Prashant Gotarne [Fri, 14 Aug 2015 04:06:09 +0000 (09:36 +0530)]
audioecho: make sure buffer gets reallocated if max_delay changes
https://bugzilla.gnome.org/show_bug.cgi?id=753490
Oleksij Rempel [Thu, 9 Jul 2015 07:51:26 +0000 (09:51 +0200)]
vp8enc: provide support for multiple pass cache files
Some files may provide different caps insight of one stream. Since vp8enc
support caps reinit, we should support cache reinit too.
If more then file cache file will be created, the naming will be:
cache
cache.1
cache.2
...
https://bugzilla.gnome.org/show_bug.cgi?id=747728
Ramiro Polla [Wed, 15 Apr 2015 20:51:51 +0000 (22:51 +0200)]
rtpmp4gdepay: fix timestamps for RTP packets with multiple AUs
Use constantDuration to calculate the timestamp of non-first AU in the
RTP packet.
If constantDuration is not present in the MIME parameters, its value
must be calculated based on the timing information from two consecutive
RTP packets with AU-Index equal to 0.
https://bugzilla.gnome.org/show_bug.cgi?id=747881
Thiago Santos [Fri, 14 Aug 2015 09:43:13 +0000 (06:43 -0300)]
souphttpsrc: remove unnecessary if, g_free is null safe
Alex Ashley [Fri, 14 Aug 2015 07:33:56 +0000 (08:33 +0100)]
souphttpsrc: add property to set HTTP method
To allow souphttpsrc to be use HTTP methods other than GET
(e.g. HEAD), add a "method" property that is a string. If this
property is not set, GET is used.
https://bugzilla.gnome.org/show_bug.cgi?id=752413
Edward Hervey [Fri, 14 Aug 2015 09:13:01 +0000 (11:13 +0200)]
check: Rename states unit test
Makes it easier to differentiate from other modules states unit test
Sebastian Dröge [Fri, 14 Aug 2015 07:21:25 +0000 (09:21 +0200)]
goom: Rename get_type() function of base class to prevent symbol conflicts
This is a problem when statically linking.
Sebastian Dröge [Thu, 13 Aug 2015 14:32:55 +0000 (16:32 +0200)]
rtpjitterbuffer: Keep the DTS estimate if we got no DTS after a jitterbuffer reset
Otherwise we will just output buffers without timestamps after a reset if no
timestamps are provided by upstream, e.g. when using RTSP over TCP.
https://bugzilla.gnome.org/show_bug.cgi?id=749536
Ravi Kiran K N [Wed, 12 Aug 2015 11:46:01 +0000 (17:16 +0530)]
matroska: Remove unused variable
https://bugzilla.gnome.org/show_bug.cgi?id=753556
Sebastian Dröge [Tue, 4 Aug 2015 17:59:17 +0000 (20:59 +0300)]
rtp: Copy metadata in the (de)payloader, but only the relevant ones
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without
tags or with only the video tag.
https://bugzilla.gnome.org/show_bug.cgi?id=751774
Thiago Santos [Mon, 10 Aug 2015 21:20:15 +0000 (18:20 -0300)]
qtdemux: fix small typo in comment
Nicolas Dufresne [Mon, 10 Aug 2015 20:19:18 +0000 (16:19 -0400)]
goom2k1/doc: Fixup previous commit
Nicolas Dufresne [Mon, 10 Aug 2015 19:55:19 +0000 (15:55 -0400)]
goom2k1/doc: Use GstGoom2k1 namespace
The doc generator isn't happy when we have class name clash. Simply
use it's own namespace.
Prashant Gotarne [Mon, 10 Aug 2015 11:40:42 +0000 (17:10 +0530)]
audioecho: removed unused variable in set_property
unused local variable 'delay' is removed.
https://bugzilla.gnome.org/show_bug.cgi?id=753450
Tim-Philipp Müller [Mon, 10 Aug 2015 11:45:27 +0000 (12:45 +0100)]
qtdemux: fix suboptimal queue iteration code
Tim-Philipp Müller [Sun, 9 Aug 2015 16:25:45 +0000 (17:25 +0100)]
qtdemux: don't use glib 2.44-only API
Alex Ashley [Wed, 29 Jul 2015 13:14:50 +0000 (14:14 +0100)]
qtdemux: add support for ISOBMFF Common Encryption
This commit adds support for ISOBMFF Common Encryption (cenc), as
defined in ISO/IEC 23001-7. It uses a GstProtection event to
pass the contents of PSSH boxes to downstream decryptor elements
and attached GstProtectionMeta to each sample.
https://bugzilla.gnome.org/show_bug.cgi?id=705991
Hyunjun Ko [Mon, 10 Aug 2015 05:13:50 +0000 (14:13 +0900)]
rtph264depay: checking if depay has sps/pps nals before insertion
https://bugzilla.gnome.org/show_bug.cgi?id=753430
Tim-Philipp Müller [Sat, 8 Aug 2015 15:44:49 +0000 (16:44 +0100)]
matroskamux: fix outdated comment
The default behaviour was changed in the 0.10 -> 1.x
transition, but the comment was not updated.
Sebastian Dröge [Sat, 8 Aug 2015 15:42:22 +0000 (17:42 +0200)]
rtptheorapay: If flushing a packet failed, go out of the loop immediately
Sebastian Dröge [Sat, 8 Aug 2015 15:41:02 +0000 (17:41 +0200)]
rtpvorbispay: If flushing a packet failed, go out of the loop immediately
Sebastian Dröge [Sat, 8 Aug 2015 15:34:50 +0000 (17:34 +0200)]
rtptheorapay: Extract pixel format from the ident header to put it into the sampling field of the caps
We always put 4:2:0 into the caps before, which obviously is wrong for 4:2:2
and 4:4:4 formats.
George Kiagiadakis [Thu, 6 Aug 2015 15:46:13 +0000 (17:46 +0200)]
rtpklv(de)pay: add "RTP" in the klass string
GstRTSPMedia uses this classification to detect the real payloader
inside a dynpay bin and asserts if it doesn't find it, therefore
it is required
https://bugzilla.gnome.org/show_bug.cgi?id=753325
Thiago Santos [Wed, 5 Aug 2015 14:13:09 +0000 (11:13 -0300)]
tests: rtpaux: use a dynamic pt in the test
1) Tests that using dynamic PT instead of the default ones work
2) If we ever decide to change the codec here we don't need to
worry about change the PT for the default one of the new codec
in the test
https://bugzilla.gnome.org/show_bug.cgi?id=746445
Hyunjun Ko [Wed, 5 Aug 2015 01:53:15 +0000 (10:53 +0900)]
rtprtxsend: print valid type where guint32 is expected
https://bugzilla.gnome.org/show_bug.cgi?id=746445
Hyunjun Ko [Thu, 6 Aug 2015 02:33:37 +0000 (11:33 +0900)]
rtppayload: set standard payload type as default
Initialize the PT to the default value of the codec and check if
it is still the default before declaring the pt to be dynamic or
not when setting the caps.
Also use the PT constants from the rtp lib when possible
https://bugzilla.gnome.org/show_bug.cgi?id=747965
Thiago Santos [Sun, 26 Jul 2015 15:07:56 +0000 (12:07 -0300)]
qtdemux: store the moof-offset also for push mode
It will be used in some cases for getting the correct offsets
from trun atoms.
https://bugzilla.gnome.org/show_bug.cgi?id=752603
Thiago Santos [Sun, 26 Jul 2015 05:09:24 +0000 (02:09 -0300)]
qtdemux: handle default-base-is-moof flag
Handle the flag from the tfhd that signals the base offset to
start from the moof atom
https://bugzilla.gnome.org/show_bug.cgi?id=752603
Glen Diener [Thu, 30 Jul 2015 00:54:35 +0000 (18:54 -0600)]
matroskademux: Preserve forward referenced track tags
https://bugzilla.gnome.org/show_bug.cgi?id=752850
Thiago Santos [Tue, 4 Aug 2015 21:07:35 +0000 (18:07 -0300)]
tests: rtpaux: fix test failure
The RTP PT for alaw is 8.
Less than 50 packets are received in the length of this test so it
would never drop a buffer or would drop only the last buffer and
it would fail sometimes when the received wouldn't receive the
retransmission packet in time.
https://bugzilla.gnome.org/show_bug.cgi?id=746445
Sebastian Dröge [Tue, 4 Aug 2015 17:59:17 +0000 (20:59 +0300)]
rtpstreamdepay: Only allow activation in push mode
We need a proper caps event from upstream with the full RTP caps as we can't
create caps ourselves from thin air. Fixes usage of rtpstreamdepay after e.g.
a filesrc or any other element that supports pull mode.
https://bugzilla.gnome.org/show_bug.cgi?id=753066
Tim-Philipp Müller [Tue, 4 Aug 2015 15:28:17 +0000 (16:28 +0100)]
soup: fix typo in translated string
https://bugzilla.gnome.org/show_bug.cgi?id=753240
Sebastian Dröge [Tue, 4 Aug 2015 09:25:46 +0000 (12:25 +0300)]
rtph264depay: Put the profile and level into the caps
Sebastian Dröge [Tue, 4 Aug 2015 09:09:12 +0000 (12:09 +0300)]
rtph264depay: Only update the srcpad caps if something else than the codec_data changed
h264parse does the same, let's keep the behaviour consistent. As we now
include the codec_data inside the stream too here, this causes less caps
renegotiation.
Sebastian Dröge [Tue, 4 Aug 2015 08:48:27 +0000 (11:48 +0300)]
rtph264depay: PPS replaces and old PPS if it has the same id, independent of SPS id
The spec says:
When a picture parameter set NAL unit with a particular value of
pic_parameter_set_id is received, its content replaces the content of the
previous picture parameter set NAL unit, in decoding order, with the same
value of pic_parameter_set_id (when a previous picture parameter set NAL unit
with the same value of pic_parameter_set_id was present in the bitstream).
Thiago Santos [Mon, 3 Aug 2015 16:45:59 +0000 (13:45 -0300)]
splitmuxsink: remove extra \n at debug message
Thiago Santos [Mon, 3 Aug 2015 16:42:20 +0000 (13:42 -0300)]
splitmuxsink: prevent deadlock when states change too fast
If the GOP is completed, pads have to start gathering for the
next one but it is possible that the the state might go to
COLLECTING_GOP_START and back to WAITING_GOP_COMPLETE before the
thread has a chance to wake up and proceed, leaving it trapped in
the check_completed_gop loop and deadlocking the other threads
waiting for it to advance.
To solve it, this patch also checks that tha input running time
hasn't changed to prevent this scenario.
Sebastian Dröge [Mon, 3 Aug 2015 14:55:01 +0000 (17:55 +0300)]
rtph264depay: Insert SPS/PPS NALs into the stream
h264parse does the same and this fixes decoding of some streams with 32 SPS
(or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255), but
the field in the codec_data for the number of SPS or PPS is only 5 (or 8) bit.
As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
This looks like a mistake in the part of the spec about the codec_data.
Eunhae Choi [Thu, 30 Jul 2015 02:29:27 +0000 (11:29 +0900)]
souphttpsrc: handle empty http proxy string
1) If the system http_proxy environment variable is not set
or set to an empty string, we must not set proxy to avoid
http connection error.
2) In case of proxy property setting, if user want to clear
the proxy setting, they should be able to set it to NULL or
an empty string again, so this is fixed too.
3) Check if the proxy string was parsed correctly.
https://bugzilla.gnome.org/show_bug.cgi?id=752866
Ravi Kiran K N [Wed, 29 Jul 2015 10:16:20 +0000 (15:46 +0530)]
dvdemux: remove unused variable
Remove unused variable 'framecount' from dvdemux
https://bugzilla.gnome.org/show_bug.cgi?id=753008
Vineeth TM [Thu, 30 Jul 2015 06:32:09 +0000 (15:32 +0900)]
rtspsrc: assertion error due to wrong condition check
In media to caps function, reserved_keys array is being used for variable i,
leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
changed it to variable j
https://bugzilla.gnome.org/show_bug.cgi?id=753009
Vineeth TM [Thu, 30 Jul 2015 06:21:20 +0000 (15:21 +0900)]
rtpmp4vdepay: rtpbuffer is being unref'ed twice
process_rtp_packet doesn't transfer the rtp buffer to mp4v_process_depay
the refernce should not be removed here
https://bugzilla.gnome.org/show_bug.cgi?id=753042
Sebastian Dröge [Wed, 29 Jul 2015 10:26:46 +0000 (11:26 +0100)]
rtspsrc: Strip keys from the fmtp that we use internally in our caps
Skip keys from the fmtp, which we already use ourselves for the
caps. Some software is adding random things like clock-rate into
the fmtp, and we would otherwise here set a string-typed clock-rate
in the caps... and thus fail to create valid RTP caps
https://bugzilla.gnome.org/show_bug.cgi?id=753009
Jan Schmidt [Wed, 29 Jul 2015 09:28:33 +0000 (19:28 +1000)]
splitmuxsink: Support mpegtsmux as a muxer.
As a fallback, look for a pad template sink_%d on
the muxer when requesting pads, to support mpegtsmux
https://bugzilla.gnome.org/show_bug.cgi?id=752999
Jan Schmidt [Wed, 24 Jun 2015 15:35:27 +0000 (01:35 +1000)]
splitmuxsrc: Use a separate lock to delay typefind.
Don't hold the main splitmux part lock over
the parent state change function, as it prevents
posting error messages that happen. Since the purpose
is to prevent typefinding from proceeding, use a
separate mutex just for that.
Vineeth TM [Wed, 29 Jul 2015 04:43:50 +0000 (13:43 +0900)]
matroska: fix memory leak
After adding to tag list, key_val is not being free'd
resulting in memory leak
https://bugzilla.gnome.org/show_bug.cgi?id=752992
Manasa Athreya [Mon, 27 Jul 2015 04:34:14 +0000 (13:34 +0900)]
qtdemux: fix 16-bit PCM audio advertised with 'raw ' fourcc
'NONE' and 'raw ' fourcc don't always contain U8 audio, it can
be more bits as well, in which case it's just like 'twos'.
https://bugzilla.gnome.org/show_bug.cgi?id=752613
Dimitrios Katsaros [Fri, 24 Jul 2015 13:10:05 +0000 (15:10 +0200)]
v4l2: Allow framerate to be large then 100pfs
This limit was arbitrary. We still fixate near 100pfs for compatibility.
https://bugzilla.gnome.org/show_bug.cgi?id=752825
Olivier Crête [Sat, 25 Jul 2015 07:25:28 +0000 (03:25 -0400)]
avidemux: Stop without posting error on flushing
This could just be a normal pipeline shutdown.
Hyunjun Ko [Thu, 23 Jul 2015 06:00:08 +0000 (15:00 +0900)]
v4l2bufferpool: set GST_BUFFER_COPY_FLAGS to copy flags also
https://bugzilla.gnome.org/show_bug.cgi?id=752618
Tim-Philipp Müller [Thu, 16 Jul 2015 17:09:30 +0000 (18:09 +0100)]
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
Dimitrios Christidis [Thu, 16 Jul 2015 15:49:26 +0000 (18:49 +0300)]
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
Stian Selnes [Tue, 21 Jul 2015 11:31:05 +0000 (13:31 +0200)]
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
Paul Hyunil [Thu, 16 Jul 2015 00:32:36 +0000 (09:32 +0900)]
qtdemux: Support subtitle when track subtype is fourcc_subt
https://bugzilla.gnome.org/show_bug.cgi?id=752655
Song Bing [Mon, 20 Jul 2015 08:59:40 +0000 (16:59 +0800)]
v4l2bufferpool: Set timestamp when queue buffer.
Should set timestamp when queue buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=752618
Havard Graff [Thu, 16 Jul 2015 13:12:17 +0000 (15:12 +0200)]
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
Tim-Philipp Müller [Thu, 16 Jul 2015 16:19:03 +0000 (17:19 +0100)]
Update mailing list address from sourceforge to freedesktop
Dimitrios Christidis [Wed, 15 Jul 2015 10:44:52 +0000 (13:44 +0300)]
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
Wim Taymans [Wed, 15 Jul 2015 16:23:05 +0000 (18:23 +0200)]
pulse: add properties to GstDevice
Add the extra properties we get from pulse to the GstDevice we expose
with the device monitor
Ravi Kiran K N [Wed, 15 Jul 2015 11:50:20 +0000 (17:20 +0530)]
audiofx: Fix typo in example pipelines
Fix typo in example pipelines of audiowsincband and audioinvert.
https://bugzilla.gnome.org/show_bug.cgi?id=752416
George Kiagiadakis [Wed, 15 Apr 2015 16:27:04 +0000 (18:27 +0200)]
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
Ilya Konstantinov [Wed, 15 Apr 2015 17:13:27 +0000 (20:13 +0300)]
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
Ilya Konstantinov [Wed, 15 Apr 2015 15:29:14 +0000 (18:29 +0300)]
osxaudiosrc: avoid get_channel_layout
This only produces a warning and serves no purpose.
https://bugzilla.gnome.org/show_bug.cgi?id=743758
Arun Raghavan [Tue, 7 Apr 2015 10:10:14 +0000 (15:40 +0530)]
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.
Arun Raghavan [Mon, 6 Apr 2015 10:52:34 +0000 (16:22 +0530)]
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.
Arun Raghavan [Mon, 6 Apr 2015 10:25:59 +0000 (15:55 +0530)]
osxaudio: Minor spelling fix (unitialize -> uninitialize)
Ilya Konstantinov [Sat, 21 Mar 2015 18:34:25 +0000 (20:34 +0200)]
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
Ilya Konstantinov [Thu, 12 Mar 2015 10:15:12 +0000 (12:15 +0200)]
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
Ilya Konstantinov [Mon, 9 Mar 2015 21:34:06 +0000 (23:34 +0200)]
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
Ilya Konstantinov [Mon, 9 Mar 2015 21:15:56 +0000 (23:15 +0200)]
osxaudio: AudioUnitInitialize on open
Call AudioUnitInitialize upon open. Otherwise, we cannot get
(hardware) stream format nor channel layout from the outer scope.
Tim-Philipp Müller [Sun, 12 Jul 2015 13:27:15 +0000 (14:27 +0100)]
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
Tim-Philipp Müller [Wed, 27 May 2015 18:19:27 +0000 (19:19 +0100)]
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
Sebastian Dröge [Thu, 9 Jul 2015 21:13:32 +0000 (00:13 +0300)]
rtpjitterbuffer: Fix indention
Sebastian Dröge [Thu, 9 Jul 2015 20:59:10 +0000 (23:59 +0300)]
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
Thiago Santos [Thu, 9 Jul 2015 12:26:09 +0000 (09:26 -0300)]
gitignore: ignore rtph263 test
Thiago Santos [Thu, 9 Jul 2015 02:47:44 +0000 (23:47 -0300)]
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;
Thiago Santos [Thu, 9 Jul 2015 02:40:45 +0000 (23:40 -0300)]
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
Thiago Santos [Wed, 8 Jul 2015 18:02:24 +0000 (15:02 -0300)]
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.
Thiago Santos [Wed, 8 Jul 2015 15:35:55 +0000 (12:35 -0300)]
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.
Sebastian Dröge [Wed, 8 Jul 2015 16:59:13 +0000 (19:59 +0300)]
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
Havard Graff [Wed, 8 Jul 2015 19:08:36 +0000 (21:08 +0200)]
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
Stian Selnes [Tue, 30 Jun 2015 09:21:31 +0000 (11:21 +0200)]
Revert "rtpjitterbuffer: Fix expected_dts calc in calculate_expected"
This reverts commit
05bd708fc5e881390fe839803b53144393d95ab0.
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.
Thiago Santos [Wed, 8 Jul 2015 02:53:02 +0000 (23:53 -0300)]
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.
Thiago Santos [Wed, 8 Jul 2015 02:56:12 +0000 (23:56 -0300)]
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.
Thiago Santos [Wed, 8 Jul 2015 00:31:08 +0000 (21:31 -0300)]
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.
Thiago Santos [Tue, 7 Jul 2015 19:46:33 +0000 (16:46 -0300)]
qtdemux: fix index size check and debug message
It is allocating samples_count + n_samples, not only n_samples
Sebastian Dröge [Wed, 8 Jul 2015 14:02:05 +0000 (17:02 +0300)]
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
Sebastian Dröge [Wed, 8 Jul 2015 12:13:17 +0000 (15:13 +0300)]
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
Vineeth T M [Wed, 8 Jul 2015 11:31:42 +0000 (20:31 +0900)]
avidemux: fix event leak
when seek fails in avidemux, event is not being freed.
https://bugzilla.gnome.org/show_bug.cgi?id=752117
Stian Selnes [Wed, 8 Jul 2015 10:02:22 +0000 (12:02 +0200)]
rtph263depay: Make sure payload is large enough
Plus new unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=752112
Vineeth TM [Tue, 7 Jul 2015 23:59:49 +0000 (08:59 +0900)]
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
Tim-Philipp Müller [Tue, 7 Jul 2015 19:25:47 +0000 (20:25 +0100)]
docs: add new RTP elements to docs
Tim-Philipp Müller [Tue, 7 Jul 2015 19:07:31 +0000 (20:07 +0100)]
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.
Tim-Philipp Müller [Tue, 7 Jul 2015 18:58:42 +0000 (19:58 +0100)]
rtpklvdepay: improve start detection and handle fragmented KLV units
Tim-Philipp Müller [Sun, 5 Jul 2015 19:25:10 +0000 (20:25 +0100)]
rtp: add SMPTE 336M KLV metadata depayloader
http://tools.ietf.org/html/rfc6597
Tim-Philipp Müller [Sat, 9 Aug 2014 09:08:42 +0000 (10:08 +0100)]
rtp: add SMPTE 336M KLV metadata payloader
http://tools.ietf.org/html/rfc6597
Stefan Sauer [Tue, 7 Jul 2015 14:59:20 +0000 (16:59 +0200)]
docs: fix "Symbol name not found at the start of the comment block"
Add symbols or change comment into a regular comment.
Stefan Sauer [Tue, 7 Jul 2015 14:58:53 +0000 (16:58 +0200)]
docs: remove outdated doc strings
Stefan Sauer [Fri, 3 Jul 2015 21:10:40 +0000 (23:10 +0200)]
docs: add missing plugins and ensure master doc is sorted
Luis de Bethencourt [Tue, 7 Jul 2015 14:54:41 +0000 (15:54 +0100)]
Revert "imagefreeze: Remove impossible error condition"
This reverts commit
d46631c5c7312ad613397f8238c7a9714ae3ae94.
pad only handle EOS events but not EOS flow, and will push the buffer again
resulting in an assertion error. So we should not handle the buffer
and return EOS flow.