When a stream gets a not-linked return, it will be marked as so and
won't download any more new fragments until a reconfigure event
is received. This will make mssdemux expose all pads, but only download
fragments for the streams that are actually being used.
Relying on the pads being linked/unlinked isn't enough in this scenario
as there might be an input-selector downstream that is actually discarding
buffers for a given linked pad.
When streams are switching, the old active stream can be blocked because
input-selector will block not-linked streams. In case the mssdemux's
stream loop is blocked pushing a buffer to a full queue downstream it will
never unblock as the queue will not drain (input-selector is blocking).
In this scenario, stream switching will deadlock as input-selector is
waiting for the newly active stream data and the stream_loop that would
push this data is blocked waiting for input-selector.
To solve this issue, whenever an stream is reactivated on a reconfigure
it will enter into the 'catch up mode', in this mode it can push buffers
from its download thread until it reaches the currrent GstSegment's position.
This works because this timestamp will always be behind or equal to the maximum
timestamp pushed for all streams, after pushing data for this timestamp,
the stream will go back to default and be pushed sequentially from the main
streaming thread. By this time, the input-selector should have already
released the thread.
https://bugzilla.gnome.org/show_bug.cgi?id=711849
This prevents locking on startup when a stream only has a single buffer
for one of the streams and mssdemux decides to push an EOS event right
after it.
WmaPro is actually wmaversion 3, and can also be found by the
WMAP fourcc.
Some manifests also contain the block_align field as "PacketSize"
in the audio track description, the libav decoders require it
to be present in caps.
Fixes#699921
wma v2 expects block_align, channels and rate fields set to its caps.
This isn't present direclty on the manifests, so mssdemux should parse
it from the waveformatex structure
https://bugzilla.gnome.org/show_bug.cgi?id=699924
bitrate info is always present on the QualityLevel xml node as part
of the adaptive selection processing, put it into caps as some
decoders require it (avdec_wmav2 for example)
https://bugzilla.gnome.org/show_bug.cgi?id=699924
g_ascii_strtoull() returns a long long integer, but we need to
pass a normal int to gst_structure_set() for fields of G_TYPE_INT,
so cast appropriately.
The buffer parameter wasn't being used, it was only to signal if
a buffer was downloaded and advance to the next fragment in the
manifest.
Replace the buffer with a boolean that has the same effect and is
safer
connection setup times seem to matter when measuring the download
rate of different streams. Streams with longer fragments have a
*relatively* lower connection setup time and achieve higher bitrates.
Using the average seems unfair here, so use each stream's measured bitrate
to select its best quality option.
We need to cancel the downloader for each stream before joining the main download task, otherwise
the download task will block until all the stream tasks finish.
When the codec is AAC-LC, some server implementation (e.g. Microsoft IIS) doesn't add the CodecPrivateData attribute. The element needs to re-create the codec data from the Quality Level attributes (channels and sampling rate).
There is no way to know if a live stream is really finished, so try to reload the manifest and check if there are more fragments to download. Else just let know it's the EOS.
