It allows encoders to detect and drop input frames which are already
late to increase the chance of the pipeline to catch up.
The QoS logic and code is directly copied from gstvideodecoder.c.
https://bugzilla.gnome.org/show_bug.cgi?id=582166
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
