We have logics in bandwidth calculation to ignore segments that is
smaller than half of target duration. The logic does not have any
effect right now as the target duration in mpd/hls params is always
zero.
This change will set target duration in mpd/hls params, thus it can fix
part of issue #581 as the last segment which is less than half of
target duration.
Issue #581.
Fixes#498.
Change-Id: Ieb2dbf4da9fc72a7b9de802cda4294f1954d29b4
Note that TTML in ISO-BMFF is not supported yet.
Also updated packager_test.py:
- Added a test using TTML passthrough.
- Computed output extension from input extension unless output_format
is specified.
Fixes#478.
Change-Id: Ia917fc4ed3c326782791ed67601fba02ea28b11d
Note that STYLE and REGION are not supported in mp4 container due to
spec limitation as 14496-30:2014 does not specify a way to signal
styles/regions inside mp4.
Closes#344.
Change-Id: I05c14df916f7b2c7ca4364ee9407e0eda4dc7a3f
- EditLists in input files are parsed and applied to sample timestamps.
- An EditList will be inserted in the ISO-BMFF output if
- There is an offset between the initial presentation timestamp (pts)
and decoding timestamp (dts). Chrome, as of M67, still uses dts in
buffered range API [1], which creates various problems when buffered
range by pts does not align with buffered range by dts. There is
another bug in Chrome that applies EditList to pts only [2]. This
means that we can insert an EditList to align pts range and dts range.
- MediaSamples have negative timestamps (e.g. for Audio Priming).
You may notice the below change on some contents:
- Some media duration is reduced by one or two frames. This is because
EditList in the input file was ignored in the previous code, so video
streams start with a zero dts and a non-zero pts; the smaller of dts
and pts was used as the starting timestamp (related to the earlier
workaround for Chrome's dts bug), so the calculated duration was
actually a bit larger than the actual duration. Now with EditList
applied, the initial pts is reduced to zero, so the media duration is
also reduced to reflect the actual and correct media duration.
It may also result in negative timestamps in TS/HLS Packed Audio, which
will be addressed in a follow up CL.
Fixes#112.
Partially address b/110782437.
[1] https://crbug.com/718641, fixed but behind MseBufferByPts.
[2] https://crbug.com/354518. Chrome is planning to enable the fix for
[1] before addressing this bug, so we are safe.
Change-Id: I59317740ad3807ca66fa74b3a18fdf7f32c96aeb
Instead, caclulating average bandwidth by dividing the sum of the
sizes of every segment by the sum of the durations of every segment.
This aligns with the requirement in HLS spec:
https://tools.ietf.org/html/draft-pantos-http-live-streaming-23 4.1.
BandwidthEstimator is also simplified to handle all blocks only.
Fixes#361
Change-Id: I89e7d415a841f4d4048f199de8dae7ffa250467b
Having "wvtt" in the codec string (in the master playlist) causes
errors on some older Apple products. As including it is optional,
we are opted to omit it to ensure support for all Apple products.
Close#402
Change-Id: Ib1072bcc26a3ff66e3a6d3204789c0c8c678d4db
The previous text to mp4 webvtt pipeline was incomplete. It
did not insert ad cues and it could only insert a segment
after a sample ended.
Now the pipeline supports ad cue insert and segment insertion
mid text sample. This required the pipeline to use the text
chunker (to split samples and insert segments) and required
a major overhaul of the text to mp4 converter.
Before the converter came before the chunker. This meant that
the converter only expected to see stream info and text samples.
Moving the converter after the cue aligner and chunker means
that the convert had to be aware of segments and cues.
The general approach is the same, however the converter will
convert the samples per-segment as the chunker will introduce
duplicate samples if a sample spans across segments.
Closes#362Closes#382
Change-Id: I0f54a40524c36a602ad3804a0da26e80851c92fd
KongQun Yang
Aaron Vaage