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.
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)
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
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.
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.
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.
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
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.
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
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
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
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
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.
