Commit graph

1432 commits

Author SHA1 Message Date
Niels De Graef
93daa1435a Use G_DEFINE_AUTOPTR_CLEANUP_FUNC unconditionally
Since we started depending on GLib 2.44, we can be sure this macro is
defined (it will be a no-op on compilers that don't support it). For
plugins we should just start using `G_DECLARE_FINAL_TYPE` which means we
no longer need the macro there, but for most types in base/gst-libs we
don't want to break ABI, which means it's better to just keep it like it
is (and use the `#ifdef` instead).
2019-06-04 20:31:09 -04:00
Mathieu Duponchelle
31ac4f4665 gstaudioaggregator: expose output-buffer-duration-fraction
The code for this is mostly lifted from audiobuffersplit, it
allows use cases such as keeping the buffers output by compositor
on one branch and audiomixer on another perfectly aligned, by
requiring the compositor to output a n/d frame rate, and setting
output-buffer-duration to d/n on the audiomixer.

The old output-buffer-duration property now simply maps to its
fractional counterpart, the last set property wins.
2019-05-16 02:55:14 +02:00
Thibault Saunier
287897e465 doc: Fix some gtk-doc comments 2019-05-13 11:34:08 -04:00
Thibault Saunier
685731e989 meson: Add variables for gir files
And flatten list of sources for dependencies
2019-05-13 10:19:22 -04:00
Sebastian Dröge
03a85de734 libs: Fix various Since markers 2019-04-23 12:28:26 +00:00
Sebastian Dröge
e96d105e8d audioaggregator: Add Since: 1.14 markers to all public structs 2019-04-23 12:28:26 +00:00
Tim-Philipp Müller
413b7168da audiometa: fix g-i warning
gstaudiometa.c:382: Warning: GstAudio: gst_buffer_add_audio_meta: return value: Invalid non-constant return of bare structure or union; register as boxed type or (skip)
2019-03-23 20:08:56 +00:00
Tim-Philipp Müller
8d1122013b audiodecoder: add _finish_subframe() method
This allows us to output audio samples without discarding
any input frames, which is useful for some formats/codecs
(e.g. the MonkeysAudio decoder implementation in ffmpeg
which will might return e.g. 16 output buffers for an
input buffer for certain files).

In the past decoder implementations just concatenated
the returned audio buffers until a full frame had been
decoded, but that's no longer possible to do efficiently
when the decoder returns audio samples in non-interleaved
layout.

Allowing subframes to be output before the entire input
frame is decoded can also be useful to decrease startup
latency/delay.

https://gitlab.freedesktop.org/gstreamer/gst-libav/issues/49
2019-03-05 19:49:13 +00:00
mrk501
361835979e audioringbuffer: Fix wrong memcpy address when reordering channels
When using multichannel audio data and being needed to reorder channels,
audio data is not copied correctly because destination address of
memcpy is wrong.

For example, the following command
$ gst-launch-1.0 pulsesrc ! audio/x-raw,channels=6,format=S16LE ! filesink location=test.raw
will reproduce this issue if there is 6-ch audio input device.

This commit fixes that.

The detailed process of this issue is as follows:
1. gst-launch-1.0 calls gst_pulsesrc_prepare (gst-plugins-good/ext/pulse/pulsesrc.c)

   1466 gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
   1467 {
   (skip...)
   1480   {
   1481     GstAudioRingBufferSpec s = *spec;
   1482     const pa_channel_map *m;
   1483
   1484     m = pa_stream_get_channel_map (pulsesrc->stream);
   1485     gst_pulse_channel_map_to_gst (m, &s);
   1486     gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
   1487         (pulsesrc)->ringbuffer, s.info.position);
   1488   }

   In my environment, after line 1485 is processed, position of spec and s are
     spec->info.position[0] = 0
     spec->info.position[1] = 1
     spec->info.position[2] = 2
     spec->info.position[3] = 6
     spec->info.position[4] = 7
     spec->info.position[5] = 8

     s.info.position[0] = 0
     s.info.position[1] = 6
     s.info.position[2] = 2
     s.info.position[3] = 1
     s.info.position[4] = 7
     s.info.position[5] = 8

   The values of spec->info.positions equal
   GST_AUDIO_BASE_SRC(pulsesrc)->ringbuffer->spec->info.positions.

2. gst_audio_ring_buffer_set_channel_positions calls
   gst_audio_get_channel_reorder_map.

3. Arguments of gst_audio_get_channel_reorder_map are
    from = s.info.position
    to = GST_AUDIO_BASE_SRC(pulsesrc)->ringbuffer->spec->info.positions

   At the end of this function, reorder_map is set to
     reorder_map[0] = 0
     reorder_map[1] = 3
     reorder_map[2] = 2
     reorder_map[3] = 1
     reorder_map[4] = 4
     reorder_map[5] = 5

4. Go back to gst_audio_ring_buffer_set_channel_positions and
   2065       buf->need_reorder = TRUE;
   is processed.

5. Finally, in gst_audio_ring_buffer_read,

   1821     if (need_reorder) {
   (skip...)
   1829           memcpy (data + i * bpf + reorder_map[j] * bps, ptr + j * bps, bps);

   is processed and makes this issue.
2019-01-29 14:49:19 +00:00
Tim-Philipp Müller
4c06e9e6eb audiometa: fix docs typo 2019-01-06 00:48:56 +00:00
Mathieu Duponchelle
1edb2c4242 audio-converter: add API to determine passthrough mode
audioconvert's passthrough status can no longer be determined
strictly from input / output caps equality, as a mix-matrix can
now be specified.

We now call gst_base_transform_set_passthrough dynamically, based
on the return from the new gst_audio_converter_is_passthrough()
API, which takes the mix matrix into account.
2018-12-17 14:23:49 +00:00
Edward Hervey
d42294114f audiobasesink: Remove dead assignment
out_samples is set and used in the 'no_align' block.
Dead assignment since 3e312e6e16
2018-12-17 12:21:01 +01:00
Marouen Ghodhbane
0f3efc4b84 audio-convert: Fix endianness conversion function init
Endianness conversion should be based on the sample width instead of the
sample depth.

Fixes #510
2018-11-30 09:14:33 +00:00
Jordan Petridis
2229d53f60
Run gst-indent through the files
This is required before we enabled an indent test in the CI.

https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/33
2018-11-28 05:51:53 +02:00
Tomasz Andrzejak
e0268c02ab audiodecoder: add API for setting caps on the source pad
This patch adds API in the audio decoder base class for setting the arbitrary
caps on the source pad.  Previously only caps converted from audio info were
possible.  This is particularly useful when subclass wants to set caps features
for audio decoder producing metadata.
2018-11-21 10:11:40 +00:00
Sebastian Dröge
d3a35870a2 audio: const gpointer is not the same as gconstpointer/const void *
See https://bugzilla.gnome.org/show_bug.cgi?id=664491
2018-11-05 08:16:16 +00:00
Seungha Yang
3499d9ea64 meson: Replace empty configuration_data() with copy keyword
Use 'copy' keyword to avoid meson warning message.
Note that 'copy' keyword in configure_file() is available
since meson 0.47.0

https://bugzilla.gnome.org/show_bug.cgi?id=797298
2018-10-17 13:48:47 +01:00
Nirbheek Chauhan
d002cd33d3 gstaudioutilsprivate: Fix warnings while setting thread priority
Also use G_OS_WIN32 instead of _WIN32 for clarity.
2018-09-24 19:44:28 +05:30
Tim-Philipp Müller
dc29bc4e13 libs: fix API export/import and 'inconsistent linkage' on MSVC
For each lib we build export its own API in headers when we're
building it, otherwise import the API from the headers.

This fixes linker warnings on Windows when building with MSVC.

The problem was that we had defined all GST_*_API decorators
unconditionally to GST_EXPORT. This was intentional and only
supposed to be temporary, but caused linker warnings because
we tell the linker that we want to export all symbols even
those from externall DLLs, and when the linker notices that
they were in external DLLS and not present locally it warns.

What we need to do when building each library is: export
the library's own symbols and import all other symbols. To
this end we define e.g. BUILDING_GST_FOO and then we define
the GST_FOO_API decorator either to export or to import
symbols depending on whether BUILDING_GST_FOO is set or not.
That way external users of each library API automatically
get the import.

While we're at it, add new GST_API_EXPORT in config.h and use
that for GST_*_API decorators instead of GST_EXPORT.

The right export define depends on the toolchain and whether
we're using -fvisibility=hidden or not, so it's better to set it
to the right thing directly than hard-coding a compiler whitelist
in the public header.

We put the export define into config.h instead of passing it via the
command line to the compiler because it might contain spaces and brackets
and in the autotools scenario we'd have to pass that through multiple
layers of plumbing and Makefile/shell escaping and we're just not going
to be *that* lucky.

The export define is only used if we're compiling our lib, not by external
users of the lib headers, so it's not a problem to put it into config.h

Also, this means all .c files of libs need to include config.h
to get the export marker defined, so fix up a few that didn't
include config.h.

This commit depends on a common submodule commit that makes gst-glib-gen.mak
add an #include "config.h" to generated enum/marshal .c files for the
autotools build.

https://bugzilla.gnome.org/show_bug.cgi?id=797185
2018-09-24 08:45:34 +01:00
Nirbheek Chauhan
1733233060 gstaudiosrc/sink: Set audio ringbuffer thread priority
On Windows, the ringbuffer thread function must have the "Pro Audio"
priority set, otherwise it sometimes doesn't get scheduled for
200-300ms, which will immediately cause an underrun unless you set
a very high latency-time and buffer-time.

This has no compile-time deps since it tries to load avrt.dll at
runtime to set the thread priority.
2018-09-11 00:41:59 +05:30
Nirbheek Chauhan
a9cab426d0 meson: Maintain macOS ABI through dylib versioning
Requires Meson 0.48, but the feature will be ignored on older versions
so it's safe to add it without bumping the requirement.

Documentation:
https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
2018-08-31 14:40:43 +05:30
Tim-Philipp Müller
4906ec8c13 audio: use right export decorator 2018-08-26 11:16:10 +02:00
Sebastian Dröge
0bf207aa53 audioaggregator: Also run the audio-specific caps fixation for audio aggregator subclasses that can't convert 2018-08-16 18:03:37 +03:00
Sebastian Dröge
320243050b audioaggregator: Fixate to some meaningful values if no sinkpad is configured yet
The default caps fixation code would select a rate of 1 for example,
which is not really ideal.
2018-08-16 18:00:24 +03:00
Sebastian Dröge
1b6eed694c audioaggregator: Properly propagate caps negotiation failures
Otherwise we'll end up doing a division by zero when clipping buffers,
and might even accept buffers for which we don't know the caps.

https://bugzilla.gnome.org/show_bug.cgi?id=796951
2018-08-14 10:24:33 +03:00
Tim-Philipp Müller
ca15315565 gst-libs: include config.h in all source files
This will be needed later when we get our export define from config.h
2018-08-13 09:23:34 +01:00
Bastian Köcher
efa9bdccf9 meson: fix install dir for generated header files
Nixos installs into a non-standard includedir, so need
to take account of the 'includedir' option instead of
just hard-coding 'include' here.

https://bugzilla.gnome.org/show_bug.cgi?id=794856
2018-08-10 12:43:38 +01:00
George Kiagiadakis
ab2548d78d audio-buffer: fix typo in assignment that causes buggy behavior 2018-07-24 15:09:25 +03:00
George Kiagiadakis
0ce20cef4f gstaudiodecoder: take into account GstAudioMeta::samples on the output buffers
This is useful if the output buffers are planar and have extra padding
on each plane, in which case size/bpf does not represent the number of
valid samples.

https://bugzilla.gnome.org/show_bug.cgi?id=705977
2018-07-23 15:27:08 +03:00
George Kiagiadakis
2d38d2f1d3 gstaudiodecoder: do not aggregate output if buffers are planar
Aggregation will break the layout, as it concatenates buffers,
and fixing it here would be much more inefficient than configuring
the actual decoder implementation to output larger buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=705977
2018-07-23 15:27:08 +03:00
George Kiagiadakis
e1bc49923f libs: audio: implement planar buffer support in gst_audio_buffer_reorder_channels()
https://bugzilla.gnome.org/show_bug.cgi?id=796743
2018-07-12 13:38:27 +03:00
George Kiagiadakis
b33d70e97f libs: audio: add a new gst_audio_buffer_truncate() function
Essentially this moves the truncation logic out of gst_audio_buffer_clip()
so that it can be used in other places, like in audiorate.

https://bugzilla.gnome.org/show_bug.cgi?id=796740
2018-07-12 12:08:10 +03:00
George Kiagiadakis
9cb09e7269 libs: audio: implement support for non-interleaved audio in gst_audio_buffer_clip()
https://bugzilla.gnome.org/show_bug.cgi?id=796740
2018-07-12 11:59:06 +03:00
George Kiagiadakis
060ecd16cd libs: audio-converter: complete code to support non-interleaved audio buffers
https://bugzilla.gnome.org/show_bug.cgi?id=705986
2018-07-11 16:26:13 +03:00
George Kiagiadakis
eefdf32d96 libs: audio-resampler: add support for consuming non-interleaved input buffers
https://bugzilla.gnome.org/show_bug.cgi?id=705986
2018-07-11 16:26:13 +03:00
George Kiagiadakis
108a911610 libs: audio-channel-mixer: add support for non-interleaved audio buffers
https://bugzilla.gnome.org/show_bug.cgi?id=705986
2018-07-11 16:26:13 +03:00
George Kiagiadakis
c946e323f6 libs: audio: Implement GstAudioBuffer & GstAudioMeta
Library bits to support non-interleaved audio

https://bugzilla.gnome.org/show_bug.cgi?id=751605
2018-07-03 14:06:43 +03:00
wangzq
9f51607723 audiobasesrc: Round down segsize to an integer number of samples
https://bugzilla.gnome.org/show_bug.cgi?id=796704
2018-06-29 07:38:20 +02:00
Tim-Philipp Müller
fae8c24590 audio: Update for g_type_class_add_private() deprecation in recent GLib
https://gitlab.gnome.org/GNOME/glib/merge_requests/7
2018-06-23 21:49:48 +02:00
Thomas Bluemel
7d3c098a7c audiobasesink: Improve clock skew corrections.
The external time should be moved only as much as needed
to get back to the ideal center point, so that the clock
is still allowed to drift both directions after the correction.
This reduces excessive back and forth corrections that were
caused by the assumption of a linear drift.

https://bugzilla.gnome.org/show_bug.cgi?id=788006
2018-06-06 16:11:45 -04:00
Mark Nauwelaerts
751e9640f9 audio: fix some GIR array annotations 2018-05-21 09:18:35 +02:00
Antoine Jacoutot
c765649505 libs: g-ir-scanner: do not hardcode libtool path
https://bugzilla.gnome.org/show_bug.cgi?id=726571
2018-05-18 13:41:25 +02:00
Olivier Crête
8583f17e62 audioaggregator: Remove custom get_next_time implementation
GstAggregator now offers  same thing in a common implementation.

https://bugzilla.gnome.org/show_bug.cgi?id=795486
2018-05-16 22:22:29 +02:00
Sebastian Dröge
5b736d2c7a audioaggregator: Update converters after updating with the new audioinfo/caps
Otherwise subclasses might accidentially use the old audioinfo/caps.
None of the subclasses currently uses the audioinfo/caps, but future
subclasses might.

https://bugzilla.gnome.org/show_bug.cgi?id=795827
2018-05-05 16:40:32 +02:00
Mark Nauwelaerts
9a360a47bf audio: fix some GIR annotations
Mostly related to out and array parameters
2018-04-23 19:33:19 +02:00
Mathieu Duponchelle
83939c81e7 audioaggregator: fix filtered getcaps
In the situation described in
https://bugzilla.gnome.org/show_bug.cgi?id=795397,

downstream_caps consists of two structures, the first with
the preferred rate, if at all possible (44100), the second
containing the full range of allowed rates, as audioresample
correctly tries to negotiate passthrough caps.

As audioaggregator cannot perform rate conversion, it wants
to return a fixated rate in its getcaps implementation,
however it previously directly used the first structure in
the caps allowed downstream, without taking the filter into
consideration, to determine the rate to fixate to.

With this, we first intersect our downstream caps with the
filter, in order not to fixate to an unsupported rate.
2018-04-23 17:13:22 +02:00
Mathieu Duponchelle
a59fbba141 audioaggregator: unref converted buffer after gst_buffer_replace 2018-04-13 01:07:21 +02:00
Nirbheek Chauhan
b5698995f1 audiobasesrc: posting errors should be always be safe
Don't try to signal an error in the ringbuffer if it hasn't been
allocated yet.

https://bugzilla.gnome.org/show_bug.cgi?id=794611
2018-04-09 17:25:32 +05:30
Nirbheek Chauhan
baadc3b302 audioringbuffer: Don't spam INFO for every buffer
This makes GST_DEBUG=4 outputs too spammy, and such frequent messages
are meant to go into DEBUG or TRACE anyway.
2018-04-07 11:09:58 +05:30
Edward Hervey
22c9e5f7c1 libs: Documentation cleanup
* Fix wrong naming, wrong types and typos
* Add missing sections
* Add missing documentation for entries
* Explicitely mark private structure entries
* Remove items that never existed
2018-04-02 08:53:28 +02:00