This is the same code that is in decklinkaudiosrc, audioringbuffer,
audiomixer and various other places. Have it once instead of copying it
everywhere.
https://bugzilla.gnome.org/show_bug.cgi?id=787560
Performing a gst_sdp_media_get_caps_from_media() would result in
changing fields in the GstSDPMedia violating the const tag in the
function declaration.
Before there would be a line with a=rtpmap:96 VP8/90000
after, that attribute would only contain a=rtpmap:96
Fix by performing modifications on duplicated strings instead of on
the internal values.
Also add a simple test for checking that the representation doesn't
change by a gst_sdp_media_get_caps_from_media()
In gst_video_time_code_is_valid, also check for invalid
ranges when using drop-frame TC. Refactor some code which
broke after the check was added.
https://bugzilla.gnome.org/show_bug.cgi?id=779010
To make the structs usable in bindings, and fix
gstrtspmessage.c:1188: Warning: GstRtsp:
gst_rtsp_message_parse_auth_credentials: return value: Invalid
non-constant return of bare structure or union; register as
boxed type or (skip)
https://bugzilla.gnome.org/show_bug.cgi?id=774416
Also the format must be fixed on the default raw caps. If not
gst_video_info_from_caps() will fail and
gst_video_decoder_negotiate_default_caps() return FALSE.
The test simulates the use case where a gap event is received before
the first buffer causing the decoder to fall back to the default caps.
https://bugzilla.gnome.org/show_bug.cgi?id=773103
Add a test that check that we handle time ranges (a range of time that maps to
the same sample).
Also update the other tests to use our check api to compare int64 values to get
better output on failure.
Split large tests into small tests and name them specifically. Use helpers to
avoid repetition. Make sure the order in the file is the same as we add the to
the suite.
Comparing floats for equality is not necessarily going to
work reliably, so use fail_unless_equals_float() for this.
Test would fail on x86 (Intel Atom x5-Z8300).
Elements inherited from GstAudioDecoder, supporting PLC and introducing
delay produce invalid timestamps. Good example is opusdec with in-band FEC
enabled. After receiving GAP event it delays the audio concealment until
the next buffer arrives. The next buffer will have DISCONT flag set which
will make GstAudioDecoder to reset it's internal state, thus forgetting
the timestamp of GAP event. As a result the concealed audio will have the
timestamp of the next buffer (with DISCONT flag) but not the timestamp
from the event.
The serialization of double typed geographical
coordinates to DMS system supported by the exif
standards was previously truncated without need.
The previous code truncated the seconds part of
the coordinate to a fraction with denominator
equal to 1 causing a bug on the deserialization
when the test for the coordinate to be serialized
was more precise.
This patch applies a 10E6 multiplier to the numerator
equal to the denominator of the rational number.
Eg. Latitude = 89.5688643 Serialization
DMS Old code = 89/1 deg, 34/1 min, 7/1 sec
DMS New code = 89/1 deg, 34/1 min, 79114800UL/10000000UL
Deserialization
DMS Old code = 89.5686111111
DMS New code = 89.5688643
The new test tries to serialize a higher precision
coordinate.
The types of the coordinates are also guint32 instead
of gint like previously. guint32 is the type of the
fraction components in the exif.
https://bugzilla.gnome.org/show_bug.cgi?id=767537
This tag match the EXIF_TAG_FOCAL_LENGTH_IN_35_MM_FILM exif tag and is
stored on a short. Hence there is a precision loss compared to the
GstTag which is a double value.
https://bugzilla.gnome.org/show_bug.cgi?id=753930
Doing so prevents us dropping buffers in the rare, but possible, situations,
when the stream changes SSRC and new sequence numbers does not differ
much from the last sequence number from previous SSRC. For example:
ssrc - 0xaaaa 101,102,103,104 ssrc - 0xbbbb 102, 103, 104, 105...
In the scenario above we don't want to drop the first 3 packets of
0xbbbb stream.
https://bugzilla.gnome.org/show_bug.cgi?id=764459
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
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
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
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
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
[API] gst_discoverer_info_to_variant
[API] gst_discoverer_info_from_variant
[API] GstDiscovererSerializeFlags
+ Serializes as a GVariant
+ Adds a test
+ Does not serialize potential GstToc (s)
https://bugzilla.gnome.org/show_bug.cgi?id=748814
When generating segment, we can't assume the first buffer is actually
the first expected one. If it's not, we need to adjust the segment to
start a bit before.
Additionally, we if don't know when the stream is suppose to have
started (no clock-base in caps), it means we need to keep everything in
running time and only rely on jitterbuffer to synchronize.
https://bugzilla.gnome.org/show_bug.cgi?id=635701
Don't feed 64-bit integer variable into vararg function that expects
an unsigned integer to go with GST_TAG_TRACK_NUMBER. This would
cause crashes on 32-bit platforms, and if not that then test
failures if the comparisons fail later (at least on big endian
platforms).
Allows subclasses to do custom caps query replies.
Also exposes the standard caps query handler so subclasses can just
extend on top of it instead of reimplementing the caps query proxying.
https://bugzilla.gnome.org/show_bug.cgi?id=741263
Refactor the encoder's caps query proxying function to a common place
and use it in the videodecoder to proxy downstream restrictions.
The new function is private to the gstvideo lib.
https://bugzilla.gnome.org/show_bug.cgi?id=741263
The set_format vfunc does not pass ownership of the caps
to the decoder, so we mustn't unref the caps there.
gst_event_new_caps() does not take ownership of the caps
passed, so we must unref the caps afterwards.
Fixes leaks when running test in valgrind in 1.4 branch.
Add test to check rendering of overlays of different sizes
that are completely or partially outside the video surface.
Once the overlay is blended to the video, verify if the
position of the blended overlay is as expected, by comparing
the pixels of the blended video with the expected values.
https://bugzilla.gnome.org/show_bug.cgi?id=739281
Make an ORC version of the 2x vertical upsampling code.
Improve unit tests, test chroma up and down sampling.
memset buffer in conversion to make valgrind happy.
Combine multiplies in 4x filters.
Rename conversion functions to make them nicer in orc.
Add ORC versions for various downsampling algorithms
Add unit test chroma resampler
Make a more complete pack/unpack test, check if the image after
pack/unpack has the same color and precision, and has correctly
duplicated subsampled pixels.
Add a video scaler object build on top of the resampler. It has
implementation to deal with interlaced video as well as horizontal and
vertical scaling functions.
Make the MIKEY message and payload objects miniobjects so that they have
a GType and are refcounted.
We can reuse the dispose method to clear our payload objects.
Add some annotations.
Implement a copy function for the MIKEY message.
Fix the unit test.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732589
With most decoder libraries, and especially when accessing codecs via
OpenMAX or similar APIs, we don't have the ability to properly related
the output buffers to a number of input samples. And could e.g. get
a fractional number of input buffers decoded at a time.
Previously this would in the end lead to an error message and stopped
playback. Change it to a warning message instead and try to handle it
gracefully. In theory the subclass can now get timestamp tracking
wrong if it completely misuses the API, but if on average it behaves
correct (and gst-omx and others do) it will continue to work properly.
Also add a test for the new behaviour.
We don't change it in the encoder yet as that requires more internal logic
changes AFAIU and I'm not aware of a case where this was a problem so far.
The KEMAC payload actually needs to have subpayloads and the key should
go into the KEY_DATA subpayload. Add support for subpayloads and
implement the KEY_DATA payload.
Add some pointers to the conversion functions that allow us to add
encryption and decryption later.
MIKEY is defined in RFC 3830 and is used to exchange SRTP encryption
parameters between a sender and a receiver in a secure way.
This library implements a subset of the features, enough to implement
RFC 4567, using MIKEY in SDP and RTSP.
Check that even if the subclass doesn't call set_output_format, the base
class should use upstream provided caps to fill the output caps that is
pushed before the gap event is forwarded, otherwise it ends again fixating
the rate and channels to 1.
https://bugzilla.gnome.org/show_bug.cgi?id=722144
Adds a test that simulates a scenario where the first buffers after
a segment can't be decoded and the decoder asks for those frames
to be released. The videodecoder base class should make sure that
the events attached to those first buffers are pushed even if the
buffers aren't going to be.
https://bugzilla.gnome.org/show_bug.cgi?id=721835
Add a simple playback test that makes sure that video decoder pushes
buffers in the same order it receives and that it respects the
set timestamps and durations
We already have internally the information on what type of stream (audio,
video, container, subtitle, ...) a certain caps is.
Instead of forcing callers to specify which CODEC_TAG category a certain
caps is, use that information to make a smart choice.
Does not break previous behaviour of gst_pb_utils_add_codec_description_to_tag_list
(if tag is specified it will be used, if caps is invalid it will be rejected,
...).
https://bugzilla.gnome.org/show_bug.cgi?id=702215
The function gst_rtp_buffer_get_payload can not be used in Python
because it lacks necessary length parameter. This patch adds a new
function, gst_rtp_buffer_get_payload_bytes, to use from Python
bindings. The new function has the advisory "Rename to:" annotation
so it can replace the gst_rtp_buffer_get_payload whan creating
bindings.
The function gst_rtp_buffer_get_extension_bytes is also added. It wraps
gst_rtp_buffer_get_extension_data which doesn't work in Python due to
incomplete annotation and because it returns the length as number of
32-bit words.
https://bugzilla.gnome.org/show_bug.cgi?id=698562
Allocate header, payload and padding in separate memory blocks in
gst_rtp_buffer_allocate().
don't use part of the payload data as storage for the extension data but store
it in a separate memory block that can be enlarged when needed.
Rework the one and two-byte header extension to make it reserve space for the
extra extension first.
Fix RTP unit test. Don't map the complete buffer or make assumptions on the
memory layout of the underlaying implementation. We can now always add extension
data because we have a separate memory block for it.
Add support RTP buffers with multiple memory blocks. We allow one block for the
header, one for the extension data, N for data and one memory block for the
padding.
Remove the validate function, we validate now when we map because we need to
parse things in order to map multiple memory blocks.
