ffmpeg 7.1+ has deprecated direct usage of ->sample_fmts in favour of a
call to avcodec_get_supported_config(). Update accordingly.
Change-Id: I0fbecbbcf52c38976f3387e34d816d11e1d09480
(cherry picked from commit 65477ac5d5)
(cherry picked from commit d055526a40)
These strings are not null terminated. Use the correct string length.
Change-Id: I15d7541af0d9cc09401180a7d45edbec9b915aa2
(cherry picked from commit 0a1d658766)
(cherry picked from commit 99415171bd)
under normal circumstances, the rtp sink enters this function first
and updates the reti->local.family var to AF_INET so that when the
function is called for an rtcp sink, it exits the function early.
However, if media is being blocked the rtcp sink is the first to
enter the function and does so with a NULL payload type. this NULL
check is therefore required to prevent a sefgault on the call to `assert`.
closes#1876
Change-Id: Id95cbbc9ae388f50dd51417c351f4a9ba09eb5f0
(cherry picked from commit 8e04c44b86)
(cherry picked from commit 7b9102a703)
using str_init_dup_str doesn't leave enough room for appending encryption
related pieces to the end of the packet when used for, eg, dtmf
injection
closes#1819
Change-Id: Iefae0e04b38f4a3eaaac32ed1ba70c7e3ee8e979
(cherry picked from commit ce66c7b8fa)
(cherry picked from commit 4fd891fcf3)
Use a BIO WRITE callback instead of BIO_read'ing from the BIO after each
operation. This is a more direct way to intercept data that needs to be
sent out.
Implement MTU-related BIO callbacks.
Deduct the assumed IP MTU overhead from the configured MTU during
startup.
Unlike the previous code, this does not necessarily send DTLS from the
same socket that received a message, nor to the same address that sent
one, and instead always uses the selected_sfd and ->endpoint. This may
or may not be a regression.
Closes#1806
Change-Id: I4d4456df3f378d00782cbfa64afdb2a038217e6c
(cherry picked from commit 08332161cf)
(cherry picked from commit 5331506303)
Remove all codecs that cannot be transcoded to when the audio player is in
use.
Add safety to make_transcoder to return failure in case an unsupported
codec is requested for transcoding.
Convert leftover passthrough handlers to SSRC passthrough if there are
any (shouldn't be).
closes#1858
Change-Id: I1822e48723622d550624c7355a1acfbf8ca38eb8
(cherry picked from commit 4573f32d9b)
(cherry picked from commit eddc467899)
If a stream has been pushed to the kernel from anything other than RTP,
even though RTP is expected, we get a forwarding entries without any
SSRCs. This is valid, but once actual RTP is received, it needs to be
passed on to user space, so that SSRC contexts can be set up.
Possible fix for #1855
Change-Id: I51b82d3cf79cf66780fdde154bebe56e0f43174b
(cherry picked from commit a0b705eef9)
(cherry picked from commit 48bae073ff)
Address must be printed with [] if it's IPv6
Change-Id: I238a774c5fc2b9957b29a325f2c00eb02d840a4b
(cherry picked from commit 2acf5a6f85)
(cherry picked from commit 86cd5f2ab4)
... to be operational and useful on supplemental codecs (DTMF etc)
Change-Id: Ifedefb143b984e6bac49dbbd744fe4647891bc7a
(cherry picked from commit abbc02296f)
(cherry picked from commit f7b6a6c86e)
Must double escape these in "" strings
Change-Id: I896ea35928ec7a7ef4dc7709d8eb37d1f6731897
(cherry picked from commit 6eecfc0463)
(cherry picked from commit a0638734e3)
on <pkg.ngcp-rtpengine.no-transcoding> profile
Change-Id: Ie67a4846f52ee3a37745ab18095584753afb2d89
(cherry picked from commit a73eeb8f09)
(cherry picked from commit 428811c37f)
The markdown package seems dead upstream, and the packaging situation
in Debian is not very good either. It currently has an RC bug which
means rtpengine itself is marked for autoremoval from testing. While
we could fix the RC bug for the markdown package, it's probably better
to switch implementation instead to something that is more lively. That
would be the discount implementation for example.
Change-Id: I25334f1539090d5e2d8008f460a7459713241d7c
(cherry picked from commit 9785aa05dd)
(cherry picked from commit 333cec2595)
Only increase the SRTCP index when encrypted SRTCP is actually in use.
This would be apparent when a stream is switched from RTCP to SRTCP.
Change-Id: I3a31377b89c5124035152d504e7d99f8cf1d96a1
(cherry picked from commit c5f56c4758)
The PT index can be -1 or -2. Ignore RTP stats in this case.
fixes#1814
Change-Id: Ifdcdbccad592fd1a27d2b31359861ecb3e725546
(cherry picked from commit ee162f4889)
having injected events on the recv list can cause out of order TS
values which results in is_in_dtmf incorrectly returning NULL and
letting the transcoded PCM frames through.
It also doesn't make sense to add DTMF to the send list unless they're
actually being sent, so injected delayed or unblocked
Change-Id: I07e2a35e27142715a5257f199326b7a3d133e2a8
(cherry picked from commit c57a3f7def)
now that the returned last_event_ts is always that of the previous
DTMF, we can ensure that the next one isn't transmitted until that
time plus the required pause.
Like the num_samples calculation, the actual time needs to be
increased by 1 packets worth of samples so tha pause lasts the
full duration required
Change-Id: I6da1dd7cbcf49f7f0431a5123df2cdc382fe3dba
(cherry picked from commit e12e408ee3)
this function is used to determine if a pause is needed on a new
injected DTMF's start ts to ensure a gap between the events. However,
if an inject request comes in after the end of the previous event
but before it would have been offset due to pause, no pause is added
This change returns the ts value from dtmf_state if the queue is
empty as that will always be the ts of the last DTMF transmitted
Change-Id: I4f3cf5115d1a8e26c0ca1bc9570c46e29391e0d0
(cherry picked from commit bdb1022677)
the num_samples was added to the start_pts, which is the first event
packet timestamp, which has already increased its ts by its event
duration. so, the total duration of events ends up being one packet
more than intended.
Change-Id: I423bb222a81c5bd78e570ff2026c72dd4dd1b100
(cherry picked from commit c92748dc9d)
Make sure we increase the output RTP sequence
number for each generated packet in the case
of packets that need to be duplicated or sent
repeatedly (DTMF end event)
Change-Id: Ia16ffefc0791d01575248ac5d8025eb30ccaec67
(cherry picked from commit 41a9894cce)
in some scenarios the start event ts can be before the *pts value, which
will result in a shortened DTMF event being transmitted than expected during
injection as the end event ts is calculated based on that initial dtmf start
value.
This change updates the end event ts by the amount the start ts was
behind, so that the resulting event has the right duration
Change-Id: Ia637d1e1c5d92de8b35317ec552c22eae23c0645
(cherry picked from commit c7fa81c764)
when DTMF is being transmitted using codec_output_rtp, the scheduled
time to send the packet is calculated using the timestamp difference
from the last transmitted applied on top of the realtime time the
last packet was sent. However, the timestamp is only updated on the
first event packet so a delay needs to be passed in to codec_output_rtp
to ensure the subsequent packets are scheduled based on the event
duration of the event packets
Change-Id: I5a2f6cf67b5f570f6099d201592d9a6fc01d60a5
(cherry picked from commit 6aa10931ad)
Looks like this didn't exist prior to 1.1.1
Fix-up for 8fba68f2c9
Change-Id: I1568ce6c583114659cc2e9997269f45043bd220a
(cherry picked from commit d6b09b53ea)
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