This fixes a bug where in some files mehd.fragment_duration is one unit
less than the actual duration of the fragmented movie, as explained below:
mehd.fragment_duration is computed by scaling the end timestamp of
the last frame of the movie in (in nanoseconds) by the movie timescale.
In some situations, the end timestamp is innacurate due to lossy conversion to
fixed point required by GstBuffer upstream.
Take for instance a movie with 3 frames at exactly 3 fps.
$ gst-launch-1.0 -v videotestsrc num-buffers=3 \
! video/x-raw, framerate="(fraction)3/1" \
! x264enc \
! fakesink silent=false
dts: 999:59:59.333333334, pts: 1000:00:00.000000000, duration: 0:00:00.333333333
dts: 999:59:59.666666667, pts: 1000:00:00.666666666, duration: 0:00:00.333333334
dts: 1000:00:00.000000000, pts: 1000:00:00.333333333, duration: 0:00:00.333333333
The end timestamp is calculated by qtmux in this way:
end timestamp = last frame DTS + last frame DUR - first frame DTS =
= 1000:00:00.000000000 + 0:00:00.333333333 - 999:59:59.333333334 =
= 0:00:00.999999999
qtmux needs to round this timestamp to the declared movie timescale, which can
ameliorate this distortion, but it's important that round-neareast is used;
otherwise it would backfire badly.
Take for example a movie with a timescale of 30 units/s.
0.999999999 s * 30 units/s = 29.999999970 units
A round-floor (as it was done before this patch) would set fragment_duration to
29 units, amplifying the original distorsion from 1 nanosecond up to 33
milliseconds less than the correct value. The greatest distortion would occur
in the case where timescale = framerate, where an entire frame duration would
be subtracted.
Also, rounding is added to tkhd duration computation too, which
potentially has the same problem.
https://bugzilla.gnome.org/show_bug.cgi?id=793959
... before the old streams is not exposed yet for MSS stream.
In case of DASH, newly configured streams will be exposed
whenever demux got moov without delay.
Meanwhile, since there is no moov box in MSS stream,
the caps will act like moov. Then, there is delay for exposing new pads
until demux got the first moof.
So, following scenario is possible only for MSS but not for DASH,
STREAM-START -> CAPS -> (configure stream but NOT EXPOSED YET)
-> STREAM-START-> CAPS (configure stream again).
In above scenario, we can reuse old stream without any stream reconfigure.
https://bugzilla.gnome.org/show_bug.cgi?id=797239
Apart from the obvious drawbacks of hardcoding, the drawback here was
that, if we subtracted 2 frames (instead of 2.6) from the target running
time, we'd request the next keyframe a bit too far into the future,
which would make our files split at the wrong position.
https://bugzilla.gnome.org/show_bug.cgi?id=797293
Flv does not support various channels in AAC stream format, for example
flvdemux detect an audio channels of 2(stereo) when the AAC really is 1(mono).
https://bugzilla.gnome.org/show_bug.cgi?id=797275
Flv does support changing the stream type and stream properties
after the headers were started to be written, and for example H264
codec_data changes can be supported.
https://bugzilla.gnome.org/show_bug.cgi?id=797256
For drop-frame framerates, when the expected next max timecode wraps
around at the end of the day, we have to subtract the offset of the
daily jam, otherwise we end up with a duration that's a few frames too
long.
https://bugzilla.gnome.org/show_bug.cgi?id=797270
This commit:
1. Reads the WebM and Matroska ContentEncryption subelements.
2. Creates a GST_PROTECTION event for each ContentEncryption, which
will be sent before pushing the first source buffer.
The DRM system id field in this event is set to GST_PROTECTION_UNSPECIFIED_SYSTEM_ID,
because it isn't specified neither by Matroska nor by the WebM spec.
3. Reads the protection information of encrypted Block/SimpleBlock and
extracts the IV and the partitioning format (subsamples).
4. Creates the metadata protection for each encrypted Block/SimpleBlock,
with those informations: KeyID (extracted from ContentEncryption element),
IV and partitioning format.
5. Adds a new caps for WebM encrypted content named "application/x-webm-enc",
with the following new fields:
"encryption-algorithm": The encryption algorithm used.
values: "None", "DES", "3DES", "Twofish", "Blowfish", "AES".
"encoding-scope": The field that describes which Elements have been modified.
Values: "frame", "codec-data", "next-content".
"cipher-mode": The cipher mode used in the encryption.
Values: "None", "CTR".
https://bugzilla.gnome.org/show_bug.cgi?id=765275
Strip ADTS headers if we detect any, apparently some Sony cameras
send AAC with ADTS headers. We could also change the stream-format
in the output caps, but that would be unexpected to pipeline builders
and would not exactly be backwards compatible.
qtdemux_update_streams() is only ever called after checking
`qtdemux->streams_aware` is TRUE. There is no need to check for that
condition again.
`qtdemux->streams_aware` is only modified when the demuxer is
hard-resetted, which is mutually exclusive with demuxing, so it cannot
be modified during the call.
https://bugzilla.gnome.org/show_bug.cgi?id=797191
Currently matroskademux does not emit no-more-pads until the first
Cluster is parsed, even though the Tracks have already been parsed and
from that point on there can be no more tracks.
This is important in MSE because the browser needs to know when the MSE
initialization segment has been completely parsed so that it can expose
the tracks to the user. Some applications depend on this been done
before they feed frames to the demuxer.
As a consequence, historically WebKit has relied on hacks such as
listening to the `pad-added` event, which made impossible to support
multiple tracks in the same file. Let's fix that.
https://bugzilla.gnome.org/show_bug.cgi?id=797187
This patch allows matroskademux to parse a second Tracks element,
erroring out if the tracks are not compatible (different number, type or
codec) and emitting new caps and tag events should they have changed.
https://bugzilla.gnome.org/show_bug.cgi?id=793333
This splits gst_matroska_demux_add_stream() into:
* gst_matroska_demux_parse_stream(): will read the Matroska bytestream
and fill a GstMatroskaTrackContext.
* gst_matroska_demux_parse_tracks(): will check there are no repeated
tracks.
* gst_matroska_demux_add_stream(): creates and sets up the pad for the
track.
https://bugzilla.gnome.org/show_bug.cgi?id=793333
This is necessary for MSE, where a new MSE initialization segment may be
appended at any point. These MSE initialization segments consist of an
entire WebM file until the first Cluster element (not included). [1]
Note that track definitions are ignored on successive headers, they must
match, but this is not checked by matroskademux (look for
`(!demux->tracks_parsed)` in the code).
Source pads are not altered when the new headers are read.
This patch has been splitted from the original patch from eocanha in [2].
[1] https://www.w3.org/TR/mse-byte-stream-format-webm/
[2] https://bug334082.bugzilla-attachments.gnome.org/attachment.cgi?id=362212https://bugzilla.gnome.org/show_bug.cgi?id=793333
The behaviour of split-now is to output the current GOP after
starting a new file.
The newly-added split-after signal will output the current GOP
to the old file if possible once a new GOP is opened.
https://bugzilla.gnome.org/show_bug.cgi?id=796982
For 59.94 FPS, it's common to set 60000 as timescale. For that
timescale, if the audio is late by as little as 0:00:00.000016666
(definitely less than one audio sample), lateness gets rounded to 1.
Added a safeguard that allows lateness up to 1 sample with the specific
trak's timescale, to make sure that values less than e.g. one audio
sample won't break the prefill mode. What will happen in this case is
that the audio will get squeezed back to the video's timestamp, which in
practice means that the audio will be 0.000016666 seconds early (with
the patch).
https://bugzilla.gnome.org/show_bug.cgi?id=797133
Accept wavpack correction streams (.wvc) on sink pad, so
that wavpackparse can also be used to packetise correction
streams.
Fix parsing of subblock ID tags - the higher bits are
flags and are not part of the ID. This resulted in
correction blocks not being recognised properly and
the output not having the right (correction) caps.
Currently, whenever we generate a 128-bit UID, we store it in a list and
return 0 if we ever encounter a collision. This is so mathematically
improbable that it's not worth checking for, so we can save memory and
time by not tracking the UID. Even if a collision happened, a list of
only 10 UIDs would be unlikely to detect it.
This article has a good description of how improbable a collision is:
https://en.wikipedia.org/wiki/Universally_unique_identifier#Collisionshttps://bugzilla.gnome.org/show_bug.cgi?id=797086
