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
Detect when the eagl surface changed its dimension (when the user rotates
the device for example) and adapt the egl internals to draw to that,
preventing that ios resizes the image again when drawing.
This is particularly harmful when eagl would scale down a image
to draw and the ios screen would scale it back up because the
surface is now bigger than when the element was configured.
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
It's not developed any more and replaced by the
libschroedinger-based elements in gst-plugins-good.
(The libschroedinger 1.0.9 release notes state "This
is an exciting release: most of the encoding tools in
dirac-research have been ported over to Schrödinger, so
now schro has the same or better compression efficiency
as dirac-research.")
TRM IDs are MusicBrainz' old audio fingerprinting system from
Relatable, they were phased out in favour of MusicIPs PUIDs.
https://wiki.musicbrainz.org/History:TRM
In some scenarios, for example in QtWebKit, might be difficult to obtain full
control on the egl display and it might be only accessible indirectly via
eglGetCurrentDisplay().
https://bugzilla.gnome.org/show_bug.cgi?id=700058
We only want to adjust the timestamps so that they start from 0 for live
streams. Non-live streams already start from 0 and after a seek we actually want
to timestamp to be the position we seek to.
Non-live streams should timestamp buffers with a running-time starting from
0. Since we already push a 0 -> -1 segment, bring the timestamps to 0
by subtracting the initial timestamp.
The xmlCleanupParser function seems to cleanup all statically
allocated libxml variables, making it unusable. We can't guarantee
that dashdemux won't need it anymore, so better not call it.
Manifest updates should be done periodically for live streams,
this patch makes the demuxer create a new manifest client for
the new version and transfers the stream position to the new
one, discarding the old one afterwards.
A small struct that keeps a short history of fragment download bitrates
to have an average measure of N last fragments instead of using only
the last downloaded bitrate
Do not use a global bitrate as the sizes of the fragments matter
when calculating the download rate as the connection setup time is
also being taken into the download duration, a smaller fragment
will have a lower bitrate than a larger one.
This avoids switching the bitrates for streams frequently because
of bitrate mismatches
Instead of downloading 1 fragment per stream per download loop,
select the stream with the earlier timestamp and get a fragment
only for that one.
The old algorithm would lead to problems when the fragment durations
were too different for streams.
dashdemux shouldn't emit the buffering message as that can pause
the pipeline. It has no proper knowledge of the downstream buffering
status so it can pause the pipeline when it isn't necessary. It should
have an internal buffer for downloading the streams ahead of playback,
but that shouldn't make it able to stop the pipeline for buffering.
A particular case in which this is bad is when a pad switch happens
(changing bitrates for example), the new pads dashdemux creates
will get linked to demuxers and new queues will be created,
these queues are initially empty and dashdemux will quickly
drain its buffers by pushing them to those queues. So it
would have no more buffers internally and would emit a
buffering message with a low ratio, causing the pipeline
to pause when it wouldn't be necessary.
Put EOS on the streams queues after the last fragment from the
last period for each stream. This way we keep it serialized
with the buffers and it will work when streams have different
ending times
The smallest queue should be used to prevent blocking the download
thread when a stream has too much data buffered, leaving the other
streams starving from fragments
Each stream has its own durations and timestamps, the fragment number
is different for each stream when seeking, so the seek has to be done
for all streams, rather than on a single stream and propagated to
others
GstDataQueue has proper locking and provides functions to limit the
size of the queue. Also has blocking calls that are useful to
our multithread scenario in Dash.
Store the buffers separately for each stream, this is clearer than
having a queue with a list of buffers. It also allows easier selection
of buffers to push in later refactors
Fragments should be pushed ASAP as downstream should be responsible for
doing the syncrhonization and proper buffering.
This has the great side effect of fixing most of the seeking A/V sync issues.
- the MPD file is updated in the download loop (only if we have a "dynamic" MPD and minimumUpdatePeriod is valid);
- properly LOCK/UNLOCK the GstMpdClient;
This fixes conflicts with the HLS plugin, which is also named
fragmented.
When building its registry, gstreamer was picking one or the other
between hls and dashdemux.
This fixes build that has been broken by commit
fb9aeac6552021b176a4c4bd07265e02a0b70e0f.
gst_mpd_client_get_target_duration has been removed, and
gst_mpd_client_get_next_fragment_duration should be used instead.
This was necessary to support variable-duration Fragments.
in the new API:
- gst_mpd_client_get_current_position returns the timestamp of the NEXT fragment to download;
- gst_mpd_client_get_next_fragment_duration returns the duration of the next fragment to download;
- gst_mpd_client_get_media_presentation_duration returns the mediaPresentationDuration from the MPD file;
also there is a new internal parser function:
- gst_mpd_client_get_segment_duration extracts the constant segment duration from the MPD file
(only used when there is no SegmentTimeline syntax element in the current representation)
In gst_mpd_client_get_next_fragment, we set the timestamp/duration of the fragment just downloaded
copying the values from the corresponding GstMediaSegment.
TODO: rework SEEKING to support seeking across different Periods.
- Periods are played in sequence, from PeriodStart to PeriodEnd
- seamless switching from one Period to the next one works fine;
- the 'new-segment' generation is broken, so if we need to switch pads for a new Period there is a crash;
- build a list of the available Periods with their start and duration time
- add the list of GstStreamPeriod in the GstMpdClient data struct
- remove cur_period from GstMpdClient and introduce an API to get the current GstStreamPeriod
- several API clean-ups
build the list of segments to be played using the SegmentTimeline syntax, if present
bugfixes:
- for dynamic MPD files, when mediaPresentationDuration is not present use minimumUpdatePeriod instead
- do not add a spurious '$' when building an URL from a template like "$Bandwidth$/init.mp4v"
- introduce gst_mpd_client_add_media_segment() to avoid code duplication
other fixes:
- fixed a buffering bug: now we stop buffering when we reach the end of manifest
- now gst_mpd_client_get_target_duration() always returns a valid duration
(in case of single-segment streams, we return either Period duration or mediaPresentation duration)
TODO: support SegmentTimeline
SegmentList nodes are allowed into Period, AdaptationSet or Representation nodes
and there is at most 1 element, so no need to keep a list;
Period nodes cannot have any Represention elements, as AdaptationSet nodes are mandatory;
this breaks compatibility with some legacy DASH test sequences.
gstmpdparser.c: In function ‘gst_mpdparser_get_list_and_nb_of_audio_language’:
gstmpdparser.c:2891: warning: ‘return’ with no value, in function returning non-void
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.
