Quite a few (broken?) files have a packet duration of 1ms, which is
most definitely wrong for either audio or video packets.
We therefore avoid using that value and instead use other metrics to
determine the buffer duration (like using the extended stream properties
average frame duration if present and valid).
The new delay is three times as much as the old one, and just happens
to let me properly decode my "Princess Mononoke" DVD, on which dvdreadsrc
was chocking before at 5:11.
While there, merge the constants used in two places into a define.
https://bugzilla.gnome.org/show_bug.cgi?id=539708
Only error out when downstream returns:
* NOT_SUPPORTED
* ERROR
* NOT_NEGOTIATED
* NOT_LINKED
If we got _UNEXPECTED, we push an EOS downstream (since maybe only one
of the streams had gone EOS) and then stop the task silently.
In the case of WRONG_STATE we just need to stop silently
https://bugzilla.gnome.org/show_bug.cgi?id=600412
When on push mode and receiving an EOS event, asfdemux
should push all pending data because we might be dealing
with a broken file that has a preroll value higher
than its actual length.
Some (broken) streams don't have the extended stream properties in
the header, resulting in applying a duration of zero on outgoing
buffers.
Fixes#611473
Some files have payload with timestamps smaller than the preroll duration.
Instead of blindly substracting the preroll value (and ending up with
insanely high timestamps on the outgoing buffers), we make sure we
never go below 0.
Fixes#610432
Turns out 4 + 4 + 2 + (4 * 2) is actually 18 and not 22. This avoids
a presumably unintentional padding of uninitialised bytes at the end
of the CONT tags chunk, which should be harmless but causes warnings
in valgrind (see #608533 for a test URL).
We previously only aggregated flow returns after the while(push) loop,
which meant that in some cases we would end-up not properly aggregating
the flow returns.
This is based on the same flow aggregation algorithm as oggdemux.
The correct basis for (Xing, VBRI) seek table calculations is the
byte size and duration provided by that metadata, rather than some
other (possibly even estimated) one. This also prevents an infinite
conversion loop in (unlikely) case where a TOC is provided without
such corresponding (duration) metdata.
Use the same strategy as accurate seeks to store
pending non-accurate seeks to avoid overwriting non-definite
stop times. When doing non-accurate seeks our position
reporting might drift off by some secs and the stream can
end up before it should.
Fixes#603695
Adds chained asfs handling to pull mode. It now checks if
there is a new asf header after the last packet (when it
is possible to know how many packets are) or it tries
checking if a processed packet that fails is an header
object.
Fixes#599718
Adds support for detecting and playing chained asfs
in push mode. asfdemux tries to detect a new asf start
by identifying the header object guid in a input buffer.
When it finds it, it resets its state, removing its pads
and creates new ones for the new file.
When receiving bogus data, we have to avoid subtracting a value
larger than 'size' from 'size' variable, resulting in a wrap
that would make 'size' a really large bogus value.
Fixes#599333
Fixes#598761
This removes a good 50% of processing time for parsing a buffer.
We do this by simply... getting the nicks that we already have handy
instead of going through the expensive glib system.
Previously, we could get false sync relatively easily - it sometimes happened
on real files. This cleans the code up a fair bit, and makes it require more
confirmation that we've found valid sync before continuing.
Metadata provided seek tables are consistent with metadata's view of
total size, which typically matches real size, but need not do so
(e.g. a truncated file). Fixes seeking and position reporting
in such truncated files (although duration based on metadata may then
still be incorrect).
asf packets in rtp packets should come with their padding fields
set to 0 and the depayload must update them to the correct
value before pushing downstream
This also fixes a bug by which the first buffer (in a multi-packet mode)
passed to asf_demux_parse_packet() would have a GST_BUFFER_SIZE of the
full incoming buffer and not just of the single asf packet.
Fixes corrupted frames introduced by latest commit.
We now have a chance for packets to be collected before we send out the
newsegment. If we're not in accurate seeking (keyunit) it will set
the segment start/time to the keyframe's timestamp.
We now *always* seek to the keyframe just before our requested position.
When we encounter the first keyframe and we were not accurate (therefore doing
keyframe seeking), we update the segment start position to the keyframe timestamp.
This will still cause some timestamp jitter, but giving a hint as to the duration
rather than nothing seems to be a better idea.
Also, this allows some scenarios (like remuxing with asfmux) to estimate the total
duration using the accumulated packet duration (which will be correct).
The simple index entries also contain the number of packets one needs
to retrieve at a given position to get a full keyframe. We therefore
use that information to retrieve all those packets in one buffer when
working in pull-mode.
In gst_asf_demux_chain_headers, when 'goto wrong_type' was called
asfdemux tried to free a const pointer that had been cast to a
normal pointer variable.
We weren't taking the preroll into account previously, meaning that we
were always seeking preroll nanoseconds too early... resulting in a lot
of dropped packets (which are before the start time).
This brings quit a bit closer to as-fast-as-possible seeking in asf files.
Post global tags only after we've added our source pads, so that
tag events get sent downstream in addition to tag messages posted
on the bus. This makes sure tags can be picked up automatically
when transcoding, but also by tagreadbin/playbin2. Fixes#519721.
While we're at it, also add a container-format tag.
When we receive a DISCONT as input, don't clear our complete state but simply
mark a discont that will be put on the next buffer. The code will be able to
handle and throw away incomplete data.
Add some more debug info.
Remove an unused variable.
Let's not put every single mp3 frame in our index, a few frames per
second should be more than enough. For now use an index interval
of 100ms-500ms depending on the upstream size, to keep the index at
a reasonable size. Factor out the code that adds the index entry
into a separate function for better code readability.
While technically upstream may be seekable even if it doesn't know
the exact size, I can't think of a use case where this distincation
is relevant in practice, so for now just assume we're not seekable
if upstream doesn't provide us with a size. Makes sure we don't
build a seek index when streaming internet radio with sources that
pretend to be seekable until you try to actually seek.
Don't overwrite the origin flow return by whatever flow we get
when trying to push the remaining internally queued payloads.
We want to do our eos logic, ie. send an EOS event or segment-done
message in any case. Makes things EOS properly when an EOS event
is forced upon the pipeline so that the source returns
FLOW_UNEXPECTED to a pulling asfdemux. Should fix#582056.
Add a multichannel map to the output caps, and send at least a CODEC and
BITRATE tag. I'm not too sure about the 5.1 and 7.1 channel maps. I have
no samples and can't find info about the channel ordering, but this is
better than nothing.
Some Xing headers apparently start the TOC at byte 1 instead of 0. Don't
reject them because of it, just subtract the initial offset when reading
the table.
Be more lenient about what we accept as changing bits in a header - basically,
only require that the mp3 sync marker is present, for the mpeg version,
layer and samplerate.
Fixes: #581464
Some mp3 streams have an offset in timestamps, requiring us to push the
frame *AFTER* segment.stop in order for the decoder to be able to push
all data up to the segment.stop position.
This also makes timestamps (more) consistent before and after a possible
seek, and moreover makes for reasonable position reporting in live stream
(whose payload timestamps should not be taken for granted).
* Improve newsegment handling, e.g. upstream might live in TIME.
* Only send newsegment if we have needed info.
* Avoid reading past end of data section.
The problem that happens is the following:
* A packet with multiple payloads comes in
* Those payloads get handled one by one
* The first payload contains the first audio payload with timestamp A
* The second payload contains the first video (key)frame with timestamp V (where V < A)
With the previous code, the following would happen:
* the first payload gets processed, then passed to queue_for_stream
* queue_for_stream detects it's the first valid timestamp received and stores
first_ts = A
* the second payload gets processed, then pass to queue_for_stream
* queue_for_stream detects the timestamp is lower than first_ts... and
discards it... resulting in losing the first keyframe of the video stream
We've been having this issue for *ages*... it's just that nobody noticed it
that much with playbin. But with playbin2's aggresive multiqueue handling, this
will result in multiqueue not being able to preroll (because the video decoder will
be dropping a ton of buffers before (maybe) receiving the next keyframe).
Tested with over 200 asf files, and they all play the first frame correctly now,
even the most braindead ones.
This might be caused by entering the if() line 1214 and then not having
any activated_streams.. resulting in reaching line 1267 without having
any valid flow value.
Fixes playback of Windows Media RTSP streams and other non-Real RTSP
streams where the server errors out because it can't handle the
Real-specific 'Required: com.real.retain-entity-for-setup' header
we've been adding unconditionally in the recent past.
For reference:
rtsp://66.111.34.191:601/broadcast/alnour.rm
rtsp://195.134.224.231/snowboard_100.wmv
On win32, we're required to link to all the libraries used - including
ones only indirectly used by other libs. So, add gstaudio, gsttag, and
(for windows only) winsock.
Parse the ETag from the describe method and pass the sessionid as the value for
the If-Match header is subsequent setup calls.
Fixes support for more RealMedia RTSP streams.
Don't introduce glitches in the output by a) relaxing the threshold for
taking upstream timestamps in preference to our calculated timestamps and
b) only set the discont flag on outgoing buffers in response to an incoming
discont buffer.
Fixes: #575046
Don't allow a change in sample rate/channels/layer/version unless we can
see another frame at the correct offset. Prevents accidently flipping
due to simple single-bit corruption.
Since SEEK event handling might perform some conversion
from TIME to BYTES, do not let upstream fool application
into (TIME) seeking not being possible.
Integer underflow made accurate seeks to near zero fail and seek to
completely the wrong place. Fix by clamping to zero, since we can't seek
to negative times anyway.
Add a new utri handler for pnm:// that for now just redirects to the same uri
with the rtsp:// protocol, which usually works nowadays.
Separate the registration of the various plugins into a separate source file.
Drop packets with an invalid replicated data length
instead of continuing with an invalid timestamp
and uninitialized payload metadata.
All other code assumes that the timestamps are valid.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c: (channel_mode_class),
(GST_TYPE_MP3_CHANNEL_MODE), (mp3_type_frame_length_from_header),
(gst_mp3parse_emit_frame), (mp3parse_get_query_types):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Do an initial class_ref on an internal enum type from within the
class_init function so that there aren't any issues when multiple
mp3parse elements are started in separate threads at the same
time. (Why we use an enum type here if the tag is registered as
a string type, I don't know). Also remove custom UNUSED macro
and use GLib's instead.
