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
For the MS/VfW codec ids, we want to write DTS timestamps instead
of PTS because that's what everyone else seems to do (and it's also
how it is in AVI). So for those input formats we use the buffer DTS
instead of the PTS. However, if there's no DTS set but only the PTS
then just take the PTS instead of dropping the input buffer. This
is useful especially for I-frame only codecs like JPEG and huffyuv,
but should also be fine as fallback in general.
Fixes regression with input JPEG frames that only have PTS set on them.
https://bugzilla.gnome.org/show_bug.cgi?id=756967
Instead, delay it until all request pads have been released. This is
because the release_pad() vfunc requires the multiqueue and muxer to
be there in order to release their request pads as well. If those
elements are destroyed earlier, release_pad() does not work, no
pads are released and some resources are leaked.
https://bugzilla.gnome.org/show_bug.cgi?id=753622
We have to reverse all samples in a buffer before processing them to properly
have continuous data from one buffer to another. As a result we will have a
negative applied rate and a rate of 1.0.
Also make sure that input buffers are correctly clipped to the segment,
otherwise our calculations are going to go wrong.
Also copy over the segment event's sequence number to the output segment while
we're at it.
https://bugzilla.gnome.org/show_bug.cgi?id=757033
Implement accept-caps handler to avoid doing a full caps query
downstream to handle it.
This commit implements accept-caps as a simplification of the _getcaps
function, so it exposes the same limitations that getcaps would.
For example, not accepting renegotiation to caps with capsfeatures when
it was last configured to a caps that it has to deinterlace.
If the QtDemuxStream are re-used they may already have caps which used
to be leaked.
Reproduced using the
validate.dash.playback.seek_forward.dash_exMPD_BIP_TC1 validate
scenario.
https://bugzilla.gnome.org/show_bug.cgi?id=756561
Negotiation to audio/x-raw,format=S8 was not possible because S8 does
not have a bit order so we ended up doing `if (!entry.fourcc) goto refuse_caps;`
https://bugzilla.gnome.org/show_bug.cgi?id=756387
They now use the new GstAudioVisualizer base class
from gst-plugins-base/gst-libs/gst/pbutils
Also fixed undefined reference to gst_audio_visualizer_get_type
Added GST_PLUGINS_BASE_LIBS to Makefile.am and re-order LIBADD.
https://bugzilla.gnome.org/show_bug.cgi?id=742875
Add statitics from each rtp source to the rtp session property.
'source-stats' is a GValueArray where each element is a GstStructure of
stats for one rtp source.
The availability of new stats is signaled via g_object_notify.
https://bugzilla.gnome.org/show_bug.cgi?id=752669
Buffer is added to the internal cache, and pushed only when accumulated
buffer duration crosses 200 ms. So when the chain ends, the buffer accumulated
is not freed. Freeing the cache when the state changes from PAUSED to READY.
https://bugzilla.gnome.org/show_bug.cgi?id=754212
By not doing this, the muxer is not effectively a rtpmuxer, rather a
funnel, since it should be a single stream that exists the muxer.
If not specified, take the first ssrc seen on a sinkpad, allowing upstream
to decide ssrc in "passthrough" with only one sinkpad.
Also, let downstream ssrc overrule internal configured one
We hence has the following order for determining the ssrc used by
rtpmux:
0. Suggestion from GstRTPCollision event
1. Downstream caps
2. ssrc-Property
3. (First) upstream caps containing ssrc
4. Randomly generated
https://bugzilla.gnome.org/show_bug.cgi?id=752694
If seeking targets an empty segment skip it as there is no media
offset to get from it. Instead look for the next one.
This doesn't make seeking in push-mode work if you seek to an
empty segment but at least won't get you to wrong offsets.
https://bugzilla.gnome.org/show_bug.cgi?id=753484
mux_start_time refers to the running_time of the buffer
that goes first in the output file. Normally this time is
0, so this variable is initialized to 0 during the state
change to PAUSED.
However, when dealing with dynamic pipelines and starting
a recording while the pipeline has already run for a while,
the running_time of the first buffer is > 0 and this causes
a problem with detecting the end of the first file(s) when
splitting by duration, because the code will later compare
the threshold_time with (last buffer running_time - mux_start_time)
and will get it wrong until mux_start_time advances enough
to make this difference < threshold_time, creating empty files
in the meantime.
https://bugzilla.gnome.org/show_bug.cgi?id=753624
During reverse playback, the media should stop playing at segment.start
This does not happen, and avidemux continues to process data even when
current timestamp is less that segment.start.
https://bugzilla.gnome.org/show_bug.cgi?id=755094
Avoid using default accept-caps handler that will query downstream
and is more expensive. Just check if the caps is compatible with
the template and check if the channels are the same.
Caps from the pad template are being leaked. In any case it is
from a static pad template and will 'leak' in the end, just doing
the cleanup for the good practice.
On reading LOAS config, flag v=1 and vA=1 combination can occur, leading to warning
"Spec says "TBD"...". Returning TRUE on this case while parameters 'sample_rate' and
'channels' are pointing to uninitialized values can end on setting random values as
rate and channels on src caps.
https://bugzilla.gnome.org/show_bug.cgi?id=755611
Otherwise we end up considering the values did not change and we wrongly
work with the old video format (which will lead to wrong
behaviour/segfaults).
https://bugzilla.gnome.org/show_bug.cgi?id=755621
eceb2ccc73 broke segment seeks by always
accumulating segments manually when activating a segment. This is only
needed when handling edit lists, not when activating a segment because of a
seek. Do the accumulation when switching edit list segments instead.
This fixes segment seeks again, while keeping edit lists playback working.
https://bugzilla.gnome.org/show_bug.cgi?id=755471
* use g_list_free_full(), don't iterate elements maually when freeing
* call gst_rtp_*_pay_clear_packet(), don't duplicate its code
* use gst_buffer_unref() to clarify that it is buffers being released,
instead of refering directly to gst_mini_object_unref()
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277
It's normal when dropping into the middle of a stream to
not always have the config available immediately, so skip LOAS
until a valid config is seen without either setting invalid
caps or erroring out.
https://bugzilla.gnome.org/show_bug.cgi?id=751386
flac contains the sample offset in the frame header, so after a seek
without index flacparse will know the exact position we landed on and
timestamp buffers accordingly. It only set the pts though, which means
the baseparse-set dts which was set to the seek position prevails, and
since the seek was based on an estimate, there's likely a discrepancy
between where we wanted to land and where we did land, so from here on
that dts/pts difference will be maintained, with dts possibly multiple
seconds ahead of pts, which is just wrong. The easiest way to fix this
is to just set both pts and dts based on the sample offset, but perhaps
parsed audio should just not have dts set at all.
https://bugzilla.gnome.org/show_bug.cgi?id=752106
One-line removal of tags_written++
This should fix rtmp output to crtmpserver, and hopefully
noone is expecting that the element count includes the end
element, as different bits of documentation say different
things about whether it should or not.
https://bugzilla.gnome.org/show_bug.cgi?id=661624
Apparently the Microsoft Azure RTMP server requires that the
videodatarate and audiodatarate metadata be provided, so
set those, even if it's to 0. Use the actual input bitrate
tags if available.
In parse_keymgmt(), don't mutate the input string that's been passed
as const, especially since we might need the original value again if
the same key info applies to multiple streams (RTX, for example).
When a resource is 404, and we have auth info - retry with the auth
info the same as if we had receive unauthorised, in case the resource
isn't even visible until credentials are supplied.
Fix a memory leak handling Mikey data.
When generating a random keystring, don't overrun the 30 byte
buffer by generating 32 bytes into it.
In gst_smpte_collected(), check upfront if input formats are same
or not. This avoids allocation of in1 and in2 buffers and
subsequent memory leak when input formats do not match.
https://bugzilla.gnome.org/show_bug.cgi?id=754153
When we haven't started yet, set the start_index when we set the index property,
so that we start at the right index position after the initial seek. The index
property was never really meant to be for writing, but it used to work, so let's
support it for backwards compatibility.
https://bugzilla.gnome.org/show_bug.cgi?id=739472
Commit 7d7e54ce68 added support for
DASH common encryption, however commit
bb336840c0 that went onto master
shortly before the CENC commit caused the calculation of the CENC
aux info offset to be incorrect.
The base_offset was being added if present, but if the base_offset
is relative to the start of the moof, the offset was being added twice.
The correct approach is to calculate the offset from the start of the
moof and use that offset when parsing the CENC aux info.
Sometimes it is useful to know this information on the
server side. Other popular implementations (vlc, ffmpeg, ...)
also send this header on every message.
This includes a new "user-agent" property that the user
can set to use a custom User-Agent string. The default
is "GStreamer/<version>"
https://bugzilla.gnome.org/show_bug.cgi?id=750101
Use constantDuration to calculate the timestamp of non-first AU in the
RTP packet.
If constantDuration is not present in the MIME parameters, its value
must be calculated based on the timing information from two consecutive
RTP packets with AU-Index equal to 0.
https://bugzilla.gnome.org/show_bug.cgi?id=747881
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 video tag.
https://bugzilla.gnome.org/show_bug.cgi?id=751774
This commit adds support for ISOBMFF Common Encryption (cenc), as
defined in ISO/IEC 23001-7. It uses a GstProtection event to
pass the contents of PSSH boxes to downstream decryptor elements
and attached GstProtectionMeta to each sample.
https://bugzilla.gnome.org/show_bug.cgi?id=705991
GstRTSPMedia uses this classification to detect the real payloader
inside a dynpay bin and asserts if it doesn't find it, therefore
it is required
https://bugzilla.gnome.org/show_bug.cgi?id=753325
Initialize the PT to the default value of the codec and check if
it is still the default before declaring the pt to be dynamic or
not when setting the caps.
Also use the PT constants from the rtp lib when possible
https://bugzilla.gnome.org/show_bug.cgi?id=747965
