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).
Handle stride alignment through the use of the video meta API. The
code is based on the corevideobuffer implementation.
If the video meta API is not supported and the underlying buffer
contains padding, the core media buffer is copied to a system memory
buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=727885
Devices suitable for decklinksrc may not have any output, hence querying
the input returns NULL. Add support for all cases where
input/output/config may be missing.
https://bugzilla.gnome.org/show_bug.cgi?id=727306
usecount is unsigned, so too many "unuse" will wrap the counter
around and the >= 0 check will always be fine.
It would be much simpler to just make the counter signed, but
moving the checks where the decrements happen allow a mistake
to be detected earlier, and thus easier to debug.
Coverity 1139791
While the code that creates the object sets priv to some existing
pointer after new, this ensures any future new not doing this will
hit the various priv!=NULL asserts in the code.
Coverity 1139935
As per discussed in bug #725383, it doesn't make much sense to default
to FALSE in the "iradio-mode" property. Better, let's sent the header
by default and just ignore headers that are not understood, if so.
https://bugzilla.gnome.org/show_bug.cgi?id=725659
Some audio decoders (at least the MP3 decoder on MTK based devices) outputs
raw audio in batches of multiple audio frames. We need to handle that
properly, otherwise the base class will be kind of unhappy.
It's impossible to create another pipeline with d3dvideosink after disposing
the previous one due to some problem in d3dvideosink. The message is: "Unable
to register Direct3D hidden window class".
I've evaluated the problem and it's that UnregisterClass() in working thread is
called before DestroyWindow() and UnregisterClass() does nothing.
https://bugzilla.gnome.org/show_bug.cgi?id=722622
The original size of 256k was too small for anything where
one would want to use shm. If the buffer's size needs to be limit, it is
better to use buffer-time in most cases anyway.
-add delsys property
-add delivery system capability to the gstreamer adapter structure
-ready for add new delivery systems
Application must ask the adapter structure to know which delivery systems are avaible.
The property delsys must be set.
https://bugzilla.gnome.org/show_bug.cgi?id=709414
Add a new color format seen on my Galaxy S3
(OMX_SEC_COLOR_FormatNV12Tiled = 0x7fc00002) to the table,
but don't actually implement it - the decoder doesn't choose it.
Remove an assert that makes the plugin fail noisily and take the app down
if it sees a color format it doesn't recognise (just skip the codec instead)
Modify the debug output when plugin scanning to print color format info to
make this sort of thing easier in the future.
