Do not allocate payload size outbuf if appending payload buffer.
The commit 137672ff18 attached payload
to the output buffer but forgot to remove payload allocation. That
effectively doubled payload size and add zero'ed or random bytes.
Makes the following pipeline work again:
gst-launch-1.0 -v audiotestsrc wave=2 ! gsmenc ! rtpgsmpay ! rtpgsmdepay ! gsmdec ! autoaudiosink
https://bugzilla.gnome.org/show_bug.cgi?id=784616
Soup allows only up to two connections per host in a session,
if we use default value. When session sharing is used, however,
more connections might be required in a session.
(e.g., multi-audio adaptive streaming case)
https://bugzilla.gnome.org/show_bug.cgi?id=784495
gst_util_uint64_scale_int takes a gint as denom parameter
whereas ctx->clock_rate is a guint32.
It happens when gst_rtp_packet_rate_ctx_reset set clock_rate
to -1.
So just define clock_rate as gint like it is done in rtpsource.h
https://bugzilla.gnome.org/show_bug.cgi?id=784250
When resurrecting a buffer, the subsequent free call can result
in the group-released handler being called again, which causes
a recursive loop. This patch blocks the signal handler during
the time that it executes, ensuring that the loop will not occur.
https://bugzilla.gnome.org/show_bug.cgi?id=759292
SoupSession's ssl-ca-file property is deprecated. Use the recommended
tls-database property.
This is a bit more complex as it requires creating a GTlsFileDatabase
object for an absolute (!) path to the CA certificates file.
https://bugzilla.gnome.org/show_bug.cgi?id=784005
Recent GnuTLS disregards the Common Name and only looks at the Subject
Alternative Name extension. Since our test-cert has no SAN extension,
validation fails.
Generate a new certificate with SAN. In addition to 127.0.0.1, for good
measure make it valid for localhost and ::1, too.
https://bugzilla.gnome.org/show_bug.cgi?id=784005
Currently we only allowed HTTP proxy. Don't filter for the scheme, just check
if it looks like an URI. Soup will warn if the URI is invalid or if
proxy protocol is not supported. This enables using SOCKS 4/5 which is
directly implemented into GIO.
https://bugzilla.gnome.org/show_bug.cgi?id=783012
Increasing this number fix a buffer starvation problem I'm hitting
with a "v4l2src ! kmssink" pipeline.
kmssink requests 2 buffer as it keeps a reference on the last rendered
one. So we were allocating 3 buffers for the pipeline.
Once the first 2 buffers have been pushed we ended up with:
- one buffer queued in v4l2
- one being pushed
- one kept as last rendered
If this 3rd buffer is released after that v4l2 used the first one to
capture we end up with a buffer starvation problem as no buffer is currently
queued in v4l2 for capture.
Fixing this by adding one extra buffer to the pipeline so when one
buffer is being pushed downstream the other can already be queued to
capture the next frame.
We were already adding 3 buffers if downstream didn't reply to the
allocation query. I reduced this number to 2 to compensate the extra
buffer which is now always added.
https://bugzilla.gnome.org/show_bug.cgi?id=783049
This patch fixes a memory leak that is caused if the dmabuf file
descriptor dup fails. Previously, _cleanup_failed_alloc() would
not unref the memory because mems_allocated had not yet been
incremented.
https://bugzilla.gnome.org/show_bug.cgi?id=784302
When set this property will allow the jitterbuffer to start delivering
packets as soon as N most recent packets have consecutive seqnum. A
faststart-min-packets of zero disables this feature. This heuristic is
also used in rtpsource which implements the probation mechanism and a
similar heuristic is used to handle long gaps.
https://bugzilla.gnome.org/show_bug.cgi?id=769536
Even though hooked up to the build system, it's clear that no one
has ever built or used this with GStreamer 1.x. It wants to link
against libgstinterfaces, which no longer exists. And uses 0.10-style
raw audio caps. And the last meaningful change was done in 2009.
Let's just remove it.
We currently send data to the RTSP connection from multiple threads:
whenever a command is to be handled and whenever RTCP is generated. This
can cause data corruption or worse if both happen at the same time.
As such, protect gst_rtsp_connection_send() and gst_rtsp_connection_receive()
calls with a mutex. While this means that we hold a mutex during the IO
operation, this is not actually a problem as the IO operation can be
interrupted (gst_rtsp_connection_flush()) at any time and is blocking by
itself anyway.
