Rework the main processing loop. We now create an audio processing
chain from small core functions. This is very similar to how the
video-converter core works and allows us to statically calculate an
optimal allocation strategy for all possible combinations of operations.
Make sure we support non-interleaved data everywhere.
Add functions to calculate in and out frames and latency.
Match the section 5.1.2 of the HTTP/1.0 spec by using relative URIs unless we
are using a proxy server. Also, send Host header for compatability with
HTTP/1.1 and some HTTP/1.0 servers.
https://bugzilla.gnome.org/show_bug.cgi?id=758922
gst_rtsp_connection_connect_with_response accepts a response pointer
which it fills with the response from setup_tunneling if the
connection is configured to be tunneled. The motivation for this is to
allow the caller to inspect the response header to determine if
additional authentication is required so that the connection can be
retried with the appropriate authentication headers.
The function prototype of gst_rtsp_connection_connect has been
preserved for compatability with existing code and wraps
gst_rtsp_connection_connect_with_response.
https://bugzilla.gnome.org/show_bug.cgi?id=749596
Depayloaders will look at rtpbuffer->buffer for the discont flag.
When we set the discont flag on a buffer in the rtp base depayloader
and we have to make the buffer writable, make sure the rtpbuffer
actually contains the newly-flagged buffer, not the original input
buffer. This was introduced with the addition of the process_rtp_packet
vfunc, but would only trigger if the input buffer wasn't flagged
already and was not writable already.
When we detect a discont and the input buffer isn't already flagged
as discont, handle_buffer() does a gst_buffer_make_writable() on the
input buffer in order to set the flag. This assumed it had ownership
of the input buffer though, which it didn't. This would still work
fine in most scenarios, but could lead to crashes or mini object
unref criticals in some cases when a discont is detected, e.g. when
using pcapparse in front of a depayloader. This problem was
introduced in bc14cdf529.
In the case where the stream doesn't have a framerate set and the frames
don't have a duration set, we still want to use the clipping path to
make sure we don't push buffers outside of the segment.
The problem was the previous iteration was setting a duration of 2s, which
meant that any buffer which was less than 2s before the segment start would
end up getting pushed.
Instead, use a saner 40ms (25fps single frame duration) to figure out whether
the frame could be within the segment or not
Some servers incorrectly parse header names with strict case-sensitivity. For
compatibility with these systems change X-Sessioncookie to x-sessioncookie.
https://bugzilla.gnome.org/show_bug.cgi?id=758921
Any latency query before this will not get the correct latency so a new
latency query should be triggered once the audio sink know its own latency.
Without this the initial latency query from the pipeline arrives too early
sometimes and the resulting latency is too short.
https://bugzilla.gnome.org/show_bug.cgi?id=758911
Some Opus files found on the wild have 0 in the version field of the
OpusHead header, instead of the correct value of 1. The files still
play, don't make this error fatal.
https://bugzilla.gnome.org/show_bug.cgi?id=758754
Commit ff6d1a2a25 changed sample's type from
gint to gsize (and renamed it to in_samples). gsize is an unsigned long,
which means it can never be a negative value and the check making sure that
in_samples is >= 0 is never going to be false. Removing it.
CID 1338689
Move the audio quantize code from audioconvert to the audio library.
work on making an audio converter helper function similar to the video
converter.
Fold fastrandom directly into the quantizer, add some ORC code to
optimize this later.
Rename _get_default_mask() to _get_fallback_mask() to make it more
clear that the function only provides a fallback if nothing else can be
done. Also clarify this in the documentation.
API: gst_audio_channel_get_fallback_mask()
Add a TRUNCATE_RANGE flag for unpack functions to fill the least
significate bits with 0 (as did the old code). Also add functions
that don't truncate. Use the TRUNC flag in audioconvert for
backwards compatibility for now.
No need to use G_GINT64_FORMAT for potentially negative values of
GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
Plus it creates more readable values in the logs.
https://bugzilla.gnome.org/show_bug.cgi?id=757480
If the flush-start is arrived during _eos_wait() in basesink,
the 'eos' flag is overwritten to TRUE after exiting the _eos_wait().
To resolve the overwritten issue,
the subclass doing the _eos_wait() call should return the right value.
If the eos flag is set to TRUE again, it will cause error(enter the eos flow)
of the following state changing from PAUSED to PLAYING in basesink.
https://bugzilla.gnome.org/show_bug.cgi?id=754980
Encrypted RTP buffers may contain encrypted padding, hence it's
necessary to have an option to relax the validation in order to
successfully map the buffer.
When the flag GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is set
gst_rtp_buffer_map() will map the buffer like if padding is not
present.
https://bugzilla.gnome.org/show_bug.cgi?id=752705
When g_option_context_parse fails, context and error variables are not getting free'd
which results in memory leaks. Free'ing the same.
And replacing g_error_free with g_clear_error, which checks if the error being passed
is not NULL and sets the variable to NULL on free'ing.
https://bugzilla.gnome.org/show_bug.cgi?id=753852
gir include search directories should respect PKG_CONFIG_PATH,
just like we do everywhere else. Makes g-i pick up the right
paths when using ./configure --with-pkg-config-path=
https://bugzilla.gnome.org/show_bug.cgi?id=755494
v210, UYVP and IYU1 are complex formats for which pixel stride does not really
have a meaning. If we copy width*pstride bytes per line, it's not going to do
the right thing. As a fallback, copy stride bytes per line. This might copy
uninitialized bytes at the end of each line, but at least copies the frame.
https://bugzilla.gnome.org/show_bug.cgi?id=755392
Otherwise the application might push new buffers into the queue while we're
flushing, potentially causing the GQueue data structure to become inconsistent
and causing crashes soon after.
https://bugzilla.gnome.org/show_bug.cgi?id=754597
The dependency setup does not seem to work for all systems,
causing the build to fail with:
gstrtpbaseaudiopayload.c:65:0:
fatal error: gst/audio/audio-enumtypes.h: No such file or directory
My setup:
gcc (Ubuntu 4.8.4-2ubuntu1~14.04) 4.8.4
autoconf (GNU Autoconf) 2.69
automake (GNU automake) 1.14.1
libtool (GNU libtool) 2.4.2
https://bugzilla.gnome.org/show_bug.cgi?id=754344
Before we just merged everything in pretty much random ways
ad-hoc instead of keeping state properly. In 0.10 that was
how it worked, but in 1.x the tag events sent should always
reflect the latest state and replace any previous tags.
So save the upstream (stream) tags, and save the tags set
by the decoder subclass with merge mode, and then update
the merged tags whenever either of those two changes.
This slightly changes the behaviour of gst_video_decoder_merge_tags()
in case it is called multiple times, since now any call replaces
the previously-set tags. However, it leads to much more predictable
outcomes, and also we are not aware of any subclass which sets this
multiple times and expects all the tags set to be merged.
If more complex tag merging scenarios are required, we'll have
to add a new vfunc for that or the subclass has to intercept
the upstream tags itself and send merged tags itself.
https://bugzilla.gnome.org/show_bug.cgi?id=679768
Before we just merged everything in pretty much random ways
ad-hoc instead of keeping state properly. In 0.10 that was
how it worked, but in 1.x the tag events sent should always
reflect the latest state and replace any previous tags.
So save the upstream (stream) tags, and save the tags set
by the decoder subclass with merge mode, and then update
the merged tags whenever either of those two changes.
This slightly changes the behaviour of gst_audio_decoder_merge_tags()
in case it is called multiple times, since now any call replaces
the previously-set tags. However, it leads to much more predictable
outcomes, and also we are not aware of any subclass which sets this
multiple times and expects all the tags set to be merged.
If more complex tag merging scenarios are required, we'll have
to add a new vfunc for that or the subclass has to intercept
the upstream tags itself and send merged tags itself.
https://bugzilla.gnome.org/show_bug.cgi?id=679768
Apparently I forgot how gobject works, there is no need to expose
it directly as one can call it from the parent_class pointer
This reverts commit 8a64592481.
Apparently I forgot how gobject works, there is no need to expose
it directly as one can call it from the parent_class pointer
This reverts commit ea9b6a7e3c.
Add gst_audio_decoder_set_use_default_pad_acceptcaps() to allow
subclasses to make videodecoder use the default pad acceptcaps
handling instead of resorting to the caps query that is, usually,
less efficient and unecessary
API: gst_audio_decoder_set_use_default_pad_acceptcaps
Add gst_video_decoder_set_use_default_pad_acceptcaps() to allow
subclasses to make videodecoder use the default pad acceptcaps
handling instead of resorting to the caps query that is, usually,
less efficient and unecessary
API: gst_video_decoder_set_use_default_pad_acceptcaps
Use the object lock to protect the internal segment when updating
against access from getting the stats property.
Fix a critical in gst-inspect or when retrieving the stats
before any segment has arrived by checking whether the
segment has been initted..
last_segment is only being updated in dequeue_buffer(),
which is only called from _pull_sample(). _pull_preroll()
simply re-uses an old or dummy segment while the actual
one sits and waits in the queue.
https://bugzilla.gnome.org/show_bug.cgi?id=751147
Subclasses can use it to select what queries they want to handle
and forward the rest to the default handling function.
API: gst_video_decoder_sink_query_default
https://bugzilla.gnome.org/show_bug.cgi?id=753623
Subclasses can use it to select what queries they want to handle
and forward the rest to the default handling function.
API: gst_audio_decoder_sink_query_default
https://bugzilla.gnome.org/show_bug.cgi?id=753623
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without
tags or with only the audio tag.
https://bugzilla.gnome.org/show_bug.cgi?id=751774
There was a confusion, six depayloaders where passing through the
timestamp while the base class was re-timestamping to running
time. This inconstancy has been unnoticed has in most use cases
the incoming segment is [0, inifnity] in which case timestamps are
the same as running time. With DTS/PTS shifting added (to avoid
negative values) and pcapparse sending a different segment this
started being an issue.
https://bugzilla.gnome.org/show_bug.cgi?id=753037
gstvideoencoder.h:228: Warning: GstVideo: "@transform_meta"
parameter unexpected at this location:
* @transform_meta: Optional. Transform the metadata on ...
Push all pending events before pushing the gap. This ensures the
segment is pushed before the gap so it can be properly translated
to the running time
Includes unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=753360
When there is no clock_base provided, the start position is
set to 0 instead of the original segment start value. This
would break synchronization if start was not 0.
https://bugzilla.gnome.org/show_bug.cgi?id=752228
Add logging categories for most video objects.
Remove some useless debug lines in video-info and videotestsrc.
Add a performance debug line in the video scaler.
GST_TYPE_RTSP_LOWER_TRANS used to be defined in there, not
including the generated file makes older gst-p-good fail to build,
so it constitues an API break.
Add process_rtp_packet() vfunc that works just like the
existing process() vfunc only that it takes the GstRTPBuffer
that the base class has already mapped (with MAP_READ),
which means that the subclass doesn't have to map it again,
which allows more performant processing of input buffers
for most RTP depayloaders.
https://bugzilla.gnome.org/show_bug.cgi?id=750235
This minor update should workaround a build system bug. While the
makefile has been updated to generate more enum type, there is nothing
that updates the header and would lead to the generated code to be
produced again. This minor doc fix should ensure no one get a build with
missing symbols.
E.g.
video-scaler.c: In function 'gst_video_scaler_horizontal':
video-scaler.c:1332:3: error: 'func' may be used uninitialized in this function [-Werror=maybe-uninitialized]
func (scale, src, dest, dest_offset, width, n_elems);
^
video-scaler.c: In function 'gst_video_scaler_vertical':
video-scaler.c:1373:3: error: 'func' may be used uninitialized in this function [-Werror=maybe-uninitialized]
func (scale, src_lines, dest, dest_offset, width, n_elems);
^
GCC's analyses seem to be correct, for the simple fact that if you pass
get_functions a known format, but no hscale or vscale, it'll return
True without having done anything.
Some callers check for the scale values to be not NULL, but then
hscale->resampler.max_taps could return 0.
A different approach to the one presented in this patch is to check
for those max_taps, too, before calling get_functions.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=752051
The padding (if any) is included in the length of the last packet, see
RFC 3550.
Section 6.4.1:
padding (P): 1 bit
If the padding bit is set, this individual RTCP packet contains
some additional padding octets at the end which are not part of
the control information but are included in the length field. The
last octet of the padding is a count of how many padding octets
should be ignored, including itself (it will be a multiple of
four).
Section A.2:
* The padding bit (P) should be zero for the first packet of a
compound RTCP packet because padding should only be applied, if it
is needed, to the last packet.
* The length fields of the individual RTCP packets must add up to
the overall length of the compound RTCP packet as received.
https://bugzilla.gnome.org/show_bug.cgi?id=751883
It's needed to check if pixel-aspect-ratio exists before fixating.
It does not exist if input caps is not set yet and allowed caps
does not contain pixel-aspect-ratio (e.g. when using GST_VIDEO_CAPS_MAKE)
https://bugzilla.gnome.org/show_bug.cgi?id=751932
POOL meta just means that this specific instance of the meta is related to a
pool, a copy should be made when reasonable and the flag should just not be
set in the copy.
The default implementation copies all metadata without tags, and metadata
with only the video tag. Same behaviour as in GstVideoFilter.
This currently does not work if the ::parse() vfunc is implemented as all
metas are getting lost inside GstAdapter.
https://bugzilla.gnome.org/show_bug.cgi?id=742385
CLAMP checks both if value is '< 0' and '> max'. Value will never be a negative
number since it is a division of an unsigned integer (i). Removing that check
and only checking if it is bigger than max and setting it appropriately.
CID #1308950
For alaw/mulaw we should also try to initialize the channel positions in the
ringbuffer's audio info. This allow pulsesink to directly use the channel
positions instead of using the default zero-initialized ones, which doesn't
work well.
https://bugzilla.gnome.org/show_bug.cgi?id=751144
Add a utility function that, given a video size and a
packed stereoscopic mode, attempts to guess if the video
is packed at half resolution per view or not, since
very few videos provide the information.
We need to scale groups of 4 bytes for YUY2 formats so round up to 4.
It's possible that there is no Y byte for the last pixel so make sure
we clamp correctly.
The API does not follow the type naming convention. Re-enable
only if one take the time to box and rename (see (rename-to SYMBOL)
annotation) all types.
When copying info from the reference input state, duplicate
all the fields of the video info. The sub-class will have the
chance to override them later.
Add flags and enums to support multiview signalling in
GstVideoInfo and GstVideoFrame, and the caps serialisation and
deserialisation.
videoencoder: Copy multiview settings from reference input state
Add gst_video_multiview_* support API and GstVideoMultiviewMeta meta
https://bugzilla.gnome.org/show_bug.cgi?id=611157
This new clock slaving method allows for installing a callback that is
invoked during playback. Inside this callback, a custom slaving
mechanism can be used (for example, a control loop adjusting a PLL or an
asynchronous resampler). Upon request, it can skew the playout pointer
just like the "skew" method. This is useful if the clocks drifted apart
too much, and a quick reset is necessary.
Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
https://bugzilla.gnome.org/show_bug.cgi?id=708362
Otherwise ssrc changes via rtpsession's (deprecated!) internal-ssrc property
are not possible anymore. rtpsession was now patched to only suggest an ssrc
if it makes sense to do so.
In 2.0 we should get rid of all the properties that are also negotiated via
caps, the code and behaviour is too confusing otherwise.
https://bugzilla.gnome.org/show_bug.cgi?id=749581
According to this section of the rfc.
https://tools.ietf.org/html/rfc5506#section-3.4.2
The validation should be updated to accept more types of RTCP
packages, with this mask change feedback packages will be also
accepted.
Change-Id: If5ead59e03c7c60bbe45a9b09f3ff680e7fa4868
Micro-optimisation: if the buffer consist of just one memory, we
know we have already mapped that memory to read the headers, so
no need to map it another time to get to the payload data, we
can just set up the payload data details right there and then
and avoid another map call in gst_rtp_buffer_get_payload().
Adds up when receiving RTP-payloaded raw video which can easily
be thousands of packets per frame.
Implement a chain_list function, which avoids lots of locking
compared to the default fallback implementation in GstPad.
We may also want to do some more sophisticated timestamp
tracking here at some point, but for now leave it up to the
jitterbuffer and/or subclasses (in case buffers in the
buffer list have no timestamp set on them, there may only
be a timestamp for the whole list on the first buffer).
This provides the exact same behaviour as the default
fallback implementation.
Summary:
So that the user can easily use the same encoding profile to render
with/without audio/video stream.
API:
gst_encoding_profile_is_disabled
gst_encoding_pofile_set_enabled
https://bugzilla.gnome.org/show_bug.cgi?id=749056
Instead of returning the first video meta found on a buffer, return the
one with the lowest id (which is usually the same thing, except on
multi-view buffers)
Use g_utf16_to_utf8() instead of the more generic g_convert(), so
that we can extract text in UTF-16 format even on embedded systems
with crippled iconv support.
This code path is exercised by the id3demux test_unsync_v23
check in gst-plugins-good.
https://bugzilla.gnome.org/show_bug.cgi?id=741144
