Commit graph

8525 commits

Author SHA1 Message Date
Vineeth TM
fb7783f8b0 mpegaudioparse: Fix buffer memory leak during failures
mapped buffer is not being unmapped during failures

https://bugzilla.gnome.org/show_bug.cgi?id=756231
2015-10-12 16:56:30 +03:00
Sebastian Dröge
ca9b6b55e6 matroskamux: Create a TIME segment when creating streamable output
Related to https://bugzilla.gnome.org/show_bug.cgi?id=754435 which
does the same for flvmux.
2015-10-11 11:37:51 +01:00
Havard Graff
240b0ac9f6 flvdemux: output speex vorbiscomment as a GstTagList
This is what speexdec expects.

https://bugzilla.gnome.org/show_bug.cgi?id=755478
2015-10-11 11:12:27 +01:00
Havard Graff
b6f133ba17 flvmux: GST_BUFFER_OFFSETs should be GST_BUFFER_OFFSET_NONE
Or else flvdemux don't understand it

https://bugzilla.gnome.org/show_bug.cgi?id=754435
2015-10-11 11:10:20 +01:00
Havard Graff
cf3a2294da flvmux: use time segment and copy timestamps when streamable
Add a basic test using speex data to verify timestamping.

https://bugzilla.gnome.org/show_bug.cgi?id=754435
2015-10-11 11:09:08 +01:00
Havard Graff
a937c59acd flvdemux: speex is also always 16KHz
This is just a cosmetic change for the logs, since the right caps
for Speex is being set elsewhere.

https://bugzilla.gnome.org/show_bug.cgi?id=755479
2015-10-11 11:07:00 +01:00
Stian Selnes
91a78053c7 rtpmanager: Add 'source-stats' to stats and notify
Add statitics from each rtp source to the rtp session property.
'source-stats' is a GValueArray where each element is a GstStructure of
stats for one rtp source.

The availability of new stats is signaled via g_object_notify.

https://bugzilla.gnome.org/show_bug.cgi?id=752669
2015-10-11 10:57:09 +01:00
Sebastian Dröge
f09da189aa rtpsession: Implement sending of reduced size RTCP packets
https://bugzilla.gnome.org/show_bug.cgi?id=750456
2015-10-11 10:47:47 +01:00
Ravi Kiran K N
82de6384dd audiofx: Remove unused variable
Remove unused variable 'degree' in audiodynamic

https://bugzilla.gnome.org/show_bug.cgi?id=756234
2015-10-10 21:32:18 +01:00
Vineeth TM
b26ce7ba6d qtdemux: Fix memory leak for corrupted file
Free brands before overriding them.

https://bugzilla.gnome.org/show_bug.cgi?id=756226
2015-10-08 15:03:36 +01:00
Sebastian Dröge
2be5416e4a rtpbin: Add missing break 2015-10-07 23:23:45 +01:00
Miguel París Díaz
f321bfeaf4 rtpmanager: Take into account packet rate for max-dropout and max-misorder calculations
https://bugzilla.gnome.org/show_bug.cgi?id=751311
2015-10-07 12:07:18 +01:00
Miguel París Díaz
4c96094fbb rtpmanager: add "max-dropout-time" and "max-misorder-time" props
https://bugzilla.gnome.org/show_bug.cgi?id=751311
2015-10-07 12:06:47 +01:00
Vineeth TM
44008938bb qtmux: Fix date memory leak
When getting date from taglist, the memory should be freed after
using it.

https://bugzilla.gnome.org/show_bug.cgi?id=756171
2015-10-07 11:22:20 +01:00
Vineeth TM
d7a80be3c7 qtmux: Fix sample memory leak
When getting sample from taglist, the memory should be freed after
using it.

https://bugzilla.gnome.org/show_bug.cgi?id=756068
2015-10-05 12:09:26 +01:00
Vineeth TM
15b08e0bd5 cutter: Fix buffer leak
Buffer is added to the internal cache, and pushed only when accumulated
buffer duration crosses 200 ms. So when the chain ends, the buffer accumulated
is not freed. Freeing the cache when the state changes from PAUSED to READY.

https://bugzilla.gnome.org/show_bug.cgi?id=754212
2015-10-05 12:03:33 +01:00
Olivier Crête
58073eaa7a rtpmux: Use default upstream event handling
https://bugzilla.gnome.org/show_bug.cgi?id=752694
2015-10-02 17:39:10 -04:00
Olivier Crête
43c213fc5d rtpmux: As 0xFFFFFFFF is a valid ssrc, check if it has been set
https://bugzilla.gnome.org/show_bug.cgi?id=752694
2015-10-02 17:39:10 -04:00
Havard Graff
d5e26ab909 gstrtpmux: allow the ssrc-property to decide ssrc on outgoing buffers
By not doing this, the muxer is not effectively a rtpmuxer, rather a
funnel, since it should be a single stream that exists the muxer.

If not specified, take the first ssrc seen on a sinkpad, allowing upstream
to decide ssrc in "passthrough" with only one sinkpad.

Also, let downstream ssrc overrule internal configured one

We hence has the following order for determining the ssrc used by
rtpmux:

0. Suggestion from GstRTPCollision event
1. Downstream caps
2. ssrc-Property
3. (First) upstream caps containing ssrc
4. Randomly generated

https://bugzilla.gnome.org/show_bug.cgi?id=752694
2015-10-02 17:39:06 -04:00
Sebastian Dröge
41a82b9706 udpsrc: Fixup last commit 2015-10-02 22:42:20 +03:00
Sebastian Dröge
26588fbdb3 Update GLib dependency to 2.40.0 2015-10-02 22:21:45 +03:00
Miguel París Díaz
bf0e4f65b4 rtpstats: add utility for calculating RTP packet rate 2015-10-02 19:25:27 +01:00
Thiago Santos
df0a31b4ee qtdemux: handle empty segments in seeking adjust
If seeking targets an empty segment skip it as there is no media
offset to get from it. Instead look for the next one.

This doesn't make seeking in push-mode work if you seek to an
empty segment but at least won't get you to wrong offsets.

https://bugzilla.gnome.org/show_bug.cgi?id=753484
2015-10-02 19:23:43 +01:00
George Kiagiadakis
5e4caca709 splitmuxsink: post messages when fragments are being opened and closed
This can be useful for applications that need to track the created fragments
(to log them in a recording database, for example)

https://bugzilla.gnome.org/show_bug.cgi?id=750108
2015-10-03 00:52:23 +10:00
Ramiro Polla
f0a47d0b60 splitmuxsink: allow non-video streams to serve as reference
In the absence of a video stream, the first stream will be used as
reference.

https://bugzilla.gnome.org/show_bug.cgi?id=753617
2015-10-03 00:44:58 +10:00
George Kiagiadakis
20754db26f splitmuxsink: initialize mux_start_time properly
mux_start_time refers to the running_time of the buffer
that goes first in the output file. Normally this time is
0, so this variable is initialized to 0 during the state
change to PAUSED.

However, when dealing with dynamic pipelines and starting
a recording while the pipeline has already run for a while,
the running_time of the first buffer is > 0 and this causes
a problem with detecting the end of the first file(s) when
splitting by duration, because the code will later compare
the threshold_time with (last buffer running_time - mux_start_time)
and will get it wrong until mux_start_time advances enough
to make this difference < threshold_time, creating empty files
in the meantime.

https://bugzilla.gnome.org/show_bug.cgi?id=753624
2015-10-03 00:44:01 +10:00
Vineeth T M
bff62bfe13 avidemux: Reverse playback does not consider segment.start
During reverse playback, the media should stop playing at segment.start
This does not happen, and avidemux continues to process data even when
current timestamp is less that segment.start.

https://bugzilla.gnome.org/show_bug.cgi?id=755094
2015-10-02 17:40:18 +03:00
Manasa Athreya
e6a4c81af5 qtdemux: Check multi trex to find track id in mp4 mpeg-dash stream
If stream has more than one trex box which is not matched to actual
track id, it makes qtdemux crashed.

Author : Manasa Athreya (manasa.athreya@lge.com)

https://bugzilla.gnome.org/show_bug.cgi?id=754864
2015-10-02 17:38:57 +03:00
Ravi Kiran K N
b71068e42d smpte: get size, stride info using VideoInfo
Use VideoInfo data to get size stride and
offset, instead of hard coded macros.

https://bugzilla.gnome.org/show_bug.cgi?id=754558
2015-10-02 17:38:01 +03:00
Ravi Kiran K N
61e3c48ad1 smpte: free mask
Free the memory allocated to 'mask' to avoid
memory leak.

https://bugzilla.gnome.org/show_bug.cgi?id=754555
2015-10-02 17:37:25 +03:00
Hyunjun Ko
b814d7ed25 rtpsource: doesn't handle probation and rtp gap in case of sender
https://bugzilla.gnome.org/show_bug.cgi?id=754548
2015-10-02 16:42:36 +03:00
Hyunjun Ko
2b1f52755d rtpmanager: add new on-new-sender-ssrc, on-sender-ssrc-active signals
Allows for applications to get internal source's RTP statistics.
(eg. sender sources for a server/client)

https://bugzilla.gnome.org/show_bug.cgi?id=746747
2015-10-02 16:39:29 +03:00
Luis de Bethencourt
711b035137 goom2k1: use the new audiovisualizer base class
Rebase to have goom using the GstAudioVisualizer base class in
gst-plugins-base/gst-libs/gst/pbutils

https://bugzilla.gnome.org/show_bug.cgi?id=742875
2015-10-01 16:24:46 +01:00
Luis de Bethencourt
17c17fc369 goom: use the new audiovisualizer base class
Rebase to have goom using the GstAudioVisualizer base class in
gst-plugins-base/gst-libs/gst/pbutils

https://bugzilla.gnome.org/show_bug.cgi?id=742875
2015-10-01 16:16:25 +01:00
Thiago Santos
5c7b051b90 deinterleave: implement accept-caps
Avoid using default accept-caps handler that will query downstream
and is more expensive. Just check if the caps is compatible with
the template and check if the channels are the same.
2015-09-30 17:35:33 -03:00
Thiago Santos
b71d9b1783 deinterleave: use the caps query filter
It was being ignored and would lead to wrong results if the
element doing the query would rely on the intersection being made.
2015-09-30 12:48:30 -03:00
Thiago Santos
74c05502f7 deinterleave: implement a caps query handler for the sinkpad
It was missing and apparently code relied on having it there
for not allowing a change in the number of channels
2015-09-30 12:48:30 -03:00
Thiago Santos
ad5bc461c8 deinterleave: fix caps leak
Caps from the pad template are being leaked. In any case it is
from a static pad template and will 'leak' in the end, just doing
the cleanup for the good practice.
2015-09-30 12:47:52 -03:00
Sebastian Dröge
1cd4baa16a matroskademux: Remove leftover assertion from 0.10
We now allocate memory via GstAllocator and as such can handle arbitrary
alignments, not only <= G_MEM_ALIGN.

https://bugzilla.gnome.org/show_bug.cgi?id=755708
2015-09-28 18:03:51 +02:00
Guillaume Marquebielle
35139ee8b7 aacparse: fix uninitialized variables in LOAS config reading
On reading LOAS config, flag v=1 and vA=1 combination can occur, leading to warning
"Spec says "TBD"...". Returning TRUE on this case while parameters 'sample_rate' and
'channels' are pointing to uninitialized values can end on setting random values as
rate and channels on src caps.

https://bugzilla.gnome.org/show_bug.cgi?id=755611
2015-09-26 22:18:26 +10:00
Jan Schmidt
866c86dd37 Fix some compiler warnings when building with G_DISABLE_ASSERT
Touches rtpmanager and gdkpixbufsink
2015-09-26 22:18:26 +10:00
Chris Bass
563ffc0d8f qtdemux: support timed-text subtitle tracks.
https://bugzilla.gnome.org/show_bug.cgi?id=752818
2015-09-26 00:51:42 +02:00
Sebastian Dröge
7046852e7d gst: Don't use deprecated gst_segment_to_position() 2015-09-26 00:12:46 +02:00
Sebastian Dröge
01c0f8723f rtpbin/rtpjitterbuffer/rtspsrc: Add property to set maximum ms between RTCP SR RTP time and last observed RTP time
https://bugzilla.gnome.org/show_bug.cgi?id=755125
2015-09-25 23:55:05 +02:00
Sebastian Dröge
a0ae6b5b5a rtpbin/session: Allow RTCP sync to happen based on capture time or send time
Send time is the previous behaviour and the default, but there are use cases
where you want to synchronize based on the capture time.

https://bugzilla.gnome.org/show_bug.cgi?id=755125
2015-09-25 23:55:00 +02:00
Thibault Saunier
802a270126 smptealpha: Do not set width/height before comparing with old values
Otherwise we end up considering the values did not change and we wrongly
work with the old video format (which will lead to wrong
behaviour/segfaults).

https://bugzilla.gnome.org/show_bug.cgi?id=755621
2015-09-25 14:20:13 +02:00
Sebastian Dröge
d7a0fd82c0 qtdemux: Accumulate segments for edit lists before activating the next segment
eceb2ccc73 broke segment seeks by always
accumulating segments manually when activating a segment. This is only
needed when handling edit lists, not when activating a segment because of a
seek. Do the accumulation when switching edit list segments instead.

This fixes segment seeks again, while keeping edit lists playback working.

https://bugzilla.gnome.org/show_bug.cgi?id=755471
2015-09-24 09:33:33 +02:00
Vikram Fugro
3c2044168d spectrum: send phase values in the GstMessage for Phase info
https://bugzilla.gnome.org/show_bug.cgi?id=755463
2015-09-23 15:41:08 +02:00
Jan Schmidt
e5d53ec7e4 matroska-mux: Don't output a warning on MONO multiview mode. 2015-09-22 00:46:01 +10:00
Sebastian Rasmussen
905295ea34 rtptheoradepay: Fix memory leaks
The same memory leaks were fixed in identical fashion for
vorbisdepay in 06efeff5d9 in 2009.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277
2015-09-20 10:13:38 +02:00
Sebastian Rasmussen
2d7bfc1314 rtp{vorbis,theora}{pay,depay}: Cosmetic cleanup
* use g_list_free_full(), don't iterate elements maually when freeing
* call gst_rtp_*_pay_clear_packet(), don't duplicate its code
* use gst_buffer_unref() to clarify that it is buffers being released,
  instead of refering directly to gst_mini_object_unref()

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277
2015-09-20 10:13:38 +02:00
Sebastian Dröge
869e21bd82 rtp{vorbis,theora}pay: Store headers in the packet buffers lists, not a NULL buffer
https://bugzilla.gnome.org/show_bug.cgi?id=755265
2015-09-20 10:13:38 +02:00
Eunhae Choi
dc74d744c3 avidemux: Fix taglist leak
gst_tag_list_insert() does not take ownership of the inserted taglist.

https://bugzilla.gnome.org/show_bug.cgi?id=755138
2015-09-17 12:03:08 +02:00
Jan Schmidt
c919548e2c aacparse: Skip LOAS AAC until a valid config is seen.
It's normal when dropping into the middle of a stream to
not always have the config available immediately, so skip LOAS
until a valid config is seen without either setting invalid
caps or erroring out.

https://bugzilla.gnome.org/show_bug.cgi?id=751386
2015-09-16 20:51:44 +10:00
Mark Nauwelaerts
b7b244f356 rtpjitterbuffer: reset just a bit more upon flush_stop 2015-09-13 15:42:06 +02:00
Mark Nauwelaerts
1e7a3473fd rtpjitterbuffer: remove dead struct member 2015-09-13 15:41:03 +02:00
Vineeth TM
2a7ba2955c multiudpsink: fix GError memory leak when hostname resolution fails
https://bugzilla.gnome.org/show_bug.cgi?id=754869
2015-09-11 10:18:14 +01:00
Thiago Santos
f9c7dc2797 matroskamux: drop HEADER flag from output buffers
Drop HEADER flag from output buffers if they are not indeed
headers.

Fixes resending of headers in tcp connection handling

https://bugzilla.gnome.org/show_bug.cgi?id=754768
2015-09-10 16:28:48 -03:00
Tim-Philipp Müller
99a6f8207f matroskamux: fix matroskamux ! matroskademux
Don't carry over DISCONT flags from the input buffers to the
output buffer, or the demuxer might reset its state when it
receives the first data buffer just after parsing the simple
block header, and then expect sane data to follow.
Fixes matroskamux ! demux erroring out.

https://bugzilla.gnome.org/show_bug.cgi?id=754768
https://bugzilla.gnome.org/show_bug.cgi?id=657805
2015-09-10 16:05:53 +01:00
Martin Kelly
00a938f134 rtsp: fix small README typo
https://bugzilla.gnome.org/show_bug.cgi?id=754807
2015-09-10 08:43:20 +01:00
Tim-Philipp Müller
fcdb79ef7b wavpackparse: set both pts and dts so baseparse doesn't make up wrong dts after seeks
https://bugzilla.gnome.org/show_bug.cgi?id=752106
2015-09-06 16:36:47 +01:00
Tim-Philipp Müller
0d88f27108 flacparse: set both pts and dts so baseparse doesn't make up wrong dts after a seek
flac contains the sample offset in the frame header, so after a seek
without index flacparse will know the exact position we landed on and
timestamp buffers accordingly. It only set the pts though, which means
the baseparse-set dts which was set to the seek position prevails, and
since the seek was based on an estimate, there's likely a discrepancy
between where we wanted to land and where we did land, so from here on
that dts/pts difference will be maintained, with dts possibly multiple
seconds ahead of pts, which is just wrong. The easiest way to fix this
is to just set both pts and dts based on the sample offset, but perhaps
parsed audio should just not have dts set at all.

https://bugzilla.gnome.org/show_bug.cgi?id=752106
2015-09-06 16:36:44 +01:00
George Chriss
1afb988256 flvmux: Make the element count in arrays not include end
One-line removal of tags_written++

This should fix rtmp output to crtmpserver, and hopefully
noone is expecting that the element count includes the end
element, as different bits of documentation say different
things about whether it should or not.

https://bugzilla.gnome.org/show_bug.cgi?id=661624
2015-09-05 23:45:37 +10:00
Jan Schmidt
db2967125b flvmux: Store incoming bitrate tags and send in the metadata
Apparently the Microsoft Azure RTMP server requires that the
videodatarate and audiodatarate metadata be provided, so
set those, even if it's to 0. Use the actual input bitrate
tags if available.
2015-09-05 23:45:37 +10:00
Jan Schmidt
b38e24995b rtspsrc: Don't parse key data more than needed.
When an auxilliary streams are present in the SDP media,
there's no need to re-parse the SDP attributes multiple
times.
2015-09-05 23:44:51 +10:00
Jan Schmidt
fe4ed1d1df rtspsrc: Fix SRTP + RTX, auth access, a leak, and an invalid memory access.
In parse_keymgmt(), don't mutate the input string that's been passed
as const, especially since we might need the original value again if
the same key info applies to multiple streams (RTX, for example).

When a resource is 404, and we have auth info - retry with the auth
info the same as if we had receive unauthorised, in case the resource
isn't even visible until credentials are supplied.

Fix a memory leak handling Mikey data.

When generating a random keystring, don't overrun the 30 byte
buffer by generating 32 bytes into it.
2015-09-05 23:44:51 +10:00
Sebastian Dröge
50e9cc7f04 udpsrc: Fix build with GLib < 2.44
G_IO_ERROR_CONNECTION_CLOSED was added in 2.44.
2015-09-04 15:18:05 +03:00
Sebastian Dröge
89137fc136 udpsrc: Ignore G_IO_ERROR_CONNECTION_CLOSED when receiving data
This happens on Windows if we use the same socket for sending packets,
and the remote sends ICMP port/host unreachable messages.

https://bugzilla.gnome.org/show_bug.cgi?id=754534
2015-09-04 12:01:52 +03:00
Sebastian Dröge
f0ca2f2ecb rtpvorbis/theoradepay: Fix handling of fragmented packets
This was broken in b1089fb520 by not considering the full packet length of a
fragmented packet but only the length of the first one.

https://bugzilla.gnome.org/show_bug.cgi?id=754417
2015-09-02 21:13:46 +03:00
Olivier Crête
dad751644e dtmfsrc: Reply to latency query 2015-09-01 15:49:07 -04:00
Jan Alexander Steffens (heftig)
3f8efd8af8 matroskademux: Align raw video frames to 32 bytes
Outputting unaligned video frames causes videoscale et al to
crash when attempting SIMD-accelerated conversion.

https://bugzilla.gnome.org/show_bug.cgi?id=736965
2015-08-31 14:35:59 +03:00
Stefan Sauer
6a8194e121 level: fix level calculations for mutliple channels
This was broken with 7b90bf3215.
2015-08-27 10:16:38 +02:00
Ravi Kiran K N
cac239ab89 smpte: Fix memory leak
In gst_smpte_collected(), check upfront if input formats are same
or not. This avoids allocation of in1 and in2 buffers and
subsequent memory leak when input formats do not match.

https://bugzilla.gnome.org/show_bug.cgi?id=754153
2015-08-27 11:13:43 +03:00
Vineeth TM
ba8cda54f4 rtspsrc: Trivial fix to check correct condition
When checking for describe method, because of missing parentheses, wrong
condition is being checked, which will result in wrong behavior.

https://bugzilla.gnome.org/show_bug.cgi?id=753912
2015-08-21 11:06:57 +03:00
Vineeth TM
77c9e2cd4d matroska: read: fix tag list memory leak
gst_toc_entry_merge_tags makes a new ref of the taglist, so it should
be unref'ed as soon as the tags are merged to the tocentry

https://bugzilla.gnome.org/show_bug.cgi?id=753904
2015-08-21 10:22:54 +03:00
Tim-Philipp Müller
29afa75858 multifilesrc: fix regression with starting from index set via index property
When we haven't started yet, set the start_index when we set the index property,
so that we start at the right index position after the initial seek. The index
property was never really meant to be for writing, but it used to work, so let's
support it for backwards compatibility.

https://bugzilla.gnome.org/show_bug.cgi?id=739472
2015-08-18 13:17:34 +01:00
Alex Ashley
5d99d0dfa0 qtdemux: fix offset calculation when parsing CENC aux info
Commit 7d7e54ce68 added support for
DASH common encryption, however commit
bb336840c0 that went onto master
shortly before the CENC commit caused the calculation of the CENC
aux info offset to be incorrect.

The base_offset was being added if present, but if the base_offset
is relative to the start of the moof, the offset was being added twice.
The correct approach is to calculate the offset from the start of the
moof and use that offset when parsing the CENC aux info.
2015-08-18 11:48:03 +01:00
Hyunjun Ko
38d269f80d rtp: copy metadata in the (de)payloaders which is missed before
https://bugzilla.gnome.org/show_bug.cgi?id=753706
2015-08-17 14:12:50 +02:00
Thiago Santos
5838940681 y4mencode: fix gst-launch version in documentation 2015-08-16 14:30:57 -03:00
Thiago Santos
a1aa942acf audioencoders: use template subset check for accept-caps
It is faster than doing a query that propagates downstream and
should be enough

Elements: speexenc, wavpackenc, mulawenc, alawenc
2015-08-16 14:30:57 -03:00
Thiago Santos
1b27badcfd videoencoders: use template subset check for accept-caps
It is faster than doing a query that propagates downstream and
should be enough

Elements: jpegenc, pngenc, vp8enc, vp9enc, y4menc
2015-08-16 14:30:57 -03:00
Tim-Philipp Müller
a39bebb5fe mpegaudioparse: use new baseparse API to fix tag handling
https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-08-16 17:21:24 +01:00
Olivier Crête
b1dfe209c2 audioparsers: use new base parse API to fix tag handling
https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-08-16 17:02:19 +01:00
Tim-Philipp Müller
a042a98159 flacparse: use new baseparse API and fix tag handling
https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-08-16 16:33:55 +01:00
Sebastian Dröge
e9aa4c7467 qtdemux: Use signed integer type to be able to check for negative subtraction results
CID 1315829
2015-08-16 13:04:02 +02:00
Luis de Bethencourt
1aee15050c rtpvorbisdepay: remove dead code
payload_buffer must be NULL in ignore_reserved. Check will always be false.

Introduced by b1089fb520

CID #1316476
2015-08-16 11:52:44 +01:00
Thiago Santos
1328289474 alawenc: port to AudioEncoder base class 2015-08-15 22:46:46 -03:00
Thiago Santos
65676c22ee audiodecoders: use default pad accept-caps handling
Avoids useless check of downstream caps when handling an
accept-caps query

Elements: flacdec, speexdec, wavpackdec, mulawdec, alawdec
2015-08-15 11:46:34 -03:00
Thiago Santos
65d2af6462 alawdec: make error handling a bit nicer
Print the element along with the debug to make it easier to trace
the failures
2015-08-15 11:31:04 -03:00
Thiago Santos
7ab3178cc4 alawdec: port to audiodecoder base class
mulawdec was already ported, alawdec was left behind.
2015-08-15 11:06:02 -03:00
Thiago Santos
41a4b68390 qtdemux: only look for more samples in moofs in pull-mode
For playback of some fragmented formats with qtdemux it will
try to look for the next moof after finishing one but it is only
possible for pull-mode. For playback of streaming fragmented formats
such as DASH it should just not try to look for another moof but
instead wait for more data.

https://bugzilla.gnome.org/show_bug.cgi?id=752602

https://bugzilla.gnome.org/show_bug.cgi?id=752603
2015-08-15 11:06:02 -03:00
Sebastian Dröge
64b06d1829 dcaparse: Don't look for a second syncword
There are streams out there that consistently contain garbage between
every frame so we never ever find a second consecutive syncword.

See https://bugzilla.gnome.org/show_bug.cgi?id=738237
2015-08-15 13:00:06 +02:00
Thiago Santos
9523fb23ed audioparsers: enable accept-template flag
Do a quick check with the pad template caps as it is enough. Users
should have figured the appropriate full caps on a previous caps query

https://bugzilla.gnome.org/show_bug.cgi?id=753623
2015-08-14 13:42:27 -03:00
George Kiagiadakis
e2f2f087ec rtspsrc: send the User-Agent header
Sometimes it is useful to know this information on the
server side. Other popular implementations (vlc, ffmpeg, ...)
also send this header on every message.

This includes a new "user-agent" property that the user
can set to use a custom User-Agent string. The default
is "GStreamer/<version>"

https://bugzilla.gnome.org/show_bug.cgi?id=750101
2015-08-14 15:59:06 +02:00
George Kiagiadakis
af03341e26 rtspsrc: wrap gst_rtsp_message_init_request in a local function
This will allow adding common request initialization, like the
user agent string, in just one place.
2015-08-14 15:59:06 +02:00
Prashant Gotarne
0671ea85af audioecho: make sure buffer gets reallocated if max_delay changes
https://bugzilla.gnome.org/show_bug.cgi?id=753490
2015-08-14 11:50:22 +01:00
Ramiro Polla
23b5a34675 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
2015-08-14 12:38:32 +02:00
Sebastian Dröge
3ede3105d6 goom: Rename get_type() function of base class to prevent symbol conflicts
This is a problem when statically linking.
2015-08-14 09:21:25 +02:00
Sebastian Dröge
68a9209408 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
2015-08-13 16:45:16 +02:00
Ravi Kiran K N
6eee26b24b matroska: Remove unused variable
https://bugzilla.gnome.org/show_bug.cgi?id=753556
2015-08-13 14:11:12 +01:00
Sebastian Dröge
b1089fb520 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
2015-08-11 12:47:23 +02:00
Thiago Santos
288b0bbb38 qtdemux: fix small typo in comment 2015-08-10 19:11:17 -03:00
Nicolas Dufresne
109995707e goom2k1/doc: Fixup previous commit 2015-08-10 16:19:18 -04:00
Nicolas Dufresne
18c43aa845 goom2k1/doc: Use GstGoom2k1 namespace
The doc generator isn't happy when we have class name clash. Simply
use it's own namespace.
2015-08-10 15:55:19 -04:00
Prashant Gotarne
cde1f12ad3 audioecho: removed unused variable in set_property
unused local variable 'delay' is removed.

https://bugzilla.gnome.org/show_bug.cgi?id=753450
2015-08-10 13:34:11 +01:00
Tim-Philipp Müller
604cc2a548 qtdemux: fix suboptimal queue iteration code 2015-08-10 12:45:50 +01:00
Tim-Philipp Müller
0fbf5f3d9e qtdemux: don't use glib 2.44-only API 2015-08-10 12:32:23 +01:00
Alex Ashley
7d7e54ce68 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
2015-08-10 12:32:17 +01:00
Hyunjun Ko
9c5c16eb57 rtph264depay: checking if depay has sps/pps nals before insertion
https://bugzilla.gnome.org/show_bug.cgi?id=753430
2015-08-10 10:49:02 +02:00
Tim-Philipp Müller
a95c761fde matroskamux: fix outdated comment
The default behaviour was changed in the 0.10 -> 1.x
transition, but the comment was not updated.
2015-08-08 16:44:49 +01:00
Sebastian Dröge
c8551b6285 rtptheorapay: If flushing a packet failed, go out of the loop immediately 2015-08-08 17:43:03 +02:00
Sebastian Dröge
4957cc7459 rtpvorbispay: If flushing a packet failed, go out of the loop immediately 2015-08-08 17:43:03 +02:00
Sebastian Dröge
983f57dc7d 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.
2015-08-08 17:43:03 +02:00
George Kiagiadakis
2e590a32eb 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
2015-08-07 10:15:05 +02:00
Hyunjun Ko
b0d6020862 rtprtxsend: print valid type where guint32 is expected
https://bugzilla.gnome.org/show_bug.cgi?id=746445
2015-08-06 01:39:43 -03:00
Hyunjun Ko
5a17572119 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
2015-08-06 01:38:43 -03:00
Thiago Santos
e0878d6325 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
2015-08-05 18:12:45 -03:00
Thiago Santos
bb336840c0 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
2015-08-05 18:12:45 -03:00
Glen Diener
cd57697a2c matroskademux: Preserve forward referenced track tags
https://bugzilla.gnome.org/show_bug.cgi?id=752850
2015-08-05 16:46:33 -04:00
Sebastian Dröge
c9ea95481c 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
2015-08-04 21:00:31 +03:00
Sebastian Dröge
9ae316974d rtph264depay: Put the profile and level into the caps 2015-08-04 12:45:06 +03:00
Sebastian Dröge
8dda570e47 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.
2015-08-04 12:45:06 +03:00
Sebastian Dröge
e0c124f76d 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).
2015-08-04 12:45:06 +03:00
Thiago Santos
72212198c7 splitmuxsink: remove extra \n at debug message 2015-08-03 13:46:16 -03:00
Thiago Santos
a930a1364a 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.
2015-08-03 13:46:16 -03:00
Sebastian Dröge
ef7863355c 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.
2015-08-03 18:24:18 +03:00
Vineeth TM
cf19525d5c 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
2015-07-30 15:51:25 +03:00
Vineeth TM
969bcf25a1 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
2015-07-30 12:20:19 +01:00
Sebastian Dröge
39a90710b7 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
2015-07-29 14:31:49 +01:00
Jan Schmidt
a0182dd943 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
2015-07-29 23:03:30 +10:00
Jan Schmidt
e7ec32801a 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.
2015-07-29 23:03:18 +10:00
Vineeth TM
72b86ae868 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
2015-07-29 09:14:31 +01:00
Manasa Athreya
e6381ef285 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
2015-07-27 19:06:43 +01:00
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
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
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
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
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
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