WAV is too generic to impose more-or-less arbitrary boundaries on the
sample rate and channel count caps. For example, there are 384 kHz WAV
files. Another example: it is in theory possible that somebody puts DSD
data into a WAV file, which will then have a sample rate of ~2.8 MHz.
For this reason, get rid of the rate and channel caps unless they are
fixed values. Downstream anyway usually knows the limitations better.
https://bugzilla.gnome.org/show_bug.cgi?id=761514
gst_rtp_buffer_add_extension_onebyte_header() and
gst_rtp_buffer_add_extension_twobytes_header() can have a const argument for
the actual extension data.
When processing EOS for a pad, send a stream-group-done
for the pad in case downstream is waiting for more
data on this stream before it can process related
streams from the group.
https://bugzilla.gnome.org/show_bug.cgi?id=768995
My collection leak fix 83f30627cd
introduced a crash in this scenario: audiotestsrc ! decodebin3 ! fakesink
The reference handling of collection in decodebin3 wasn't very clear and
my attempt to fix the leak introduced a regression where we went one
reference short in some other scenarios.
Fixing this by:
- Giving a strong reference to DecodebinInput making things clearer
- Fixing get_merged_collection() which was sometimes returning an
existing reference and sometimes a new one.
https://bugzilla.gnome.org/show_bug.cgi?id=769080
This reverts commit a16cd5d2a5.
Setting the stop time on the segment breaks reconfiguration, as the
encoder signals an EOS, but we reconfigure it an continue to produce
buffers.
This information should not be required via the segment downstream
since we already have the sample count being used to generate buffer
durations.
https://bugzilla.gnome.org/show_bug.cgi?id=768763
If the duration is not known from the chain, it might be known
by the startup seek.
This fixes failure to seek.
Merged with a patch from Tim-Philipp Müller <tim@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=768991
After we reset the resampler, there is no history anymore in the resampler
and the previously calculated output size is no longer valid.
Recalculate the new output size after a reset to make sure we don't try
to convert too much.
The _pull_sample() and _pull_preroll() functions block
until a sample is available, EOS happens or the pipeline
is shut down (returning NULL in the last two cases).
This adds _try_pull_sample() and _try_pull_preroll()
functions with a timeout argument to specify the maximum
amount of time to wait for a new sample.
To avoid code duplication, wait forever if the timeout is
GST_CLOCK_TIME_NONE and use that to implement
_pull_sample/_pull_preroll with the original behavior.
Add also corresponding action signals "try-pull-sample"
and "try-pull-preroll".
https://bugzilla.gnome.org/show_bug.cgi?id=768852
If sanity checks on the buffer size allocated by XvShmCreateImage() fail,
call on g_set_error(), rather than just logging a warning, as this
failure is fatal.
Add a sanity check on buffer size when the video format is RGB. This adds to
existing checks on various YUV pixel formats.
https://bugzilla.gnome.org/show_bug.cgi?id=767712
The collection owned by GstDecodebin3 has to be unreffed when disposing.
gst_event_new_stream_collection() doesn't consume the collection passed
to it so no need to give it an extra ref.
https://bugzilla.gnome.org/show_bug.cgi?id=768811
MultiQueueSlot owns a ref on the active stream so it should release it
when being freed.
DecodebinInputStream owns ref on the active and pending stream so they
should be dropped when being freed.
https://bugzilla.gnome.org/show_bug.cgi?id=768811
gst_stream_get_caps() returns a reffed caps.
The caps passed to gst_query_set_caps_result() are not transfered.
The caps in gst_parse_pad_stream_start_event() was either acquired
using gst_pad_get_current_caps() which returns a new ref or
explicitly reffed.
https://bugzilla.gnome.org/show_bug.cgi?id=768811
Remove unnecessary helper struct for callbacks. The bclass
member of the helper struct was not used, so we can just
remove it and the GET_CLASS() call and simplify the whole
affair by passing the depayloader directly to the callback.
parsed, framed, stream-format and alignment are only relevant for parsers and
should not matter here. We still want to be able to use an encoder that can
only output byte-stream if the input was avc.
https://bugzilla.gnome.org/show_bug.cgi?id=768566
Caps are cleaned up for missing plugins, and for creating encoding profiles
and caps descriptions.
Fields like streamheader, parsed, framed, stream-format and alignment are not
relevant here. The last ones all because a parser will take care of them.
https://bugzilla.gnome.org/show_bug.cgi?id=768566
When a discont buffer is processed, the state is re-initialized, which
nullifies the allowed_tags.
The problem is when a subrip string with tags is processed and allowed_tags is
NULL. The function subrip_unescape_formatting() calls g_strjoinv with a
str_array as NULL, leading to a GLib-CRITICAL.
This patch removes the allowed_tags resetting, in parser_state_init(), but
move it into gst_sub_parse_format_autodetect().
https://bugzilla.gnome.org/show_bug.cgi?id=768525
We need to take into account the input segment flags to know whether
we should drain the decoder after a new keyframe in trick mode.
Otherwise we would have to wait for the next frame to be outputted (and
the segment to be activated) which ... well ... kind of beats the whole
point of this draining :)
