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.
Live streams force the demuxer to keep reloading the Manifest from
time to time, as the new fragments are being added as they are recorded.
The demuxer should also try to keep up and detect when it had to skip
fragments, marking the discont flag when that happens.
Curiously, the spec doesn't seem to mention when/how a live stream is supposed
to end, so keep trying downloads until the demuxer errors out.
Use pad tasks to download data and an extra task that gets the earlier
buffer (with the smallest timestamp) and pushes on the corresponding
pad.
This prevents that the audio stream rushes ahead on buffers as its
fragments should be smaller
When connection-speed changes, signal that we might need a bitrate
switch. During the switch, a new pad group is added and the old one
is drained and removed.
New pads also need to push newsegments before starting to stream
This speed limits the maximum bitrate of streams. Currently it
is only read when starting the pipeline, but it should be used
for switching bitrates during playback to adapt to network
changes.
mssdemux should set the streams it has exposed as active so that
the manifest won't use the non-active streams to compute total bitrates
or providing fragments
When the stream can't have its caps detected, better not to expose it.
If no streams are known, signal an error about no playable streams to
the application
Adds basic handling for seek in time events. Needs to cooperate
with the downstream qtdemux so that it forwards the seeks and
the corresponding newsegments
This is important for downstream to properly timestamp the samples
The default value is 10000000, but this can be set in the stream
or at the top-level manifest entry
Keep a ref on pad to prevent it being unreffed while the mssdemux
streams are still using it. Also reset the element when going to
ready instead of when going to null.
Use shorter names for the MSS manifest helper structure and functions.
Also continues the implementation of the stream fetching and pushing loop.
Now it uses the base url correctly and already fetches and pushes the fragments
downstream
