Commit graph

18212 commits

Author SHA1 Message Date
Tim-Philipp Müller
84a9493a81 audiomixer: use new gst_aggregator_pad_drop_buffer() 2015-02-13 16:25:52 +00:00
Tim-Philipp Müller
54d288fc19 videoaggregator: use new gst_aggregator_pad_drop_buffer() 2015-02-13 16:25:52 +00:00
Tim-Philipp Müller
3c2ee8ece5 aggregator: use new gst_aggregator_pad_drop_buffer() 2015-02-13 16:25:45 +00:00
Tim-Philipp Müller
592c2c8105 aggregator: add gst_aggregator_pad_drop_buffer()
steal_buffer() + unref seems to be a wide-spread idiom
(which perhaps indicates that something is not quite
right with the way aggregator pad works currently).
2015-02-13 16:25:45 +00:00
Tim-Philipp Müller
55abf436a0 aggregator: only post latency message if anything changed
Perhaps we should check for element state as well and
only post it if in PLAYING state.
2015-02-13 16:25:14 +00:00
Tim-Philipp Müller
9c21327a8d tests: remove GST_DISABLE_PARSE guards from two tests that don't require it 2015-02-13 16:25:14 +00:00
Arun Raghavan
df43022899 opensles: Make debug category naming a bit more consistent 2015-02-13 15:05:07 +05:30
Alessandro Decina
6e325be516 applemedia: avfvideosrc: fix crash when non-GL memory is used 2015-02-13 19:12:54 +11:00
Mathieu Duponchelle
b6f2a962b5 adaptivedemux: Fix logic in fragment_download_finish.
This was preventing us from surfacing a meaningful error.

https://bugzilla.gnome.org/show_bug.cgi?id=743703
2015-02-12 23:07:36 +01:00
Mathieu Duponchelle
7ca6d9634a dashdemux: Fix handling of live streams with timeshift buffers.
By implementing get_live_seek_range.

As shown by :

gst-validate-1.0 playbin \
uri=http://dev-iplatforms.kw.bbc.co.uk/dash/news24-avc3/news24.php

This patch handles live seeking, by setting a live seek range
comprised between now - timeShiftBufferDepth and now.

The inteersting thing with this stream is that one can actually
ask fragments up to availabilityStartTime, but it seems quite clear
in the spec that content is only guaranteed to exist up to
timeShiftBufferDepth.

One can test live seeking this way :

gst-validate-1.0 playbin \
uri=http://dev-iplatforms.kw.bbc.co.uk/dash/news24-avc3/news24.php \
--set-scenario seek_back.scenario

with scenario being:

description, seek=true
seek, playback-time=position+5.0, start="position-600.0",
flags=accurate+flush

This example will play the stream, wait for five seconds, then seek back
to a position 10 minutes earlier.

https://bugzilla.gnome.org/show_bug.cgi?id=744362
2015-02-12 22:47:26 +01:00
Mathieu Duponchelle
6b864813cc adaptivedemux: Set first segment time to segment start.
Otherwise as long as a seek wasn't executed, the position was
reported incorrectly:

gst-validate-1.0 playbin \
uri=http://dev-iplatforms.kw.bbc.co.uk/dash/news24-avc3/news24.php

https://bugzilla.gnome.org/show_bug.cgi?id=744362
2015-02-12 22:47:26 +01:00
Edward Hervey
43464aa204 glimagesink: Fix debug statement 2015-02-12 17:34:10 +01:00
Edward Hervey
f09272f434 gl: Add/Update more debug statements
Where possible, use the _OBJECT variants in order to track better from
which object the debug statement is coming from

Define (and use) GST_CAT_DEFAULT where applicable

Use GST_PTR_FORMAT where applicable
2015-02-12 15:57:55 +01:00
Edward Hervey
929a6a75d2 gl: Make glcontext debug category used in more places
As a bonus it makes the egl context debug messages visible now :)
2015-02-12 15:57:55 +01:00
Edward Hervey
638f99a37a glimagesink: Sprinkle more debug 2015-02-12 15:57:55 +01:00
Thiago Santos
cb94de5d2f dashdemux: set proper stream type to streams
strncmp returns 0 when it compares equal and not a
boolean
2015-02-12 11:16:25 -03:00
Tim-Philipp Müller
5230b2354f audiomixer: calculate stream_time used to sync pad values correctly
Use pad (input) segment to calculate the stream time from the
input timestamp, not the aggregator (output) segment.
2015-02-12 11:41:10 +00:00
Sebastian Dröge
31f59cb344 avfvideosrc: Only use the duration of the sample if it is valid 2015-02-12 11:12:09 +02:00
Mathieu Duponchelle
c98348c141 adaptivedemux: track per-fragment bitrates.
And use the average to go up in resolution, and the last fragment
bitrate to go down.

This allows the demuxer to react rapidly to bitrate loss, and
be conservative for bitrate improvements.

+ Add a construct only property to define the number of fragments
  to consider when calculating the average moving bitrate.

https://bugzilla.gnome.org/show_bug.cgi?id=742979
2015-02-12 00:07:15 +01:00
Sebastian Dröge
767fefafb3 x265enc: Add support for 10bit 4:2:0 and 4:4:4 formats 2015-02-11 14:44:16 +01:00
Sebastian Dröge
037928dcf6 Improve and fix LATENCY query handling
This now follows the design docs everywhere, especially the maximum latency
handling.

https://bugzilla.gnome.org/show_bug.cgi?id=744106
2015-02-11 14:16:21 +01:00
Song Bing
629b350806 camerabin examples: memory leak in camerabin examples code
should unref caps after set to profile. profile will ref it.

https://bugzilla.gnome.org/show_bug.cgi?id=744219
2015-02-10 15:48:26 -03:00
Song Bing
2d0e6d96c6 camerabin examples: memory leak in camerabin examples code
should unref after set object. The object will be refed when set
property.

https://bugzilla.gnome.org/show_bug.cgi?id=744219
2015-02-10 15:48:26 -03:00
Sebastian Dröge
4ccd425772 openslesringbuffer: Only allocate at most half the number of internal buffers as external audioringbuffer ones
Otherwise we might end up reading too much from the audioringbuffer, which
would result in reading silence.
2015-02-10 16:18:34 +01:00
Sebastian Dröge
9ed4054a94 decklinkvideosrc: Apply the diff between element and pipeline "base" time in all cases
Even if both clocks have the same rate, we need to apply this diff. Only when
it's the same clock we don't, as it's our clock then.
2015-02-10 16:01:19 +01:00
Sebastian Dröge
f83ac624e3 decklinkvideosrc: Add the diff between the pipeline base time and when we start to play
Add the diff between the external time when we went to playing and
the external time when the pipeline went to playing. Otherwise we
will always start outputting from 0 instead of the current running
time.
2015-02-10 15:30:44 +01:00
Sebastian Dröge
d4575e759b decklinkvideosink: Actually include the change mentioned in the last commit 2015-02-10 14:53:55 +01:00
Sebastian Dröge
e8167dd118 decklinkvideo{src,sink}: Sample the internal clock immediately after starting the device
Otherwise we might stay at 0.0s for too long because we will take the first
timestamp we ever see as 0.0... which will be after we started the device.
2015-02-10 14:48:09 +01:00
Sebastian Dröge
a8033553f9 videoaggregator: When receiving timeout before caps, make sure to also advance our frame counter
Otherwise we will directly go EOS on the next non-timeout.
2015-02-10 10:57:38 +01:00
Sebastian Dröge
69a37365f1 aggregator: Pause srcpad task on flow errors
Otherwise we will call the task function over and over again until
upstream finally handled the flow return and shuts us down.
2015-02-10 10:57:38 +01:00
Matthew Waters
783245f1fe videoaggregator: fix buffer selection when duration=-1
If the src framerate and videoaggreator's output framerate were
different, then we were taking every single buffer that had duration=-1
as it came in regardless of the buffer's start time.  This caused the src
to possibly run at a different speed to the output frames.

https://bugzilla.gnome.org/show_bug.cgi?id=744096
2015-02-10 14:44:21 +11:00
Sebastian Dröge
b739cf8b7f decklink: Fix compiler warning with gcc
gstdecklink.cpp: In member function 'virtual HRESULT GStreamerDecklinkInputCallback::VideoInputFrameArrived(IDeckLinkVideoInputFrame*, IDeckLinkAudioInputPacket*)':
gstdecklink.cpp:498:22: error: comparison between signed and unsigned integer expressions [-Werror=sign-compare]
     if (capture_time > m_input->clock_start_time)
                      ^
gstdecklink.cpp:503:22: error: comparison between signed and unsigned integer expressions [-Werror=sign-compare]
     if (capture_time > m_input->clock_offset)
                      ^
2015-02-09 21:04:17 +01:00
Sebastian Dröge
f0e85023c2 decklink{audio,video}src: Implement clock slaving if the pipeline clock is not the decklink clock 2015-02-09 17:17:37 +01:00
Sebastian Dröge
408f0870a6 decklinkaudio{src,sink}: Only start streams / scheduled playback if there is a videosrc at this point 2015-02-09 17:15:21 +01:00
Sebastian Dröge
349113d15a decklinkaudiosrc: Don't provide a clock
The audio source only works together with the video source, and the video
source is already providing a clock.
2015-02-09 17:04:27 +01:00
Sebastian Dröge
4de7fdfa3a decklinkaudiosink: Throttle reading from the ringbuffer
The driver has an internal buffer of unspecified and unconfigurable size, and
it will pull data from our ring buffer as fast as it can until that is full.
Unfortunately that means that we pull silence from the ringbuffer unless its
size is by conincidence larger than the driver's internal ringbuffer.

The good news is that it's not required to completely fill the buffer for
proper playback. So we now throttle reading from the ringbuffer whenever
the driver has buffered more than half of our ringbuffer size by waiting
on the clock for the amount of time until it has buffered less than that
again.
2015-02-09 16:22:39 +01:00
Sebastian Dröge
a6bcd09b6c decklinkaudiosink: Start scheduled playback when going to PLAYING
The ringbuffer's acquire() is too early, and ringbuffer's start() will only be
called after the clock has advanced a bit... which it won't unless we start
scheduled playback.
2015-02-09 16:22:39 +01:00
Alessandro Decina
052e4804eb vtdec: fix texture sharing on iOS
Move GLES calls to the context thread. Fix
CVOpenGLESTextureCacheCreateTextureFromImage usage on iOS.
2015-02-09 21:18:24 +11:00
Alessandro Decina
ad08cdf8e1 libgstgl: set user data and GDestroyNotify for wrapped textures 2015-02-09 20:46:51 +11:00
Sebastian Dröge
22b0a4dad5 decklink{audio,video}src: Take the capture times from the pipeline clock
Not from the decklink clock. Both will return exactly the same time once the
decklink clock got slaved to the pipeline clock and received the first
observation, but until then it will return bogus values. But as both return
exactly the same values, we can as well use the pipeline clock directly.
2015-02-08 21:56:04 +01:00
Tim-Philipp Müller
6725ff5598 videoaggregator: don't set element factory details
This is an abstract base class, not an element.
2015-02-08 14:32:22 +00:00
Rob Clark
b4df029030 dvdspu: some optimizations
Detect invisible pixels, and skip gstspu_vobsub_blend_comp_buffers()
when there are only invisible pixels.  This significantly reduces the
CPU load in cases of DVDs which don't use the clip_rect to exclude
processing for parts of the screen where the video is visible.

https://bugzilla.gnome.org/show_bug.cgi?id=667221
2015-02-08 06:17:01 +11:00
Jan Schmidt
cb7c99b8ee dvdspu: Avoid warning if the video for a still frame arrives after the SPU
When we don't have a segment on the video pad yet during a still frame
don't freak out.
2015-02-08 06:16:49 +11:00
Jan Schmidt
52166768ed resindvd: Don't send meaningless segment position downstream
Don't place a meaningless value in the position field in
outgoing segments - values have to be in the range start->stop.

https://bugzilla.gnome.org/show_bug.cgi?id=740558
2015-02-08 06:16:12 +11:00
Luis de Bethencourt
1f670cb9ba Revert "aiffparse: make sure we have data before handling sink event"
This reverts commit a4baabb48d.
2015-02-06 16:22:26 +00:00
Luis de Bethencourt
a4baabb48d aiffparse: make sure we have data before handling sink event
Check AIFF_PARSE_DATA is True before handling sink event, if not goto exit.
2015-02-06 15:13:43 +00:00
Tim-Philipp Müller
dc2c4840f9 audiomixer: remove now-unused base_time field in object structure 2015-02-06 10:47:20 +00:00
Sebastian Dröge
a5002ea59d aggregator: Streamline latency calculations
Min latency can never be invalid, latency property can never be invalid
either. So no need to check for all these things in various places.
2015-02-06 11:03:57 +01:00
Sebastian Dröge
65b1db2aa2 aggregator: If upstream has no max latency but the subclass has, take the subclass max latency 2015-02-06 11:03:56 +01:00
Sebastian Dröge
ea50bc1917 aggregator: Fix min>max latency error check
We have to include the upstream latency, our own latency and the subclass
latency in the calculations.

FIXME: This is still not entirely correct
2015-02-06 11:03:56 +01:00