Commit graph

15308 commits

Author SHA1 Message Date
Olivier Crête
7917bea855 avidemux: Stop without posting error on flushing
This could just be a normal pipeline shutdown.
2015-07-25 03:25:28 -04:00
Hyunjun Ko
afcd462918 v4l2bufferpool: set GST_BUFFER_COPY_FLAGS to copy flags also
https://bugzilla.gnome.org/show_bug.cgi?id=752618
2015-07-23 10:19:46 -04:00
Tim-Philipp Müller
c1382e97fa tests: add minmal matroskademux test for subtitle output
Some of the subtitle chunks will have embedded
NUL-terminators (last three), some don't (first three),
some will have markup, some won't, some will be valid
UTF-8 (all but last), some won't (last stanza).

https://bugzilla.gnome.org/show_bug.cgi?id=752421
2015-07-21 14:25:12 +01:00
Dimitrios Christidis
744167056c matroskademux: fix for subtitle buffers with NUL terminators
Commit 45892ec8 created a regression where g_utf8_validate() would fail
if the subtitle buffer had a NUL terminator as part of the data.

https://bugzilla.gnome.org/show_bug.cgi?id=752421
2015-07-21 14:25:12 +01:00
Stian Selnes
45e05706e2 rtpvp8depay: Check available bytes before copy
Need to check that the number of bytes we want to copy from the adapter
actually is available and handle the error case gracefully. This error
may happen if malformed packets are received and we don't have a
complete frame.

https://bugzilla.gnome.org/show_bug.cgi?id=752663
2015-07-21 13:14:01 +01:00
Paul Hyunil
3740e69957 qtdemux: Support subtitle when track subtype is fourcc_subt
https://bugzilla.gnome.org/show_bug.cgi?id=752655
2015-07-21 12:24:15 +01:00
Song Bing
c5950cd04a v4l2bufferpool: Set timestamp when queue buffer.
Should set timestamp when queue buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=752618
2015-07-20 16:49:41 -04:00
Havard Graff
764bbf99a8 rtpmux: handle different ssrc's on sinkpads
Do this by not putting the ssrc from the src pads in the caps used to
probe other sinkpads, and then  intersecting with it later.

https://bugzilla.gnome.org/show_bug.cgi?id=752491
2015-07-16 16:46:11 -04:00
Tim-Philipp Müller
2e3a5ba227 Update mailing list address from sourceforge to freedesktop 2015-07-16 17:19:03 +01:00
Dimitrios Christidis
45892ec8be matroskademux: fix trailing '*' displayed with some text subtitles
The subtitle buffer we push out should not include a NUL terminator
as part of the data, we just add such a terminator for safety, but
it should not be included in the buffer size.

A NUL terminator is not valid UTF-8, so checks will fail if it's
included in the size, and the NUL will be replaced by the fallback
character specified when converting, i.e. '*'.

https://bugzilla.gnome.org/show_bug.cgi?id=752421
2015-07-16 13:18:06 +01:00
Wim Taymans
0421d40d9c pulse: add properties to GstDevice
Add the extra properties we get from pulse to the GstDevice we expose
with the device monitor
2015-07-15 18:29:21 +02:00
Ravi Kiran K N
1c00801585 audiofx: Fix typo in example pipelines
Fix typo in example pipelines of audiowsincband and audioinvert.

https://bugzilla.gnome.org/show_bug.cgi?id=752416
2015-07-15 13:51:13 +01:00
George Kiagiadakis
bbfa46363c splitmuxsink: add a "format-location" signal that allows better control over filenames
In certain applications, splitting into files named after a base
location template and an incremental sequence number is not enough.

This signal gives more fine-grained control to the application to
decide how to name the files.

https://bugzilla.gnome.org/show_bug.cgi?id=750106
2015-07-14 18:45:49 +02:00
Ilya Konstantinov
e9fbdc3682 osxaudiosrc: no resampling on OS X
Unlike Remote IO, AUHAL doesn't have built-in resampling
for sources -- confirmed by Core Audio engineer Doug Wyatt:
http://lists.apple.com/archives/coreaudio-api/2006/Sep/msg00088.html

https://bugzilla.gnome.org/show_bug.cgi?id=743758
2015-07-14 17:49:50 +05:30
Ilya Konstantinov
f33954ae1d osxaudiosrc: avoid get_channel_layout
This only produces a warning and serves no purpose.

https://bugzilla.gnome.org/show_bug.cgi?id=743758
2015-07-14 17:49:50 +05:30
Arun Raghavan
8f0f976375 osxaudio: Avoid making a duplicate structure in caps for mono/stereo case
For 1ch or 2ch devices, we just need to set the caps to allow both
options since CoreAudio will up/downmix appropriately.

Also fixes the condition for the 2ch case to be exact, rather than at
least 2 channels since the downmix will not take place in the >stereo
case.
2015-07-14 17:49:50 +05:30
Arun Raghavan
691ecebe22 osxaudio: Don't set the format on an initialized AudioUnit
We need to initialize the AudioUnit early to be able to probe the
underlying device, but according to the AudioUnitInitialize() and
AudioUnitUninitialize() documentation, format changes should be done
while the AudioUnit is uninitialized. So we explicitly uninitialize the
AudioUnit during a format change and reinitialize it when we're done.
2015-07-14 17:49:50 +05:30
Arun Raghavan
22f6d62796 osxaudio: Minor spelling fix (unitialize -> uninitialize) 2015-07-14 17:49:50 +05:30
Ilya Konstantinov
f107f7306b osxaudio: Fix lockup in _audio_unit_property_listener
_audio_unit_property_listener is called either from a Core Audio thread
or as a result of a Core Audio API (e.g. AudioUnitInitialize)
from our own thread. In the latter case, osxbuf can be already locked
(GStreamer's mutex is not recursive).

We introduce the flag cached_caps_valid and use it instead of nullifying
cached_caps when we cannot lock on osxbuf.

https://bugzilla.gnome.org/show_bug.cgi?id=743758
2015-07-14 17:49:50 +05:30
Ilya Konstantinov
a8b2666aa7 osxaudio: Invalidate cached caps on format change
Listen for changes in hardware stream format and channel layout, and
invalidate cached caps (since they contain the preferred caps).

https://bugzilla.gnome.org/show_bug.cgi?id=743758
2015-07-14 17:49:50 +05:30
Ilya Konstantinov
0e5d698c6f osxaudio: Overhaul of probing caps
- Probing caps is unified between source and sink
- Hardware stream format is now reported as preferred capabilities
  (dynamically updated when hardware configuration changes)
- Get hardware channel layout from Remote IO just like from HAL
- More comprehensive mapping between AudioChannelLabel and
  GstAudioChannelPosition
- Support for unpositioned channel layouts
- Announce stereo-mono upmixing/downmixing in caps

https://bugzilla.gnome.org/show_bug.cgi?id=743758
2015-07-14 17:49:50 +05:30
Ilya Konstantinov
8a8884e150 osxaudio: AudioUnitInitialize on open
Call AudioUnitInitialize upon open. Otherwise, we cannot get
(hardware) stream format nor channel layout from the outer scope.
2015-07-14 17:49:50 +05:30
Tim-Philipp Müller
6717c86061 rtp: depayloaders: implement process_rtp_packet() vfunc
For more optimised RTP packet handling: means we don't
need to map the input buffer again but can just re-use
the mapping the base class has already done.

https://bugzilla.gnome.org/show_bug.cgi?id=750235
2015-07-12 14:28:29 +01:00
Tim-Philipp Müller
fe787425bc rtpvrawdepay: implement process_rtp_packet() vfunc
For more optimised RTP packet handling: means we don't
need to map the input buffer again but can just re-use
the map the base class has already done.

https://bugzilla.gnome.org/show_bug.cgi?id=750235
2015-07-12 14:28:25 +01:00
Sebastian Dröge
582ade2c42 rtpjitterbuffer: Fix indention 2015-07-10 00:13:32 +03:00
Sebastian Dröge
ae8acc0973 rtpjitterbuffer: Always estimate DTS from the current clock time
Estimating it from the RTP time will give us the PTS, so in cases of PTS!=DTS
we would produce wrong DTS. As now the estimated DTS is based on the clock,
don't store it in the jitterbuffer items as it would otherwise be used in the
skew calculations and would influence the results. We only really need the DTS
for timer calculations.

https://bugzilla.gnome.org/show_bug.cgi?id=749536
2015-07-10 00:13:22 +03:00
Thiago Santos
a1bee6eb46 gitignore: ignore rtph263 test 2015-07-09 09:26:09 -03:00
Thiago Santos
241e0c2722 rtpjitterbuffer: fix build error with gcc (Debian 4.9.2-21) 4.9.2
Replace static constants with macros to make gcc happy

  CC       elements/elements_rtpjitterbuffer-rtpjitterbuffer.o
elements/rtpjitterbuffer.c:387:1: error: initializer element is not constant
 static const GstClockTime PCMU_BUF_DURATION = PCMU_BUF_MS * GST_MSECOND;
 ^
elements/rtpjitterbuffer.c:388:1: error: initializer element is not constant
 static const guint PCMU_BUF_SIZE = 64000 * PCMU_BUF_MS / 1000;
 ^
elements/rtpjitterbuffer.c:390:5: error: initializer element is not constant
     PCMU_BUF_CLOCK_RATE * PCMU_BUF_MS / 1000;
2015-07-08 23:49:12 -03:00
Thiago Santos
3edf9e4f58 rtpjitterbuffer: run indent and fix some comments
Fix indent on this file and break some comment lines into two to make
it fit 80 chars per line
2015-07-08 23:49:09 -03:00
Thiago Santos
30b3aa3030 qtdemux: rework segment event handling for adaptive streaming
When a new time segment is received upstream is going to restart
with a new atom. Make the neededbytes and todrop variables
reflect that to avoid waiting too much or dropping the
initial bytes that contain the header.
2015-07-08 23:23:53 -03:00
Thiago Santos
38520a1e12 qtdemux: push data from adapter before starting new segment
The adapter might have data remaining from the previous segment,
push it all before clearing the adapter and starting a new segment.

It can accumulate data if it had pushed and got not-linked, returning
immediately without processing all the data. Before starting a new
segment this data should be handled.
2015-07-08 23:23:53 -03:00
Sebastian Dröge
6e7c724afa rtpjitterbuffer: Calculate DTS from the clock if we had none for the first packet after a reset
https://bugzilla.gnome.org/show_bug.cgi?id=749536
2015-07-08 23:19:52 +03:00
Havard Graff
ddd032f56b rtpjitterbuffer: fix gap-time calculation and remove "late"
The amount of time that is completely expired and not worth waiting for,
is the duration of the packets in the gap (gap * duration) - the
latency (size) of the jitterbuffer (priv->latency_ns). This is the duration
that we make a "multi-lost" packet for.

The "late" concept made some sense in 0.10 as it reflected that a buffer
coming in had not been waited for at all, but had a timestamp that was
outside the jitterbuffer to wait for. With the rewrite of the waiting
(timeout) mechanism in 1.0, this no longer makes any sense, and the
variable no longer reflects anything meaningful (num > 0 is useless,
the duration is what matters)

Fixed up the tests that had been slightly modified in 1.0 to allow faulty
behavior to sneak in, and port some of them to use GstHarness.

https://bugzilla.gnome.org/show_bug.cgi?id=738363
2015-07-08 23:18:48 +03:00
Stian Selnes
40524e5a49 Revert "rtpjitterbuffer: Fix expected_dts calc in calculate_expected"
This reverts commit 05bd708fc5.

The reverted patch is wrong and introduces a regression because there
may still be time to receive some of the packets included in the gap
if they are reordered.
2015-07-08 23:18:48 +03:00
Thiago Santos
ee7ddf6c67 qtdemux: flush samples before adding more from moof
Avoids accumulating all samples from a fragmented stream that could
lead to a 'index-too-big' error once it goes over 50MB of data. It
could reach that before 2h of playback so it doesn't take that long.

As upstream elements are providing data in time format they should
be the ones that have more information about the full media index
and should be able to seek if possible.
2015-07-08 11:53:44 -03:00
Thiago Santos
6ee4b31c0e qtdemux: rename upstream_newsegment to upstream_format_is_time
upstream_newsegment isn't really clear on what it means, it is set
to TRUE when the upstream element sends a segment in TIME format, so
rename it to be more clear about it.

It is important to know this because it means that upstream has
a notion of time and qtdemux is likely being driven by an upstream
element that is reading from a higher level abstraction than a file,
such as a DASH, MSS or DLNA element.
2015-07-08 11:53:44 -03:00
Thiago Santos
5994b30257 qtdemux: fix leak by flushing previous sample info from trak
In fragmented streaming, multiple moov/moof will be parsed and their
previously stored samples array might leak when new values are parsed.
The parse_trak and callees won't free the previously stored values
before parsing the new ones.

In step-by-step, this is what happens:

1) initial moov is parsed, traks as well, streams are created. The
   trak doesn't contain samples because they are in the moof's trun
   boxes. n_samples is set to 0 while parsing the trak and the samples
   array is still NULL.
2) moofs are parsed, and their trun boxes will increase n_samples and
   create/extend the samples array
3) At some point a new moov might be sent (bitrate switching, for example)
   and parsing the trak will overwrite n_samples with the values from
   this trak. If the n_samples is set to 0 qtdemux will assume that
   the samples array is NULL and will leak it when a new one is
   created for the subsequent moofs.

This patch makes qtdemux properly free previous sample data before
creating new ones and adds an assert to catch future occurrences of
this issue when the code changes.
2015-07-08 11:53:44 -03:00
Thiago Santos
63f35eeb12 qtdemux: fix index size check and debug message
It is allocating samples_count + n_samples, not only n_samples
2015-07-08 11:53:44 -03:00
Sebastian Dröge
4e23481d9f rtpjitterbuffer: Calculate receive time if we don't have any
This is required to properly schedule packet loss timers and make
sure all our calculations work properly.

https://bugzilla.gnome.org/show_bug.cgi?id=749536
2015-07-08 17:02:05 +03:00
Sebastian Dröge
243730ced4 rtpjitterbuffer: Handle seqnum gaps in TCP streams without erroring out or overflowing calculations
That is, handle DTS==GST_CLOCK_TIME_NONE correctly.

https://bugzilla.gnome.org/show_bug.cgi?id=749536
2015-07-08 15:15:00 +03:00
Vineeth T M
5439fc9a0c avidemux: fix event leak
when seek fails in avidemux, event is not being freed.

https://bugzilla.gnome.org/show_bug.cgi?id=752117
2015-07-08 12:57:43 +01:00
Stian Selnes
8a0dbff3f4 rtph263depay: Make sure payload is large enough
Plus new unit test.

https://bugzilla.gnome.org/show_bug.cgi?id=752112
2015-07-08 11:36:55 +01:00
Vineeth TM
ffe9cbc1f6 rtpklvdepay: fix printf format compiler warning
v_len is of type guint64, but while print the value(16 + len_size + v_len)
G_GSIZE_FORMAT is being used instead of G_GUINT64_FORMAT

https://bugzilla.gnome.org/show_bug.cgi?id=752100
2015-07-08 08:49:37 +01:00
Tim-Philipp Müller
5314ee3fa4 docs: add new RTP elements to docs 2015-07-07 20:26:47 +01:00
Tim-Philipp Müller
4ed4d0b84c tests: rtp-payloading: add basic unit test for KLV payloading
Also make it so that the mtu is always set if specified, not
only in case of the rather weird bufferlist test code path.
This allows us to easily make the payloader fragment a payload
across multiple output packets by setting a small MTU on it.
2015-07-07 20:11:28 +01:00
Tim-Philipp Müller
b105e1e3d1 rtpklvdepay: improve start detection and handle fragmented KLV units 2015-07-07 20:11:27 +01:00
Tim-Philipp Müller
740f10bae9 rtp: add SMPTE 336M KLV metadata depayloader
http://tools.ietf.org/html/rfc6597
2015-07-07 20:11:27 +01:00
Tim-Philipp Müller
7db7da1acb rtp: add SMPTE 336M KLV metadata payloader
http://tools.ietf.org/html/rfc6597
2015-07-07 20:11:23 +01:00
Stefan Sauer
12930c2f8c docs: fix "Symbol name not found at the start of the comment block"
Add symbols or change comment into a regular comment.
2015-07-07 17:12:02 +02:00
Stefan Sauer
093e8f8a75 docs: remove outdated doc strings 2015-07-07 17:12:02 +02:00