -03:00
Vincent Penquerc'h
6ee5922f2f audioringbuffer: do not require 4 byte multiple for encoded MPEG
Bytes per frame doesn't make sense for encoded audio.

https://bugzilla.gnome.org/show_bug.cgi?id=776038
2016-12-13 10:16:07 +00:00
Sebastian Dröge
71e819ae7d audio-converter: In passthrough, also don't copy if in and out block are the same
In and out array are usually different, they are stack allocated arrays.
However the blocks inside them still can be the same.

https://bugzilla.gnome.org/show_bug.cgi?id=775369
2016-11-30 10:43:50 +02:00
Petr Kulhavy
010b9547d3 audio-converter: optimize endian conversion
Optimize LE<->BE conversion by adding a dedicated fast path instead of
using the generic converter. Implement transform_ip function in order to do the
endian swap in place.

This saves buffer allocation for the intermediate format, can be done in place
and also performs the conversion in one step instead of unpack-convert-pack.

For all bit widths the naive algorithm is implemented, which provides the best
performance when compiled with -O3. ORC was considered but eventually removed
as it requires a dedicated function for in-place conversion (due to the
"restrict" parameters).

A more complex algorithm for the 24-bit conversion with unrolled loop and
32-bit processing is implemented in the #if 0 section. It performs better if
compiled with -O2. With -O3 however the naive algorithm performs better.

https://bugzilla.gnome.org/show_bug.cgi?id=773073
2016-11-28 17:24:17 +02:00
Petr Kulhavy
640c54d8f8 audio-convert: simplify the chain free process
It is not needed to store a pointer to every single chain element to free it.
Instead walk the channel list backwards and free the chain elements one by one.

Rename GstAudioConverter->chain_pack to chain_end.

https://bugzilla.gnome.org/show_bug.cgi?id=773073
2016-11-28 17:24:13 +02:00
Scott D Phillips
e740103669 videodecoder, audiodecoder: parse format before checking in src_query_default
The logic change in these commits misordered the parsing and checking of
format in position queries:

 2b06e54 videodecoder: Don't answer BYTES queries
 1840b02 audio: Don't answer BYTES queries

https://bugzilla.gnome.org/show_bug.cgi?id=774484
2016-11-16 07:57:31 +11:00
Jan Schmidt
1840b0233a audio: Don't answer BYTES queries
Refuse to answer BYTES queries ourselves. The only
time they make sense is on raw elementary streams,
in which case upstream would already have answered.

They especially don't make sense for encoders to answer
based on upstream values - although perhaps later
we could make it do TIME->BYTES conversion on the source
pad based on bitrate.

https://bugzilla.gnome.org/show_bug.cgi?id=757631
2016-11-16 00:14:47 +11:00
Scott D Phillips
6e71583d07 Cast away const from GstMetaInfo in *_get_meta_info() functions
MSVC warns about the const in the implicit argument conversion in the
calls to g_once_init_{enter,leave}. It's OK so explicitly cast it.

https://bugzilla.gnome.org/show_bug.cgi?id=774293
2016-11-15 14:52:22 +02:00
Thibault Saunier
5bdb38995f meson: Fix build when orc is disabled
Making sure not to use the orc_dep variable in case
orc has been explicitely disabled.
2016-11-14 17:26:02 -03:00
Thibault Saunier
6917cb629e meson: Generate girs
https://bugzilla.gnome.org/show_bug.cgi?id=773944
2016-11-09 18:06:19 -03:00
Petr Kulhavy
54f4d3772c audio-channels: map buffer read-write only if channels differ
gst_audio_buffer_reorder_channels() was always mapping the buffer read-write
regardless whether any reordering was needed.  If the from and to channel order
is identical return immediately without remapping the buffer.

Add a small helper function gst_audio_channel_positions_equal() which is used
in both gst_audio_reorder_channels() and gst_audio_buffer_reorder_channels().

https://bugzilla.gnome.org/show_bug.cgi?id=773833
2016-11-09 19:42:47 +02:00
Nicolas Dufresne
eddb543719 audiosrc: Leave read loop if no longer running
In the case a src stops providing data (read calls returns 0). The audio
src thread will never leave. Instead, check the condition and leave the
loop.
2016-11-03 13:19:12 -04:00
Marcin Kolny
89e711663f audioclock: use GstAudioClock* as first argument in GstAudioClock methods
All the GstAudioClock method declarations required object of GstClock type
as a first argument, but in fact, required GstAudioClock object (runtime
check in function body). Instead of checking type in run-time, we can
change functions declaration, to accept only GstAudioClock methods. Then,
runtime check is not necessary anymore, since always GstAudioClock object
is passed to a function.

https://bugzilla.gnome.org/show_bug.cgi?id=756628
2016-11-01 19:54:01 +02:00
Nirbheek Chauhan
20ced51df2 Fix incorrect return type in several functions
All these should return GstFlowReturn, not gboolean
2016-10-27 23:06:26 +05:30
Thibault Saunier
9842a6708d Revert "meson: Use the new pic argument on static libs"
This reverts commit e3c22605ae.

pic was added after 0.35 and will be present in 0.36 (meson documentation
was wrong).
2016-10-20 17:17:58 -03:00
Thibault Saunier
e3c22605ae meson: Use the new pic argument on static libs
We depend on 0.35 already
2016-10-20 16:07:33 -03:00
Stefan Sauer
7986a8ed3d audio: don't deref NULL
gst_buffer_copy_region() can return NULL when the buffer meta-data is invalid.

See https://bugzilla.gnome.org/show_bug.cgi?id=772200
2016-10-20 15:42:50 +02:00
Havard Graff
a32b25db6a audioencoder: Error-handling for pushing headers
https://bugzilla.gnome.org/show_bug.cgi?id=773105
2016-10-20 14:12:53 +03:00
Havard Graff
63c1f4aa33 audioencoder: Plug buffer-leak
https://bugzilla.gnome.org/show_bug.cgi?id=773107
2016-10-20 14:09:49 +03:00
Stefan Sauer
409534f242 audio: fix doc string again.
There was a second '*' at the start of the line. Reword + reformat to make it
obvious.
2016-10-17 09:47:54 +02:00
Nirbheek Chauhan
69abe1de13 meson: Add missing audio-enumtypes.h dep in audio-resampler static libs
Seen on the Jenkins CI:

FAILED: subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o
ccache cc  '-Isubprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta' '-fdiagnostics-color=always' '-I../subprojects/gst-plugins-base/gst-libs/gst/audio' '-Isubprojects/gst-plugins-base/gst-libs/gst/audio' '-Isubprojects/gst-plugins-base/.' '-I../subprojects/gst-plugins-base/.' '-Isubprojects/gst-plugins-base/gst-libs' '-I../subprojects/gst-plugins-base/gst-libs' '-Isubprojects/gstreamer/libs' '-I../subprojects/gstreamer/libs' '-Isubprojects/gstreamer/.' '-I../subprojects/gstreamer/.' '-pipe' '-Wall' '-Winvalid-pch' '-DHAVE_CONFIG_H' '-msse4.1' '-fPIC' '-O0' '-g' '-fPIC' '-I/usr/include/glib-2.0' '-I/usr/lib/glib-2.0/include' '-pthread' '-Isubprojects/gstreamer/gst' '-MMD' '-MQ' 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o' '-MF' 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o.d' -o 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o' -c ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.c
In file included from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h:24:0,
                 from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-private.h:23,
                 from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-macros.h:25,
                 from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.h:23,
                 from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.c:24:
../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h:26:39: fatal error: gst/audio/audio-enumtypes.h: No such file or directory
 #include <gst/audio/audio-enumtypes.h>
                                       ^
compilation terminated.
2016-10-15 21:48:39 +05:30
Stefan Sauer
86d18fa9ff audio: fix typo in doc string 2016-10-14 14:21:52 +02:00
Arun Raghavan
87905cc48b meson: Enable SSE intrinsics in audio-resampler
This files need to be built with the specific C flags for the
corresponding processor optimisations.
2016-10-03 11:01:36 +05:30
Arun Raghavan
4de66632d7 meson: Enable Orc in build
Top-level meson.build code updated from gst-plugins-good.
2016-10-03 11:01:24 +05:30
Arun Raghavan
20ba0299cc audio-resampler: Add a missing header to noinst_HEADERS 2016-09-29 19:55:45 +05:30
Arun Raghavan
a1ae17ef69 audiorsample: Fix build on 32-bit x86
Turns out _mm_cvtsi128_si64() isn't available on 32-bit, so only build
SSE 4.1 optimisations on x86-64 for now.
2016-09-29 19:45:16 +05:30
Arun Raghavan
4b5f78337a audioresample: Separate out CFLAGS used for SSE* code
This makes sure that we only build files that need explicit SIMD support
with the relevant CFLAGS. This allows the rest of the code to be built
without, and specific SSE* code is only called after runtime checks for
CPU features.

https://bugzilla.gnome.org/show_bug.cgi?id=729276
2016-09-29 18:37:08 +05:30
Arun Raghavan
f4cba79063 audioresample: Fix some gobject introspection warnings 2016-09-28 19:09:26 +05:30
Nirbheek Chauhan
5c4f4ac1bd Add support for Meson as alternative/parallel build system
https://github.com/mesonbuild/meson

With contributions from:

Tim-Philipp Müller <tim@centricular.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)

Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded

... and many more. For more details see:

http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html

Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
2016-08-20 11:09:51 +01:00
Michael Olbrich
44da322083 audioclock: use GST_STIME_FORMAT for the correct argument
GST_STIME_ARGS is used for time_offset not for last_time.
This fixes the format string accordingly.

https://bugzilla.gnome.org/show_bug.cgi?id=768990
2016-07-20 12:28:54 +01:00
Sebastian Dröge
61f59857a7 audio: Ship audio-resampler-neon.h 2016-07-07 00:27:18 +03:00
Sebastian Dröge
e5d7064b29 audiodecoder: Protect samples_in/bytes_out and audio info with object lock
It might cause invalid calculations during the CONVERT query otherwise.
2016-07-04 11:12:55 +02:00
Sebastian Dröge
5cbd1a7bca audioencoder: Protect samples_in/bytes_out and audio info with object lock
It might cause invalid calculations during the CONVERT query otherwise.
2016-07-04 11:12:55 +02:00
Sebastian Dröge
8d8262a00c audioencoder/decoder: Move encoded audio conversion function to a common place
No need to duplicate this non-trivial function.
2016-07-04 11:12:55 +02:00
Vincent Penquerc'h
c76e8c77eb audiodecoder: fix criticals fixating a non existent field
https://bugzilla.gnome.org/show_bug.cgi?id=766970
2016-07-04 10:05:45 +01:00
Sebastian Dröge
5de9d5809b audioconvert: Handle fallback channel mask for mono correctly
It's 0 and no mask should be set for mono at all.

https://bugzilla.gnome.org/show_bug.cgi?id=757472
2016-06-29 18:14:51 +02:00
Guillaume Desmottes
937a42247a audioaggregator: fix buffer leak
If the pad was still owning a buffer when being destroyed it was leaked.

Fix a leak with the test_flush_start_flush_stop test.

https://bugzilla.gnome.org/show_bug.cgi?id=766663
2016-06-21 10:52:47 +03:00
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
Sebastian Dröge
a7b86878fb audio: Move audioaggregator base class to a library
It's useful enough already to be used in other elements for audio aggregation,
let's give people the opportunity to use it and give it some API testing.

https://bugzilla.gnome.org/show_bug.cgi?id=760733
2016-01-22 12:39:48 +02:00
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
Wim Taymans
7abed02858 audio: update ORC dist files 2015-11-06 17:54:21 +01:00
Wim Taymans
e3f0f3b91e audio-converter: move audio converter to audio libs
Move the audio-converter helper to the audio library.
2015-11-06 17:53:22 +01:00
Wim Taymans
dfa25a40fc audio-channel-mix: move channel mixer to audio libs
Move the channel mixer code to the audio library
2015-11-06 17:39:33 +01:00
Wim Taymans
b8bea9d8be audio: add debug categories 2015-11-06 17:29:22 +01:00
Wim Taymans
59db8ce542 audio-quantize: update docs
Update docs
Add another flag for the quantizer
2015-11-06 13:02:19 +01:00
Wim Taymans
dfbeb78342 audio: update orc files 2015-11-06 12:37:14 +01:00
Wim Taymans
c36ac3ce45 audioconvert: move audio quantize code to libs
Move the audio quantize code from audioconvert to the audio library.
work on making an audio converter helper function similar to the video
converter.
Fold fastrandom directly into the quantizer, add some ORC code to
optimize this later.
2015-11-06 12:10:48 +01:00
Wim Taymans
a7789854d5 audio-channels: rename get_default_mask
Rename _get_default_mask() to _get_fallback_mask() to make it more
clear that the function only provides a fallback if nothing else can be
done. Also clarify this in the documentation.

API: gst_audio_channel_get_fallback_mask()
2015-11-05 12:50:18 +01:00
Wim Taymans
f86ed8cdf6 audio-channels: make method to get default channel-mask
Add a new method to get the default channel-mask.
Use the new method on audiodecoder and audioconvert.

API: gst_audio_channel_get_default_mask()
2015-11-05 10:52:53 +01:00
Sebastian Dröge
35ea6fdddf audio: Add GstAudioClippingMeta for specifying clipping on encoded audio buffers
https://bugzilla.gnome.org/show_bug.cgi?id=757153
2015-11-03 20:35:33 +02:00
Tim-Philipp Müller
1f2fdd3789 audio: update disted orc backup files 2015-11-03 16:38:09 +00:00
Luis de Bethencourt
94a7f9fc4e audioclock: use GST_STIME_FORMAT for GstClockTimeDiff
GST_STIME_FORMAT is more appropriate for GstClockTimeDiff since it can
handle negative values better.

https://bugzilla.gnome.org/show_bug.cgi?id=757480
2015-11-03 14:08:29 +00:00