Make M3U8 and GstM3U8MediaFile refcounted. The contents
of it and GstM3U8MediaFile are pretty much immutable
already, but if we make it refcounted we can just
return a ref to the media file from _get_next_fragment()
instead of copying over all fields one-by-one, and then
copying them all into the adaptive stream structure fields again.
Move state from client into m3u8 structure. This will
be useful later when we'll have multiple media playlists
being streamed at the same time, as will be the case with
alternative renditions.
This has the downside that we need to copy over some
state when we switch between variant streams.
The GstM3U8Client structure is gone, and main/current
lists are not directly in hlsdemux. hlsdemux had as
many CLIENT_LOCK/UNLOCK as the m3u8 code anyway...
Don't clear decryption state immediately after
initialising it in the start_fragment. Don't clear
the state of all streams when we want to only clear
the current stream.
https://bugzilla.gnome.org//show_bug.cgi?id=768757
Add demuxer instance-wide decryption key cache. The current and
last key url are per-stream, so make a shared cache. Move the
decryption handling into the stream object, and use the shared
cache for the keys.
Prepare hlsdemux for more than one single stream. Currently hlsdemux
assumes there'll only ever be one stream and most of the stream-specific
state is actually in the hlsdemux structure. Add a stream subclass
instead and move some stream-specific members there instead.
When switching fragments we don't want to keep any data around from the last
one, and also forget about all data when doing flushing seeks or selecting new
bitrates.
https://bugzilla.gnome.org/show_bug.cgi?id=764684
Handling the ghostpad and its internal pad was causing more issues
than helping because of their coupled activation/deactivation
actions.
As we have to install custom chain,event and query functions it is
better to use a floating sink pad internally in the demuxer and just
use those pad functions to push through a standard pad in the demuxer
https://bugzilla.gnome.org/show_bug.cgi?id=757951
Properly handle snap flags during reverse seeking. In this case
the before/after are also reversed, so handle those as such.
For example: with a sequence of 1s fragments:
|-- 0 --|-- 1 --|-- 2 --|-- 3 --|
If you seek to 1.5s it is inside fragment 1. With reverse and
snap-before: should play from the end of fragment 1
snap-after: should play from the end of fragment 0
The URI attribute from the EXT-X-KEY tag and the URI attribute from the
EXT-X-I-FRAMES-ONLY tag are both quoted-string attibutes that have their
quotation marks removed during parsing. The CODECS attribute of the
EXT-X-STREAM-INF is also a quoted-string attribute, but this attribute
was not being un-quoted.
This commit changes the parser to always unquote all quoted-string
attributes and adjusts the unit tests to this new bevahiour for the
CODECS attribute.
An additional test is added to check that parsing of all of the fields
in the EXT-X-STREAM tag is correct, including those that contain comma
characters.
https://bugzilla.gnome.org/show_bug.cgi?id=758384
Clear error as soon as we determine that the download failed,
otherwise there are code paths where we might return without
clearing it ever, which would leak the GError then. Also, we
can pass a NULL GError pointer to _fetch_uri(), so just do that
instead of passing one that we're going to just free again
right away anyway.
Setting the seek flags to GST_SEEK_FLAG_SNAP_* will change the seek
target time to a segment boundary.
Based on original work by Ben Willers <bwillers@digisoft.tv>.
https://bugzilla.gnome.org/show_bug.cgi?id=759108
As HLS does not provide any way of knowing the server's clock, and we do
buffering of "live" streams, at some point we will fall behind the server in
many cases and would have to advance to a fragment that is not in the playlist
anymore.
Previously we would've just resynced to the next oldest fragment that is still
there, but this causes problems as from this point onwards we would always
fall off the playlist again all the time.
Instead we now resync and move to the 3rd newest fragment like we would do
when starting playback of a live stream.
https://bugzilla.gnome.org/show_bug.cgi?id=758987
If connection speed is set, playlist according
to connection speed is selected as current playlist.
Problem is that the current variant of main playlist still
points to previously set variant.
If previously set variant doesn't correspond to current
playlist, then it causes unnecessary change of playlist
to the same playlist after first fragment is downloaded,
because of not updated current variant.
To fix this, we need to make sure that current variant
of main playlist corresponds to the current playlist
https://bugzilla.gnome.org/show_bug.cgi?id=758946
Don't jump backward to 3 files from the end of the playlist
when switching variants - it just means we downloaded
fragments fast and caught up to the end of the playlist.
Disable that by treating a variant switch as a playlist
update, not a restart due to a seek or so.
If the stream is discont, we must provide a timestamp in any case. Elements
like tsdemux are not going to output anything if we give a NONE timestamp
after a discont.
Also marking a stream as discont if a playlist change was not successful would
lead to the above situation, but in that case we are not required at all to
mark the stream discont as we're still at the old playlist.
If a (master) playlist contains a variant list entry without a
URI then during parsing of the next variant list entry we are
a) leaking the entry we're currently parsing (new_list), and
b) free'ing the pointer to the previous list entry (list) without
updating the pointer.
Hence when then adding the URI for the latest parsed entry, incorrect
information is stored, as the information is used from 'list' which
is not valid memory anymore, also leading to crashes.
Fix this by correctly storing the new variant list entry pointer
as needed.
https://bugzilla.gnome.org/show_bug.cgi?id=756861
Nicer to read, two lines of code less, and also the callback
function should've been a GCompareFunc that returns a gint
and not a boolean (it did work correctly, was just confusing).
In order to ensure the sequence_position will always be consistently updated,
store the current file duration.
This way, when we advance, we can always increment the position based on what
was previously outputted.
https://bugzilla.gnome.org/show_bug.cgi?id=752132
Allows playlists that are missing the mediasequence information to
be correctly parsed. If the playlist was updated without reseting
the mediasequence it would constantly increase over subsequent updates,
leading to issues during playback.
For live streams, we want to make sure there's a certain distance
between the sequence to play and the last (earliest) fragment.
The problem is that it assumes there are at least 3 fragments in
the playlist, which might not always be the case (like in the case
of a server restarting and gradually adding fragments).
In order to avoid ending up with negative sequence numbers (which
will just loop forever), limit the new target sequence number to
the highest of:
* either the first sequence number of the playlist (fallback)
* or 3 fragments from the last one (standard behaviour)
Move the TAG defines directly into the code, not sure what
their purposes is, these are printf format strings so having
them directly as literals in the code where they're used
makes the code easier to follow.
Remove playlist_str GString variable from GstM3U8Playlist struct,
since it's only used temporarily in playlist_render(). Might just
as well keep it local then.
Some live streams (eg youtube) don't remove fragments in order to allow
seeking back in time (live + vod).
When gst_m3u8_client_has_next_fragment is called, we are getting wrong fragment
because current_file points in first file of the fragments list resulting in
watching the stream from the beginning again.
This patch sets current_file to nth fragment for live streams, then on
gst_m3u8_client_has_next_fragment will keep up with the live stream.
https://bugzilla.gnome.org/show_bug.cgi?id=753344
This reverts commit 4ca3a22b6b.
The connection-speed=0 is used as a special value in the property
of hlsdemux to mean 'automatic' selection, m3u8.c doesn't need
to know about that as it should be as simple as possible.
So this patch hides this automatic selection documented in hlsdemux
into m3u8 logic and I think the gets harder to understand the code.
It also makes the hlsdemux unit tests work again
https://bugzilla.gnome.org/show_bug.cgi?id=749328
In live situations, it is not uncommon for the current fragment to end
up out of the (updated) play range (lowest/highest sequence). But the next
fragment to play *is* present in the play range.
When advancing, if we can't find the current GstM3U8MediaFile, don't abort
straight away. Instead, look if a GstM3U8MediaFile with the next sequence value
is present, and if so switch to it.
https://bugzilla.gnome.org/show_bug.cgi?id=750028
It's better to just select some random variant playlist instead of stopping,
chances are that it's still continuing to work and we might just have to
select a different variant again later.
We should only refresh the currently selected variant playlist (if any,
otherwise the main playlist), not the main playlist. And only try to
refresh the main playlist if updating the variant playlist fails.
Some servers (Wowza) use the request of the main playlist to create a
"session", which is then part of the URI of the variant playlist and
also the fragments. Refreshing the main playlist would generate a new
session, and the server rate limits that usually. And after a few retries
the server just kicks us out.
Also as a side effect we now use the same downloader for all playlists, so
that we only have 2 instead of 3 connections to the server. And also
previously we just ignored the downloaded data from the main playlist that
the base class gave to us.
When the segment is very short it might be the case that the
typefinding fails and when finishing the segment hlsdemux would
consider the remaining data (pending_buffer) as an encryption
leftover.
This patch fixes it and makes sure an error is properly posted
if typefind failed by refactoring buffer handling to a function
and using it from the data_received and finish_fragment functions.
We also have to update the current_file GList pointer in the M3U playlist
client, otherwise we are just continuing playback from the current position
instead of seeking.
upstream might send buffer lists instead of buffers and hlssink's
probe won't get called and a new segment won't be created when needed.
This patch fixes it by adding a chain_list function to the sink pad
that will just pass through the whole bufferlist if no segment needs
to be requested at the moment or convert the list into buffers to
check the proper timestamp to request the next key-unit that will
start the segment.
https://bugzilla.gnome.org/show_bug.cgi?id=746906
Move the property from subclasses to adaptivedemux, it allows
selecing the percentage of the measured bitrate to be used when
selecting stream bitrates
Allow the playlist-length to accept '0' as a value, indicating
that no segment should be removed from the playlist. This allows
generating playlists to be used as VOD when complete.
Allows to set a bitrate directly instead of measuring it internally
based on the received chunks. The connection-speed was removed from
mssdemux and hlsdemux as it is now in the base class
Add more power to the chunk_received function (renamed to data_received)
and also to the fragment_finish function.
The data_received function must parse/decrypt the data if necessary and
also push it using the new push_buffer function that is exposed now. The
default implementation gets data from the stream adapter (all available)
and pushes it.
The fragment_finish function must also advance the fragment. The default
implementation only advances the fragment.
This allows the subsegment handling in dashdemux to continuously download
the same file from the server instead of stopping at every subsegment
boundary and starting a new request
Optimize loop by moving condition outside of it and reuse the
find_next_fragment function to check if there is next instead of
replicating the same loop
Duration queries can be done a few times per second and would cause
the segment list to be traversed for every one. Caching the duration
prevents that.
The duration values in playlists are approximate only, and for
playlist versions 2 and older they are only rounded integer values.
They cannot be used to timestamp buffers. This resulted in playback
gaps and skips because the actual duration of fragments is slightly
different. The solution is to only set the pts of the very first
buffer processed, not for each fragment.
hlsdemux assumes that seeking is not allowed for live streams,
however seek is possible if there are sufficient fragments in the
manifest. For example the BBC have live streams that contain 2 hours
of fragments.
The seek code for both live and on-demand is common code. The
difference between them is that an offset has to be calculated
for the timecode of the first fragment in the live playlist.
When hlsdemux starts to play a live stream, the possible seek range
is between 0 and A seconds. After some time has passed, the beginning of
the stream will no longer be available in the playlist and the seek
range is between B and C seconds.
Seek range:
start 0 ........... A
later B ........... C
This commit adds code to keep a note of the B and C values
and the highest sequence number it has seen. Every time it updates the
media playlist, it walks the list of fragments, seeing if there is a
fragment with sequence number > highest_seen_sequence. If so, the values
of B and C are updated. The value of B is used when timestamping
buffers.
It also makes sure the seek range is never closer than three fragments
from the end of the playlist - see 6.3.3. "Playing the Playlist file"
of the HLS draft.
https://bugzilla.gnome.org/show_bug.cgi?id=725435
For small amounts some data might be mistyped and it would cause
the pipeline to fail. For example if you have AAC inside mpegts,
for small amounts, the AAC samples would cause the typefinder to
think it is AAC and not mpegts.
https://bugzilla.gnome.org/show_bug.cgi?id=736061
