Commit graph

1464 commits

Author SHA1 Message Date
Havard Graff
0826fb95b7 audio: video: Optimize by using cached quark for meta tag
Avoid taking the global quark lock for every single buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/295>
2020-06-27 09:23:10 +00:00
Sebastian Dröge
63933da9e8 audiodecoder: Add max-errors property
The number of consecutive decode errors that should be tolerated before
returning flow error should be up to the application, not the element.

Hence max-error should be exposed as a property.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/720>
2020-06-23 07:17:00 +00:00
Sebastian Dröge
f2af205a78 Fix up and add various "Since" markers and other related docs fixes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/713>
2020-06-19 12:17:55 +03:00
Guillaume Desmottes
008d72d5da audio: add missing space in GST_AUDIO_FORMATS_ALL
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/694>
2020-06-10 10:43:42 +02:00
Guillaume Desmottes
e2f6b85fd9 audio: sort formats by quality
Will ensure that we pick the "best" format when negotiating caps.

Fix #649

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/689>
2020-06-09 08:09:58 +00:00
Guillaume Desmottes
02fd2f12f9 audio: add gst_audio_make_raw_caps()
More binding friendly version of GST_AUDIO_CAPS_MAKE().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/676>
2020-06-02 11:57:42 +00:00
Guillaume Desmottes
58a6303a5f audio-format: remove empty space prefix from GST_AUDIO_FORMATS_ALL
This space prevent deserialization using gst_value_deserialize().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/676>
2020-06-02 11:57:42 +00:00
Guillaume Desmottes
75411ce1e7 audio-format: add gst_audio_formats_raw()
The existing GST_AUDIO_FORMATS_ALL macro is not binding friendly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/676>
2020-06-02 11:57:42 +00:00
Seungha Yang
4a774e878f audiosink: Keep baseclass extensible
Add a structure for future extension.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/716
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/547>
2020-05-28 19:14:29 +09:00
Jan Schmidt
fd3942d06b audiodecoder: Handle instant-rate-change event
When receiving an instant-rate-change event, store the updated
seek flags and replace the flags in any input segments with them
to allow for instant switching between trickmodes and not.
2020-04-01 21:01:38 +00:00
Jan Schmidt
f9c5db7d56 audiobasesink: Handle an extra case of buffers being out of segment
It's possible that a buffer might be within the segment proper,
but not within the "valid" part we're playing, which is only
things after the 'offset' part of the segment. In that case,
the running-times of the buffer-start and buffer-stop will be
GST_CLOCK_TIME_NONE, and we'd better not schedule playback that
far in the future.
2020-04-01 21:01:38 +00:00
Niels De Graef
21a107294d streamvolume: Use G_DECLARE_INTERFACE 2020-03-20 06:20:43 +00:00
Guillaume Desmottes
545d0b144f audio: annotate @buf in finish_frame methods 2020-03-18 15:38:25 +01:00
Niels De Graef
ec84cf92f9 *aggregator: Add g_autoptr support for *ConvertPad 2020-03-16 15:47:58 +00:00
Jonas Holmberg
af909c6d82 audioencoder: fix segment event leak
Segment event was leaked if format != _TIME.
2019-12-20 12:43:35 +00:00
Jochen Henneberg
33ae846607 audioringbuffer: Reset reorder flag before check
This function might be revisited with different channel position mapping
while audio source goes into play so the reorder flag needs to be reset
before the checks happen.
2019-11-17 14:10:31 +00:00
Sebastian Dröge
89f613abf5 audio-buffer: Don't fail to map buffers with zero samples
Instead initialize the map infos, etc to NULL like gst_buffer_map()
would be doing on a zero-sized buffer.

This fixes a crash in audioresample if the first output buffer would
contain zero samples.
2019-11-14 14:47:44 +01:00
Seungha Yang
2f89c3aff1 audio-info: Allow from_caps() with encoded audio format
Similar to gst_video_info_from_caps() which allows encoded video format,
don't error gst_audio_info_from_caps() with encoded audio format.
Because gst_audio_info_set_format() supports encoded format, current
behavior does not seem to be consistent.
2019-10-25 12:32:03 +09:00
Tim-Philipp Müller
289d8e53e2 Remove autotools build system 2019-10-13 14:15:43 +01:00
Axel Mårtensson
feb1e24347 audiosink: fix resuming after pause
For resuming after paused, gst_audio_sink_ring_buffer_start() needs to
be called to notify the ringbuffer to continue to play.
2019-09-27 05:34:57 +00:00
Philippe Renon
0dc1b6049e audiosink: expose more audioringbuffer vmethods to child sinks
The newly exposed vmethods are pause, resume, stop and clear_all.
The existing reset vmethod is deprecated.

The audio sink will fallback to calling reset if pause or stop
are not provided and will fallback to calling start if
resume is not provided. There is no default clear_all
implementation.
Existing audio sinks continue to work as before.

This change is useful for sinks that need to distinguish
between a pause and a stop (currently both are handled
by a reset) and is needed for https://bugzilla.gnome.org/show_bug.cgi?id=788362

https://bugzilla.gnome.org/show_bug.cgi?id=788361
2019-09-27 05:34:57 +00:00
Nirbheek Chauhan
6f7c9e43bc audio: Use LoadPackagedLibrary when building for UWP
Universal Windows Platform apps are not allowed to use LoadLibrary to
load arbitrary DLLs from the filesystem. They can only use
LoadPackagedLibrary to load DLLs that have been packaged with the app
as assets.

See also: https://gitlab.freedesktop.org/gstreamer/gstreamer/merge_requests/190
2019-09-24 15:17:39 +00:00
Doug Nazar
0c955c16ce audio-resampler: Update NEON to handle remainders not multiples of 4
If the remainder is not evenly divisable by 4, we'd miss the check
for zero and continue the loop until crashing. Change the branch
to take into account negatives as well.

This more closely matches the SSE loop.
2019-09-02 23:25:39 -04:00
Thibault Saunier
909baa2360 Pass the code through codespell 2019-08-30 13:05:36 +00:00
Mathieu Duponchelle
4ccc7a51d1 {audio,video}aggregator: define autoptr cleanup functions 2019-08-28 14:52:22 +00:00
Hou Qi
b65d1c6de9 audiodecoder: fix ctitical info assertion 'GST_IS_CAPS (dec->priv->ctx.caps)' failed
Matroskademux will send gap event when lag of video and audio is over 3 seconds.
audiodecoder needs to handle gap event and set default output caps.
Only audio info is set, while output caps is ignored. This cause the assertion failed.

Need to fill output caps in gst_audio_decoder_negotiate_default_caps() with
negotiated caps to avoid critical info printed when check it later.
2019-08-28 00:59:56 +00:00
Mathieu Duponchelle
f65145371b audioaggregator: add missing Since tag 2019-08-12 19:11:06 +02:00
Sebastian Dröge
1ec1123178 audioaggregator: Split getcaps() function into two
One for convert pads and one for normal sink pads.
2019-07-18 08:46:42 +03:00
Sebastian Dröge
0a21c28484 audioaggregator: Always take first configure pad's rate and downstream caps into account when calculating allow sink caps
While we can convert between all formats apart from the rate, we
actually need to make sure that we comply with a) the rate of the first
configured pad and b) also all the allowed rates from downstream.
2019-07-18 08:43:14 +03:00
Sebastian Dröge
7080d216a8 audioaggregator: If we don't have a GstAudioAggregatorConvertPad, don't assume that we can actually convert 2019-07-18 08:43:14 +03:00
Mathieu Duponchelle
bced52d2e8 audioaggregator: always use downstream's rate requirements
We were previously only fixating the rate in the getcaps
implementation when downstream was requiring a discrete value,
causing negotiation to fail when upstream was capable of rate
conversion, but not made aware that it had to occur.

Instead of fixating the rate, we can simply update our sink
template caps with whatever GValue the downstream caps are holding
as their rate field.

Allows negotiation to successfully complete with pipelines such as:

audiotestsrc ! audio/x-raw, rate=48000 ! audioresample ! audiomixer name=m ! \
audio/x-raw, rate={800, 1000} ! autoaudiosink \
audiotestsrc ! audio/x-raw, rate=44100 ! audioresample ! m.
2019-07-18 08:43:14 +03:00
Doug Nazar
fb842a3fdb audiodecoder: Fix leak on failed audio gaps
If we fail to process the gap event we need to unref the event or
we end up with a leak.
2019-06-26 03:51:03 -04:00
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