Unless stopRequest is set, we should unlock conditionally -- otherwise,
the 'create:' method can wake up to an empty buffer queue
and pull a nil buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=748054
gst_ks_device_provider_probe() is a no-braier, just runs ks_enumerate_devices()
and reports the results.
Monitoring is a bit more tricky. We have to create a dummy message-processing
window and register device change notifications for it.
As kernel streaming can (and should) be used for audio capture and audio
playback, this change also has certain placeholders for such.
https://bugzilla.gnome.org/show_bug.cgi?id=747757
The autodetection mode was broken because a race condition in the input mode
setting. The mode could be reverted back when it was replaced in
the streaming thread by the old mode in the middle of mode changed callback.
We now fill GErrors for everything that could throw an exception, and method
calls now always return a gboolean and their value in an out-parameter to
distinguish failures from other values.
The shm-area-property tells the name of the shm area used by the element. This
is useful for cases where shmsink is not able to clean up (calling
shm_unlink()), e.g. if it is in a sandbox.
https://bugzilla.gnome.org/show_bug.cgi?id=675134
while having the default vtdec at secondary rank. This allows decodebin/playbin
to prefer the hardware based decoders, and if that fails to initialize because
hardware resources are busy to fall back to e.g. the libav based h264 decoder
instead of the software based vtdec (which is slower), and only fall back to
the software based vtdec if there is no higher ranked decoder available.
Using requestMediaDataWhenReadyOnQueue the layer will execute a block
when it would like more frames. Using this we can provide the current
frame and avoid needlessly filling the layer's buffer queue causing
older frames to be displayed when under resource pressure.
Otherwise we might set bogus values or GST_CLOCK_TIME_NONE.
Also make sure to reset the caps field to NULL after unreffing
the caps to prevent accidential use afterwards, and unref any
old caps before we remember new caps.
Otherwise we will still have a reference to the surface left, which would
prevent activating the sink again later. E.g. after we lost the device.
Hopefully fixes https://bugzilla.gnome.org/show_bug.cgi?id=744615
Add the diff between the external time when we went to playing and
the external time when the pipeline went to playing. Otherwise we
will always start outputting from 0 instead of the current running
time.
gstdecklink.cpp: In member function 'virtual HRESULT GStreamerDecklinkInputCallback::VideoInputFrameArrived(IDeckLinkVideoInputFrame*, IDeckLinkAudioInputPacket*)':
gstdecklink.cpp:498:22: error: comparison between signed and unsigned integer expressions [-Werror=sign-compare]
if (capture_time > m_input->clock_start_time)
^
gstdecklink.cpp:503:22: error: comparison between signed and unsigned integer expressions [-Werror=sign-compare]
if (capture_time > m_input->clock_offset)
^
The driver has an internal buffer of unspecified and unconfigurable size, and
it will pull data from our ring buffer as fast as it can until that is full.
Unfortunately that means that we pull silence from the ringbuffer unless its
size is by conincidence larger than the driver's internal ringbuffer.
The good news is that it's not required to completely fill the buffer for
proper playback. So we now throttle reading from the ringbuffer whenever
the driver has buffered more than half of our ringbuffer size by waiting
on the clock for the amount of time until it has buffered less than that
again.
The ringbuffer's acquire() is too early, and ringbuffer's start() will only be
called after the clock has advanced a bit... which it won't unless we start
scheduled playback.
Not from the decklink clock. Both will return exactly the same time once the
decklink clock got slaved to the pipeline clock and received the first
observation, but until then it will return bogus values. But as both return
exactly the same values, we can as well use the pipeline clock directly.
There is no reason to pre-roll more buffers here as we have our own ringbuffer
with more segments around it, and we can immediately provide more buffers to
OpenSL ES when it requests that from the callback.
Pre-rolling a single buffer before starting is necessary though, as otherwise
we will only output silence.
Lowers latency a bit, depending on latency-time and buffer-time settings.
4 is the "typical" number of buffers defined by Android's OpenSL ES
implementation, and its code is optimized for this. Also because we
have our own ringbuffer around this, we will always have enough
buffering on our side already.
Allows for more efficient processing.
The pseudo buffer pool code was using gst_buffer_is_writable()
alone to try and figure-out if cached buffer could be reused.
It needs to check for memory writability too. Also check map
result and fix map flags.
https://bugzilla.gnome.org/show_bug.cgi?id=734264
Use YUV instead of RGB textures, then convert using the new apple specific
shader in GstGLColorConvert. Also use GLMemory directly instead of using the
GL upload meta, avoiding an extra texture copy we used to have before.
When doing texture sharing we don't need to call CVPixelBufferLockBaseAddress to
map the buffer in CPU. This cuts about 10% relative cpu time from a vtdec !
glimagesink pipeline.
Otherwise we might start the scheduled playback before the audio or video streams are
actually enabled, and then error out later because they are enabled to late.
We enable the streams when getting the caps, which might be *after* we were
set to PLAYING state.
Otherwise we might start the streams before the audio or video streams are
actually enabled, and then error out later because they are enabled to late.
We enable the streams when getting the caps, which might be *after* we were
set to PLAYING state.
This API has been deprecated for eternities and microsoft
stopped shipping the headers in 2010 accoding to wikipedia,
so let's just remove it and focus on bringing the plugins
based on the newer APIs up to snuff.
This fixes handling of flushing seeks, where we will get a PAUSED->PLAYING
state transition after the previous one without actually going to PAUSED
first.
Otherwise we will overflow the internal buffer of the hardware
with useless frames and run into an error. This is necessary until
this bug in basesink is fixed:
https://bugzilla.gnome.org/show_bug.cgi?id=742916
decklinkvideosink might be added later to the pipeline, or its state might
be handled separately from the pipeline. In which case the running time when
we (last) went into PLAYING state will be different from the pipeline's.
However we need our own start time to tell the Decklink API, which running
time should be displayed at the moment we go to PLAYING and start scheduled
rendering.
... and hope that everything will be fine. This shouldn't really happen but
previously happened during shutdown. It should be fixed in videoencoder now,
but better be on the safe side here.
Use AVF provided timings to timestamp output buffers. Use the running time at
the time the first buffer is produced to base timestamps on. Report 1-frame
latency based on the negotiated framerate instead of hardcoding 4ms latency.
The property is in kbit/s and we store it in bit/s, so just multiply and
divide by 1000. No need to put a factor of 8 in there.
kVTCompressionPropertyKey_AverageBitRate is also in bit/s according to
its documentation.
This updates the dshowvideosink to work with the GStreamer 1.0.x APIs, and
avoids the use of deprecated GLib threading API that are now used since
GLib 2.32+
https://bugzilla.gnome.org/show_bug.cgi?id=699364
No idea where the DecklinkAPIDispatch.cpp comes from on Windows,
but this should still work. Will just become a problem once we
use other parts of the API.
Otherwise we're going to starve other elements if the decklink clock
is slower than the pipeline clock, or starts much later.
Of course this will still cause problems if the decklink clock and ours are
completely out of sync, or running at a very different rate. But this at least
works better now.
If we just count the frames and calculate timestamps from that, all frames
will arrive late in the sink as we have a live source here. Instead take
the pipeline clock at capture time as reference.
We have to handle the callback object a bit different:
a) it needs a virtual destructor
b) we need to set the callback to NULL when we're done with the output
c) create a new one every time
https://bugzilla.gnome.org/show_bug.cgi?id=740616
We will run into an assertion in set_caps() if we try to change
caps while the source is already running. Don't try to find new
caps in GstBaseSrc::negotiate() to prevent caps changes.
The object lock only protects the session, as we modify
the session from other threads when the bitrate property
is changed. Don't hold it much longer than for session
related things.
And we need to release the video decoder stream lock before
enqueueing a frames. It might wait for our callback to dequeue
a frame from another thread, which will then take the stream
lock too and deadlock.
It is not required on OSX apparently and was only added in 10.9.6 there.
Calculating the correct level from the configuration is not trivial, so let's
just not set a level at all here.
DVB-T2 supports 5, 10 and 1.712 MHz
Order of the enum values (new values after _AUTO)
has been kept congruent with the one in the v4l
API for consistency
Previously known as DMB-T/H, this is the
terrestial DTV broadcast standard currently
used by the People's Republic of China,
Hong Kong, Laos and Macau (officially),
and by Malaysia, Iraq, Jordan, Syria and
Lebanon (experimentally).
These apply to ISDB-T, DVB-T2 and DTMB
Order of the enum values (new rates after _AUTO)
has been kept congruent with the one in the v4l
API for consistency.
According to the v4l-dvb API docs, these are only
used for DVB-T2 at the moment.
Order of the enum values (new rates after _AUTO)
has been kept congruent with the one in the v4l
API for consistency.
iOS has special stride requirements that we don't know yet, so copy
input buffers into buffers allocated by iOS for now.
Later we should check the stride and probably provide a buffer pool for these
buffers so upstream can directly write in there.
The first buffer does not contain more garbage than any other MP3 decoder
outputs and we don't really know how much we have to drop or not.
After this change the output has the same duration as with mad.
Valid values range from 1 to 7 as stated.
DTV_ISDBT_LAYER_ENABLED bitmask is built from
OR-ing 0x1 0x2 0x4. If all bits are set
(0x00000111 = 7) it means all layers should be
demodulated.
Change avoids attempting to convert to kHz if unneeded.
There are quite some ZAP format variants out there. Among
their subtle little differences, some store transponder
frequencies in Mhz and others in kHz. The latter been the
most common variant.
It works the same as the 'tune' property that is used only to signal
the element that it should tune, but it is more natural to be used
as a signal rather than a property.
It is also proxied at the dvbbasebin element
gst_buffer_pool_get_config() returns a copy to the bufferpool's
configuration, which must be passed to gst_structure_free() after
use if not given away to gst_buffer_pool_set_config().
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734537
gst_pad_get_pad_template_caps() returns a reference which is unreferenced,
so creating a copy using gst_caps_copy() results in a reference leak.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734536
gst_pad_get_pad_template_caps() returns a reference which is unreferenced,
so creating a copy using gst_caps_copy() results in a reference leak. Also
the caps are pushed as an event downstream, but this doesn't consume the
caps so it must still be unreferenced.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734534
gst_pad_get_pad_template_caps() returns a reference which is unreferenced,
so creating a copy using gst_caps_copy() results in a reference leak.
Also remove the incorrect comment to avoid confusion in the future.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734533
ISDB-T and ISDB-Tb (the Brazilian variant) are the
terrestial DTV standards used by Japan, Philippines,
Maldives, Thailand, most South American countries
and Botswana. Changeset adds the set of previously
missing (and required) ISDB-T parameters, adapter
and frontend setup logic and proxies the new
properties on dvbbasebin.
Tested to work with the live aerial broadcast by
Tv Paraíba HD in Campina Grande (Brazil).
https://bugzilla.gnome.org/show_bug.cgi?id=732875
Allows proper tuning around high/low band boundaries when using
non "standard" LNBs.
Not all LNBs (Low noise block down converters) are made equal.
This is particularly true for universal LNBFs, where, even though
there are seemingly standard values for the local oscillator
frequencies, these can vary from manufacturer to manufacturer
and LNB model. Change also proxies the new LNB properties in
dvbbasebin.
https://bugzilla.gnome.org/show_bug.cgi?id=732818
The pixel buffer release callback is called if the void *
dataPtr given to the CVPixelBufferCreateWithPlanarBytes
is not NULL.
According to the documentation dataPtr is supposed to be a
"plane description block" but no specific type is given.
https://bugzilla.gnome.org/show_bug.cgi?id=711847
It was previously a mix and match of both variants, introducing just too much
confusion.
The prefix are from now on:
* GstMpegts for structures and type names (and not GstMpegTs)
* gst_mpegts_ for functions (and not gst_mpeg_ts_)
* GST_MPEGTS_ for enums/flags (and not GST_MPEG_TS_)
* GST_TYPE_MPEGTS_ for types (and not GST_TYPE_MPEG_TS_)
The rationale for chosing that is:
* the namespace is shorter/direct (it's mpegts, not mpeg_ts nor mpeg-ts)
* the namespace is one word under Gst
* it's shorter (yah)
Interestingly, Coverity implies that close takes an unsigned
argument, while my close(2) man page shows it taking a signed
argument. I guess it may be platforms specific.
Coverity 1214602
New approach attempts to be more accurate by measuring
the elapsed time by iteration. Also:
* Use a 10 seconds default timeout and a half a second
polling step. New values should better match the tuning
process on real-life scenarios.
* Correct elapsed_time computation.
* Add _retry_ioctl() to avoid bailing out on temporary
ioctl EINTR failures (no need to check for EAGAIN cause
we are opening the frontend on blocking mode)
* Small corrections to fail condition handling
Check if libnativehelper is loaded in the process and if
it has these awful wrappers for JNI_CreateJavaVM and
JNI_GetCreatedJavaVMs that crash the app if you don't
create a JniInvocation instance first. If it isn't we
just fail here and don't initialize anything.
See this code for reference:
https://android.googlesource.com/platform/libnativehelper/+/master/JniInvocation.cpp
* Drop remaining sleep() logic in favor of polling
* Use best guess delivery system if none is set
* Make tuning/locking timeout configurable
* Add signals for tuning start, done and fail
* Drop gst_dvbsrc_frontend_status(). It was used only
for signal LOCK checking. This is now part of the
tuning/locking loop
* Break up frontend configuration and tuning
on separate functions
Plus:
* Add some more useful DEBUG/TRACE messages
* Move over misplaced DVB API message
* Fix wrong comment for default DVB buffer size (http://linuxtv.org/downloads/v4l-dvb-apis/dmx_fcalls.html#DMX_SET_BUFFER_SIZE)
This patch builds up on previous work done by
Fabrizio (Misto) Milo <mistobaan@gmail.com>
https://bugzilla.gnome.org/show_bug.cgi?id=641204
On Samsung Galaxy S4 it is impossible to have more than one
hardware decoder at the same time. If we do not release it
explicitly the GC only releases it whenever the whole application
is finished not whenever the activity is finished and thus a player
will not be able to work correctly
gst_amc_color_format_copy will copy in/out a frame resides at a
GstAmcBuffer. Lots of codes in gst_amc_video_*_fill_buffer are moved to
this new function.
Some hack logic needs also to be present in create_src|sink_caps, for
working around some broken codecs. These hacks are hidden
in color_format/video_format conversion -- the prototypes of these
functions are also changed to include more args for hack judgement.
Also in case of multi-color_formats mapped to one video_format, then
map that video_format back will not give the original color_format, which
causes gst_amc_codec_configure failed with something like
'does not support color format N'.
The new prototype involves with GstAmcCodecInfo and mime, which
ensures the converted color_format is supported by the codec.
A COLOR_FormatYCbYCr to GST_VIDEO_FORMAT_YUY2 mapping is also added, in
order to work around bugs in OMX.k3.video.decoder.avc(which incorrectly
reports supporting COLOR_FormatYCbYCr, which is actually
COLOR_FormatYUV420SemiPlanar). There are already hacks for this in
gst_amc_video_format_to_color_format, gst_amc_color_format_to_video_format
and gst_amc_color_format_info_set, but the codec will still not work(be
ignored because of "has unknown color formats") without adding this mapping.
If the application is using the new ART runtime it will otherwise
load dalvik and start a dalvik VM next to the ART VM.
Does not work very well obviously.
c400eef377 introduced some defines to handle
older kernel headers. However, the check is done before the corresponding
kernel header (dvb/frontend.h) is included. As a result the macros are
always defined with results in 'redefined' errors with newer kernel
headers.
Move the check after the include to fix this.
https://bugzilla.gnome.org/show_bug.cgi?id=730570
We need to sleep a bit before destroying the player object
because of a bug in Android in versions < 4.2.
OpenSLES is using AudioTrack for rendering the sound. AudioTrack
has a thread that pulls raw audio from the buffer queue and then
passes it forward to AudioFlinger (AudioTrack::processAudioBuffer()).
This thread is calling various callbacks on events, e.g. when
an underrun happens or to request data. OpenSLES sets this callback
on AudioTrack (audioTrack_callBack_pullFromBuffQueue() from
android_AudioPlayer.cpp). Among other things this is taking a lock
on the player interface.
Now if we destroy the player interface object, it will first of all
take the player interface lock (IObject_Destroy()). Then it destroys
the audio player instance (android_audioPlayer_destroy()) which then
calls stop() on the AudioTrack and deletes it. Now the destructor of
AudioTrack will wait until the rendering thread (AudioTrack::processAudioBuffer())
has finished.
If all this happens with bad timing it can happen that the rendering
thread is currently e.g. handling underrun but did not lock the player
interface object yet. Then destroying happens and takes the lock and waits
for the thread to finish. Then the thread tries to take the lock and waits
forever.
We wait a bit before destroying the player object to make sure that
the rendering thread finished whatever it was doing, and then stops
(note: we called gst_opensles_ringbuffer_stop() before this already).
