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.
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.
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.
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.
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.
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=788362https://bugzilla.gnome.org/show_bug.cgi?id=788361
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
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.
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.
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.
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.
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).
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.
