Julien Isorce [Fri, 1 Nov 2013 17:09:42 +0000 (17:09 +0000)]
tests/check: add rtpcollision::test_rtx_ssrc_collision unit test
check that rtxrtpsend changes its retransmission ssrc when
collision happens
George Kiagiadakis [Wed, 6 Nov 2013 10:34:13 +0000 (12:34 +0200)]
tests/check: add rtprtx::test_rtxreceive_data_reconstruction
This unit test verifies that retransmitted rtp packets coming out
of rtprtxreceive are the same as the original ones.
George Kiagiadakis [Tue, 5 Nov 2013 07:33:51 +0000 (09:33 +0200)]
rtprtxsend: use a realistic limit for the value of max-size-packets
G_MAXINT16 is chosen because if the queue contains more than
G_MAXINT16 packets, seqnum comparison will not work properly.
George Kiagiadakis [Mon, 4 Nov 2013 18:05:03 +0000 (20:05 +0200)]
rtprtxsend: use a GSequence to implement the buffer queue
This has the advantage that searching the queue to find the
buffer with the requested seqnum is done with binary search.
George Kiagiadakis [Mon, 4 Nov 2013 16:38:24 +0000 (18:38 +0200)]
rtprtxsend: retransmit packets in the same order as the rtx requests
George Kiagiadakis [Sat, 2 Nov 2013 17:56:44 +0000 (19:56 +0200)]
tests/check: Add unit test for rtxsend's max_size_time property
George Kiagiadakis [Tue, 29 Oct 2013 17:27:00 +0000 (18:27 +0100)]
rtprtxsend: Handle the max_size_time property
This property allows you to specify the amount of buffers
to keep in the retransmission queue expressed as time (ms)
instead of buffer count (which is the max_size_buffers property).
George Kiagiadakis [Sat, 2 Nov 2013 13:21:08 +0000 (15:21 +0200)]
rtprtxsend: keep important buffer information in a private structure
This is to avoid mapping a buffer every time we need to read a seqnum
or a timestamp.
George Kiagiadakis [Fri, 1 Nov 2013 10:58:47 +0000 (11:58 +0100)]
tests/check: Add rtprtx::test_rtxsender_packet_retention
This unit test verifies that the rtxsend element correctly maintains
a buffer of already transmitted rtp packets and that it can
re-transmit all of them correctly on demand. It also verifies
that the limit of this buffer (max-size-packets property) is respected.
Julien Isorce [Fri, 1 Nov 2013 16:22:13 +0000 (16:22 +0000)]
tests/check: add rtprtx::test_drop_multiple_sender unit test
Several senders / one receiver
Similar than test_drop_one_sender but with multiple senders
mixed through the funnel element.
It drops some packets and checks that they are retransmited
correctly.
Julien Isorce [Fri, 1 Nov 2013 16:21:00 +0000 (16:21 +0000)]
tests/check: add rtprtx::test_drop_one_sender unit test
Test for one sender / one receiver
Build the pipeline
videotestsrc ! rtpvrawpay ! rtprtxsend ! rtprtxreceive ! fakesink
and drop some buffers between rtprtxsend and rtprtxreceive
Then it checks that every dropped packet has been re-sent.
It also checks that not too much requests has been sent.
Julien Isorce [Fri, 1 Nov 2013 16:17:51 +0000 (16:17 +0000)]
tests/check: add rtprtx::test_push_forward_seq
add simple unit test that manually push buffers
in rtprtxsend connected to rtprtxreceive.
Drops some buffers and make sure they are retransmisted.
Julien Isorce [Fri, 1 Nov 2013 15:52:03 +0000 (15:52 +0000)]
rtpmanager: add new rtprtxsend / rtprtxreceive elements
The purpose of the sender RTX object is to keep a history
of RTP packets up to a configurable limit (in time). It will
listen for custom retransmission events from downstream. When
it receives a request for retransmission, it will look up the
requested seqnum in its list of stored packets. If the packet
is available, it will create a RTX packet according to RFC 4588
and send this as an auxiliary stream.
The receiver will listen to the custom retransmission events
from the downstream jitterbuffer and will remember the SSRC1
of the stream and seqnum that was requested. When it sees a
packet with one of the stored seqnum, it associates the SSRC2
of the stream with the SSRC1 of the master stream. From then
on it knows that SSRC2 is the retransmission stream of SSRC1.
This algorithm is stated in RFC 4588. For this algorithm to
work, RFC4588 also states that no two pending retransmission
requests can exist for the same seqnum and different SSRCs or
else it would be impossible to associate the retransmission with
the original requester SSRC.
When the RTX receiver has associated the retransmission packets,
it can depayload and forward them to the source pad of the element.
RTX is SSRC-multiplexed
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711084
Julien Isorce [Tue, 5 Nov 2013 16:36:46 +0000 (16:36 +0000)]
doc: add design for rtp retransmission
Describe how rtprtxsend and rtprtxreceive generally work
but also how the association algorithm is implemented.
Reynaldo H. Verdejo Pinochet [Thu, 2 Jan 2014 23:23:05 +0000 (20:23 -0300)]
souphttpsrc: use status code macro instead of 407
Rest of the code is using the _PROXY_AUTHENTICATION_REQUIRED
macro too. Easier to understand if you don't recall HTTP
error codes by heart.
Reynaldo H. Verdejo Pinochet [Wed, 1 Jan 2014 00:31:43 +0000 (21:31 -0300)]
shout2send: change audio_format field to format
This element and the underlying libshout2 library
can handle video media files too. The code already
handles video/webm so the name gets confusing. Also
add and use DEFAULT_FORMAT macro Instead of hardwiring
SHOUT_FORMAT_VORBIS at init
https://bugzilla.gnome.org/show_bug.cgi?id=721342
Reynaldo H. Verdejo Pinochet [Tue, 31 Dec 2013 23:09:29 +0000 (20:09 -0300)]
shout2send: clarify meaning of the URL prop
https://bugzilla.gnome.org/show_bug.cgi?id=721342
Reynaldo H. Verdejo Pinochet [Fri, 27 Dec 2013 15:27:32 +0000 (12:27 -0300)]
shout2send: docs, add a sample pipeline
And finish adding shout2send to the docs while at it
https://bugzilla.gnome.org/show_bug.cgi?id=721342
Reynaldo H. Verdejo Pinochet [Tue, 31 Dec 2013 18:00:22 +0000 (15:00 -0300)]
gdkpixbufoverlay: remove spurious @see_also
Matthieu Bouron [Fri, 6 Dec 2013 17:08:54 +0000 (17:08 +0000)]
deinterlace: support any video formats and any caps features if deinterlace mode allows it
https://bugzilla.gnome.org/show_bug.cgi?id=719636
Sebastian Rasmussen [Tue, 31 Dec 2013 12:31:52 +0000 (13:31 +0100)]
v4l2: Handle v4l2_ioctl() errors even in error handling
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=721268
Jeremy Huddleston Sequoia [Wed, 1 Jan 2014 20:11:43 +0000 (12:11 -0800)]
osxvideo: unifdef -DRUN_NS_APP_THREAD
Jeremy Huddleston Sequoia [Wed, 1 Jan 2014 20:10:01 +0000 (12:10 -0800)]
osxvideo: Assume SDK and deployment target are at least Snow Leopard
Jeremy Huddleston Sequoia [Wed, 1 Jan 2014 20:23:50 +0000 (12:23 -0800)]
configure: Disable osxvideo on Leopard and earlier
This also moves the "other platforms" check in OS X video to before the
variable is read
https://bugzilla.gnome.org/show_bug.cgi?id=721245
Wim Taymans [Tue, 31 Dec 2013 13:57:27 +0000 (14:57 +0100)]
tests: add AUX receiver unit test
Wim Taymans [Tue, 31 Dec 2013 12:20:01 +0000 (13:20 +0100)]
tests: improve rtpbin test
Wim Taymans [Tue, 31 Dec 2013 12:16:46 +0000 (13:16 +0100)]
rtpbin: add some docs about AUX elements
Wim Taymans [Tue, 31 Dec 2013 12:01:22 +0000 (13:01 +0100)]
tests: add AUX sender unit test
Wim Taymans [Tue, 31 Dec 2013 11:31:25 +0000 (12:31 +0100)]
rtpbin: add support for AUX sender and receiver
AUX elements are elements that can be inserted into the rtpbin
pipeline right before or after 1 or more session elements.
The AUX elements are essential for implementing functionality such
as error correction (FEC) and retransmission (RTX).
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711087
Wim Taymans [Tue, 31 Dec 2013 11:22:39 +0000 (12:22 +0100)]
tests: add decoder test
Wim Taymans [Mon, 30 Dec 2013 16:36:42 +0000 (17:36 +0100)]
rtpbin: make request_element method internally
We can use the same method to create encoder and decoder elements, they
are just internal elements that we create.
Stéphane Cerveau [Tue, 31 Dec 2013 09:25:28 +0000 (10:25 +0100)]
wavparse: Skip id3 tag
Skip id3 tag during wav parse.
https://bugzilla.gnome.org/show_bug.cgi?id=721241
Sebastian Dröge [Tue, 31 Dec 2013 09:10:05 +0000 (10:10 +0100)]
osx: Make OSX version checks more consistent
And especially also consider update versions, e.g. 10.5 with updates
will be 1051 or similar and thus bigger than MAC_OS_X_VERSION_10_5 but
still won't have the API we want to use.
Jeremy Huddleston [Tue, 31 Dec 2013 09:07:22 +0000 (10:07 +0100)]
osxvideosink: Fix build on updated OS X Leopard
https://bugzilla.gnome.org/show_bug.cgi?id=721245
Edward Hervey [Mon, 30 Dec 2013 16:23:22 +0000 (17:23 +0100)]
avimux: Add missing break
I guess no-one noticed we no longer could mux WMV3 ...
COVERITY CID 1139759
Edward Hervey [Mon, 30 Dec 2013 16:20:37 +0000 (17:20 +0100)]
rtpvrawpay: Add missing break
COVERITY CID 1139762
Wim Taymans [Mon, 30 Dec 2013 16:00:45 +0000 (17:00 +0100)]
rtpsession: internal-ssrc is no longer deprecated
Wim Taymans [Mon, 30 Dec 2013 15:59:20 +0000 (16:59 +0100)]
rtpbin: add Since tags
Wim Taymans [Mon, 30 Dec 2013 15:52:28 +0000 (16:52 +0100)]
rtpbin: add signal for new jitterbuffer
Emit a signal when a new jitterbuffer is created so that the app can
have a chance to configure it.
Wim Taymans [Mon, 30 Dec 2013 15:28:57 +0000 (16:28 +0100)]
rtpbin: handle multiple encoder instances
Keep track of elements that are added to multiple sessions and make sure
we only add them to the rtpbin once and that we clean them when no
session refers to them anymore.
Wim Taymans [Mon, 30 Dec 2013 14:16:09 +0000 (15:16 +0100)]
tests: add unit test for encoder element
Wim Taymans [Mon, 30 Dec 2013 14:15:43 +0000 (15:15 +0100)]
rtpbin: fix memory leaks
Wim Taymans [Mon, 30 Dec 2013 14:03:34 +0000 (15:03 +0100)]
tests: fix leak
Wim Taymans [Mon, 30 Dec 2013 14:00:50 +0000 (15:00 +0100)]
rtpbin: expect the pads on the encoders
Don't use request pads for the encoder elements, the signal handler
should request the pads and make sure they are available with the right
name.
Wim Taymans [Mon, 30 Dec 2013 13:56:07 +0000 (14:56 +0100)]
rtpbin: request-rtp-encoder are no action signals
The request-rtp-encoder signals are not action signals so mark them
correctly and use an accumulator to collect the result value.
Stefan Sauer [Mon, 30 Dec 2013 13:36:45 +0000 (14:36 +0100)]
wavparse: emit midi-base-note tag from data in 'smpl' chunk
Add parsing of the 'smpl' chunk. Right now we only grab the midi-base-note and
emit it as a tag.
George Kiagiadakis [Thu, 26 Dec 2013 10:05:19 +0000 (12:05 +0200)]
gstrtpsession: suggest upstream to use the new "internal-ssrc" after a collision
When a collision is found on the internal ssrc, we have to change it.
Ideally, we want also the payloader upstream to follow this change and use
the new internal ssrc. Ideally we want this condition to be always met:
if there is one payloader sending on this session, its ssrc should match the
internal ssrc.
George Kiagiadakis [Thu, 26 Dec 2013 09:04:29 +0000 (11:04 +0200)]
rtpsession: allow setting internal-ssrc again
Edward Hervey [Mon, 30 Dec 2013 12:31:45 +0000 (13:31 +0100)]
y4mencode: Remove dead code
set/get property isn't used
Edward Hervey [Mon, 30 Dec 2013 12:30:24 +0000 (13:30 +0100)]
rtpqcelpdepay: Remove uneeded variable
Aleix Conchillo Flaqué [Thu, 5 Dec 2013 23:53:52 +0000 (15:53 -0800)]
rtpbin: allow dynamic RTP/RTCP encoders/decoders
* gst/rtpmanager/gstrtpbin.[ch]: four new action signals have been
added (request-rtp-encoder, request-rtp-decoder, request-rtcp-encoder
and request-rtcp-decoder). The user will be able to provide encoders
or decoders dynamically. The encoders must follow the srtpenc API and
the decoders the srtpdec API. Having separate signals for RTP and RTCP
allows the user to use different encoders/decoders or provide the same
one (e.g. that would be the case for srtpenc).
Also, rtpbin now allows application/x-srtp in its pads.
https://bugzilla.gnome.org/show_bug.cgi?id=719938
Wim Taymans [Fri, 27 Dec 2013 15:51:32 +0000 (16:51 +0100)]
rtpjitterbuffer: dynamically recalculate RTX parameters
Use the round-trip-time and average jitter to dynamically calculate the
retransmission interval and expected packet arrival time.
Based on patches from Torrie Fischer <torrie.fischer@collabora.co.uk>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711412
Wim Taymans [Fri, 27 Dec 2013 15:50:52 +0000 (16:50 +0100)]
rtpjitterbuffer: calculate average jitter
Wim Taymans [Fri, 27 Dec 2013 15:48:48 +0000 (16:48 +0100)]
rtpsession: use RTT from the Retransmission event
Place the estimated RTT in the Retransmission event and let the session
manager use that instead of the hardcoded value.
Wim Taymans [Fri, 27 Dec 2013 14:57:39 +0000 (15:57 +0100)]
jitterbuffer: take more accurate running-time for NACK
Don't use the current time calculated from the tmieout loop for when we
last scheduled the NACK because it might be unscheduled because of a max
packet misorder and then we don't accurately calculate the current time.
Instead, take the current element running time using the clock.
Sebastian Dröge [Mon, 30 Dec 2013 10:06:38 +0000 (11:06 +0100)]
wavpackdec: Send a CAPS event in the unit test
Thiago Santos [Fri, 27 Dec 2013 05:14:02 +0000 (02:14 -0300)]
qtdemux: improve mss_mode/fragmented special handling
Make it clear what should be handled purely by mss mode:
1) Expose the streams on the first moof as there are no moov atoms
2) Properly cleanup streams on flushes
Add a note about the meaning of upstream_newsegment and mss_mode
for future reference.
Make all other special fragment handling shared for both dash
and mss streams.
Thiago Santos [Thu, 12 Dec 2013 13:50:27 +0000 (10:50 -0300)]
qtdemux: drain the adapter before pushing EOS
In a fragmented scenario, qtdemux is operating in push mode
and it gets a fragmented buffer. While processing its data
downstream gets unlinked (or a input-selector changes its
active pad and returns not-linked). Qtdemux stops processing
this fragment and returns not-linked upstream, leaving the
remaining data in its adapter.
When it gets an EOS it should make sure that all the data it
had received is pushed before pushing EOS.
Reynaldo H. Verdejo Pinochet [Fri, 27 Dec 2013 02:21:47 +0000 (23:21 -0300)]
shout2send: drop IP only requirement for _set_host()
libshout2 (we require > 2.0 at config time) supports
both IP and hostname for _set_host(). Dropped an
outdated FIXME regarding this limitation, adjusted
some comments and changed the param blurb to reflect
this too.
Reynaldo H. Verdejo Pinochet [Fri, 27 Dec 2013 00:43:34 +0000 (21:43 -0300)]
shout2send: Retarget FIXME to 2.0
Wim Taymans [Thu, 26 Dec 2013 10:21:36 +0000 (11:21 +0100)]
rtspsrc: use aggregate control for PLAY/PAUSE/TEARDOWN
Use the aggregate control instead of the original request url to perform
PAUSE/PLAY and TEARDOWN.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=721003
Sebastian Dröge [Tue, 24 Dec 2013 13:40:25 +0000 (14:40 +0100)]
rndbuffersize: Proxy CAPS, ALLOCATION, SCHEDULING and srcpad events properly
Nicola Murino [Mon, 23 Dec 2013 23:43:39 +0000 (00:43 +0100)]
matroskamux: adpcm max block align is 8192
Brendan Long [Mon, 23 Dec 2013 18:23:27 +0000 (12:23 -0600)]
vp9dec: Require vpx >= 1.3.0 for building vp9dec and vp9enc
Previous versions did not have a stable bitstream for VP9.
https://bugzilla.gnome.org/show_bug.cgi?id=720986
Sebastian Dröge [Mon, 23 Dec 2013 14:46:48 +0000 (15:46 +0100)]
matroskamux: Use correct codec id for ADPCM/DVI
Sebastian Dröge [Mon, 23 Dec 2013 14:44:30 +0000 (15:44 +0100)]
matroskademux: Check for the correct size of codec_data in the ACM case
Nicola Murino [Sat, 14 Jan 2012 18:58:17 +0000 (19:58 +0100)]
matroskamux: basic adpcm support
https://bugzilla.gnome.org/show_bug.cgi?id=664339
Sebastian Dröge [Fri, 20 Dec 2013 10:45:38 +0000 (11:45 +0100)]
qtdemux: Fix calcuation of descriptor length
https://bugzilla.gnome.org/show_bug.cgi?id=720813
Tim-Philipp Müller [Sun, 22 Dec 2013 22:33:39 +0000 (22:33 +0000)]
Automatic update of common submodule
From dbedaa0 to d48bed3
Tim-Philipp Müller [Sun, 22 Dec 2013 21:56:03 +0000 (21:56 +0000)]
po: set gettext domain in Makevars so we don't have to patch the generated Makefile.in.in
https://bugzilla.gnome.org/show_bug.cgi?id=705455
Tim-Philipp Müller [Thu, 19 Dec 2013 16:50:10 +0000 (16:50 +0000)]
udpsrc: on receive error only unmap and unref buffer if one was alloced and mapped
coverity CID 1139866.
Tim-Philipp Müller [Thu, 19 Dec 2013 12:47:22 +0000 (12:47 +0000)]
multiudpsink: fix misleading comment
Those are not allocated on the stack.
Sebastian Dröge [Tue, 17 Dec 2013 17:28:25 +0000 (18:28 +0100)]
vpx: Mark VP9 support as non-experimental
There was a libvpx release with VP9 support now and the bitstream
is frozen too.
Todd Agulnick [Mon, 16 Dec 2013 05:04:11 +0000 (21:04 -0800)]
Some compiler warning fixes to satisfy XCode compiler
https://bugzilla.gnome.org/show_bug.cgi?id=720513
Sebastian Dröge [Mon, 16 Dec 2013 15:17:07 +0000 (16:17 +0100)]
id3v2mux: Set picture type in the APIC frames
Sebastian Dröge [Mon, 16 Dec 2013 15:14:52 +0000 (16:14 +0100)]
id3v2mux: Set image-description from the info struct, not the caps
Sebastian Dröge [Mon, 16 Dec 2013 09:02:37 +0000 (10:02 +0100)]
wavpackparse: Post AUDIO_CODEC tag
Sebastian Dröge [Mon, 16 Dec 2013 09:00:37 +0000 (10:00 +0100)]
sbcparse: Post AUDIO_CODEC tag
Sebastian Dröge [Mon, 16 Dec 2013 08:58:31 +0000 (09:58 +0100)]
flacparse: Post AUDIO_CODEC tag
https://bugzilla.gnome.org/show_bug.cgi?id=720512
Sebastian Dröge [Mon, 16 Dec 2013 08:56:29 +0000 (09:56 +0100)]
dcaparse: Post AUDIO_CODEC tag
Sebastian Dröge [Mon, 16 Dec 2013 08:54:38 +0000 (09:54 +0100)]
amrparse: Post AUDIO_CODEC tag
Sebastian Dröge [Mon, 16 Dec 2013 08:49:48 +0000 (09:49 +0100)]
ac3parse: Post AUDIO_CODEC tag
Sebastian Dröge [Mon, 16 Dec 2013 08:46:16 +0000 (09:46 +0100)]
aacparse: Post AUDIO_CODEC tag
Sebastian Dröge [Mon, 16 Dec 2013 08:41:14 +0000 (09:41 +0100)]
mpegaudioparse: Use pbutils functionality to create the AUDIO_CODEC tag
Olivier Crête [Fri, 13 Dec 2013 22:36:36 +0000 (17:36 -0500)]
rtpsession: Add error message if the app tries to set the internal-ssrc
Olivier Crête [Fri, 13 Dec 2013 21:08:35 +0000 (16:08 -0500)]
rtpsession: Only count nacks when a nack packet is received
Not when any RTCP feedback packet is.
Olivier Crête [Fri, 13 Dec 2013 04:22:41 +0000 (23:22 -0500)]
tests: Initialize segment in rtpcollision test
Olivier Crête [Fri, 13 Dec 2013 20:57:36 +0000 (15:57 -0500)]
rtpsession: Process PSFB FIR requests which lack the media ssrc
According to RFC 5104 section 4.3.1.2, RTCP PSFB FIR message SHALL
have a media_ssrc field set to 0. The actual media ssrc is in the FCI.
So in that case, we ignore the retained feedback and just let it through
to the rtp_session_process_fir() function which will check for the actual
SSRC inside the FCI.
Fixes a regression introduced by commit
57c27ec3
George Kiagiadakis [Thu, 14 Nov 2013 14:19:29 +0000 (16:19 +0200)]
rtpsession: fix rb blocks disappearing after the first rtcp cycle with multiple senders
Previously, when the session had multiple internal sender SSRCs, it would
issue SR reports with RB blocks only on the first RTCP timeout and afterwards
SR reports would be sent empty. This was because the "generation" number
in RTPSource would increase more than once during the same cycle and afterwards
it would always be greater than the session's generation, which would cause
it to be skipped from being included in RBs.
This commit fixes this problem by:
1) Increasing the RTPSource generation only at the end of each cycle,
which essentially fixes the problem but only when the internal senders
are less than GST_RTCP_MAX_RB_COUNT.
2) Keeping for each RTPSource a set of SSRCs which stores which SSRC's
SR the given RTPSource has been reported in, which also fixes the problem
when the internal senders are more than GST_RTCP_MAX_RB_COUNT. This is
necessary because of the fact that any RTPSource is marked as reported
in itself's SR and makes it impossible to know if it has been reported
in other SRs too or not, and which.
George Kiagiadakis [Thu, 14 Nov 2013 14:23:35 +0000 (16:23 +0200)]
tests/check: add an rtpsession unit test to verify all RBs are included in all SRs, roundrobin
This test checks that when we have multiple internal sender sources
in rtpsession, SRs contain RBs for every other sender source, and that
they are included roundrobin when they exceed ST_RTCP_MAX_RB_COUNT,
which is the max number of RBs that can fit in a SR.
Wim Taymans [Thu, 12 Dec 2013 15:01:10 +0000 (16:01 +0100)]
docs: improve docs
Julien Isorce [Tue, 5 Nov 2013 18:03:48 +0000 (18:03 +0000)]
doc: add design-rtpcollision.txt that explains when GstRTPCollision is created
It also talks about "BYE only the corresponding source, not the whole session."
Julien Isorce [Tue, 5 Nov 2013 12:31:54 +0000 (12:31 +0000)]
tests/check: improve rtpcollision::test_master_ssrc_collision to ensure that a collision does not BYE the whole session
Conflicts:
tests/check/elements/rtpcollision.c
Julien Isorce [Fri, 1 Nov 2013 17:07:57 +0000 (17:07 +0000)]
tests/check: add rtpcollision::test_master_ssrc_collision unit test
It checks the payloader changes its ssrc when collision happens
George Kiagiadakis [Thu, 12 Dec 2013 09:38:43 +0000 (10:38 +0100)]
rtpsession: keep extra stats for scheduling BYE
Keep an extra stats structure for scheduling the BYE packets. When we
decide to schedule BYE, make a copy of the current stats into the
bye_stats. Then while we schedule the BYE, update and use only the
bye_stats. When we finished scheduling the BYE packet, we use the
regular stats again.
George Kiagiadakis [Thu, 12 Dec 2013 09:34:38 +0000 (10:34 +0100)]
rtpsession: when we schedule BYE, only deal with BYE sources
When we are doing the RTCP timeout to schedule BYE packets, don't
generate RTCP for all sources but only for the sources marked as BYE.
George Kiagiadakis [Thu, 12 Dec 2013 09:32:48 +0000 (10:32 +0100)]
rtpsession: reset state after scheduling BYE
After we do RTCP, we are not scheduling bye anymore.
George Kiagiadakis [Thu, 12 Dec 2013 09:31:38 +0000 (10:31 +0100)]
rtpsession: also count NACKS when no signal was pending
George Kiagiadakis [Thu, 12 Dec 2013 09:09:25 +0000 (10:09 +0100)]
session: ignore RTCP packets for the BYE sources
When we are scheduling BYE packets, ignore all RTCP for the sources that
are scheduling a BYE packet. Other sources that are not scheduling BYE
should continue receiving RTCP packets as usual.
Julien Isorce [Mon, 4 Nov 2013 11:48:21 +0000 (11:48 +0000)]
rtpsession: determine if the session is doing point-to-point
In this case T_dither_max is set to 0 according to RFC 4585