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
The RTP PT for alaw is 8.
Less than 50 packets are received in the length of this test so it
would never drop a buffer or would drop only the last buffer and
it would fail sometimes when the received wouldn't receive the
retransmission packet in time.
https://bugzilla.gnome.org/show_bug.cgi?id=746445
We need a proper caps event from upstream with the full RTP caps as we can't
create caps ourselves from thin air. Fixes usage of rtpstreamdepay after e.g.
a filesrc or any other element that supports pull mode.
https://bugzilla.gnome.org/show_bug.cgi?id=753066
h264parse does the same, let's keep the behaviour consistent. As we now
include the codec_data inside the stream too here, this causes less caps
renegotiation.
The spec says:
When a picture parameter set NAL unit with a particular value of
pic_parameter_set_id is received, its content replaces the content of the
previous picture parameter set NAL unit, in decoding order, with the same
value of pic_parameter_set_id (when a previous picture parameter set NAL unit
with the same value of pic_parameter_set_id was present in the bitstream).
If the GOP is completed, pads have to start gathering for the
next one but it is possible that the the state might go to
COLLECTING_GOP_START and back to WAITING_GOP_COMPLETE before the
thread has a chance to wake up and proceed, leaving it trapped in
the check_completed_gop loop and deadlocking the other threads
waiting for it to advance.
To solve it, this patch also checks that tha input running time
hasn't changed to prevent this scenario.
h264parse does the same and this fixes decoding of some streams with 32 SPS
(or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255), but
the field in the codec_data for the number of SPS or PPS is only 5 (or 8) bit.
As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
This looks like a mistake in the part of the spec about the codec_data.
1) If the system http_proxy environment variable is not set
or set to an empty string, we must not set proxy to avoid
http connection error.
2) In case of proxy property setting, if user want to clear
the proxy setting, they should be able to set it to NULL or
an empty string again, so this is fixed too.
3) Check if the proxy string was parsed correctly.
https://bugzilla.gnome.org/show_bug.cgi?id=752866
In media to caps function, reserved_keys array is being used for variable i,
leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
changed it to variable j
https://bugzilla.gnome.org/show_bug.cgi?id=753009
Skip keys from the fmtp, which we already use ourselves for the
caps. Some software is adding random things like clock-rate into
the fmtp, and we would otherwise here set a string-typed clock-rate
in the caps... and thus fail to create valid RTP caps
https://bugzilla.gnome.org/show_bug.cgi?id=753009
Don't hold the main splitmux part lock over
the parent state change function, as it prevents
posting error messages that happen. Since the purpose
is to prevent typefinding from proceeding, use a
separate mutex just for that.
Some of the subtitle chunks will have embedded
NUL-terminators (last three), some don't (first three),
some will have markup, some won't, some will be valid
UTF-8 (all but last), some won't (last stanza).
https://bugzilla.gnome.org/show_bug.cgi?id=752421
Need to check that the number of bytes we want to copy from the adapter
actually is available and handle the error case gracefully. This error
may happen if malformed packets are received and we don't have a
complete frame.
https://bugzilla.gnome.org/show_bug.cgi?id=752663
The subtitle buffer we push out should not include a NUL terminator
as part of the data, we just add such a terminator for safety, but
it should not be included in the buffer size.
A NUL terminator is not valid UTF-8, so checks will fail if it's
included in the size, and the NUL will be replaced by the fallback
character specified when converting, i.e. '*'.
https://bugzilla.gnome.org/show_bug.cgi?id=752421
In certain applications, splitting into files named after a base
location template and an incremental sequence number is not enough.
This signal gives more fine-grained control to the application to
decide how to name the files.
https://bugzilla.gnome.org/show_bug.cgi?id=750106
For 1ch or 2ch devices, we just need to set the caps to allow both
options since CoreAudio will up/downmix appropriately.
Also fixes the condition for the 2ch case to be exact, rather than at
least 2 channels since the downmix will not take place in the >stereo
case.
We need to initialize the AudioUnit early to be able to probe the
underlying device, but according to the AudioUnitInitialize() and
AudioUnitUninitialize() documentation, format changes should be done
while the AudioUnit is uninitialized. So we explicitly uninitialize the
AudioUnit during a format change and reinitialize it when we're done.
_audio_unit_property_listener is called either from a Core Audio thread
or as a result of a Core Audio API (e.g. AudioUnitInitialize)
from our own thread. In the latter case, osxbuf can be already locked
(GStreamer's mutex is not recursive).
We introduce the flag cached_caps_valid and use it instead of nullifying
cached_caps when we cannot lock on osxbuf.
https://bugzilla.gnome.org/show_bug.cgi?id=743758
- Probing caps is unified between source and sink
- Hardware stream format is now reported as preferred capabilities
(dynamically updated when hardware configuration changes)
- Get hardware channel layout from Remote IO just like from HAL
- More comprehensive mapping between AudioChannelLabel and
GstAudioChannelPosition
- Support for unpositioned channel layouts
- Announce stereo-mono upmixing/downmixing in caps
https://bugzilla.gnome.org/show_bug.cgi?id=743758
For more optimised RTP packet handling: means we don't
need to map the input buffer again but can just re-use
the mapping the base class has already done.
https://bugzilla.gnome.org/show_bug.cgi?id=750235
For more optimised RTP packet handling: means we don't
need to map the input buffer again but can just re-use
the map the base class has already done.
https://bugzilla.gnome.org/show_bug.cgi?id=750235
Estimating it from the RTP time will give us the PTS, so in cases of PTS!=DTS
we would produce wrong DTS. As now the estimated DTS is based on the clock,
don't store it in the jitterbuffer items as it would otherwise be used in the
skew calculations and would influence the results. We only really need the DTS
for timer calculations.
https://bugzilla.gnome.org/show_bug.cgi?id=749536
