The samplerate field in the STSD atom is not right for some ALAC files
(usually when audio is 96kHz/24bits), so the audio caps must be
extracted from the codec data.
https://bugzilla.gnome.org/show_bug.cgi?id=700382
I took the opportunity to simplify that code a bit. We now use
gst_buffer_make_writable() to make the buffer writable and map twice the
same buffer, with first map being read/write, and second read only. This
get rid of the critical:
GStreamer-CRITICAL **: gst_structure_set_name: assertion `IS_MUTABLE
https://bugzilla.gnome.org/show_bug.cgi?id=700044
The exist one case where that we endup with original caps in ret, in which
case we are not guaratied to have writable caps. Simply ensure this is the
caps are writable before entering the loop.
https://bugzilla.gnome.org/show_bug.cgi?id=700044
This reverts commit 84ae670ab4.
Actually this is not how it is supposed to work. videomixer
creates a [0,-1] segment and then puts frames of the different
streams there based on their running times in their own segments.
For receiving video data via RTSP when the video is sent via
multicast there is no way to specify the udpsrc buffer-size.
On windows the native network buffer is not large and with video
i-frames being huge the buffer is to small and you get i-frame corruption,
it looks terrible, and there is no (easy) way to set the udpsrc buffer-size.
https://bugs.freedesktop.org/show_bug.cgi?id=52264
Whenever the demuxer has a new caps on a stream, it should set the
new_caps variable to true and a new caps event will be pushed before
the next buffer
qtdemux takes its buffers from a GstAdapter. Those buffers are created
from the larger buffer that it obtained from upstream and they carry
the same flags, including DISCONT if it is set. In these cases, all
buffers that qtdemux is going to push would be marked as DISCONT.
This scenario can make parsers/decoders flush on every buffer leading
to no decoding at all hapenning. This patch prevents this by unsetting
the flag if it shouldn't be set.
* Explicitly init variables for fragmented formats at init
* Do not use GstClockTime type if the variable isn't a timestamp
* Fix a style/readability issue at an if block
* Group 2 mss mode conditional blocks together to improve readability
Conflicts:
gst/isomp4/qtdemux.c
This can confuse downstream when they get a byte segment after receiving
the natural time segment from qtdemux that it sends when starting to
push buffers. This is specially the case with parsers that try to
convert the position from byte to time format and might miss the
correct position for playback to start.
Reset different variables on state changes to ready and when
handling a flush-stop. For handling flush stops we should check
if there is an upstream adaptive demuxer driving the pipeline as this
means that qtdemux will get a new moov atom. For 'standard' isomedia
streams this isn't true and qtdemux should keep the previous moov
information around.
Conflicts:
gst/isomp4/qtdemux.c
Whenever dashdemux switches bitrates it sends a new moov with the
new stream configuration. qtdemux should now handle this by splitting
the exposing and configuration of streams into separate functions. When
the stream is new it is configured and exposed, when it is a new bitrate
of an existing stream it is only reconfigured.
Conflicts:
gst/isomp4/qtdemux.c
smoothstreaming streams should be handled as a special kind of
fragmented isomedia. In MSS the fragments will not contain a
'moov' atom with the media descriptions, this has to be extracted
from the caps.
Additionally, there should be another demuxer upstream that is likely
going to be the one to answer/act on queries and events, so qtdemux has
to forward those upstream.
