The GLib slice allocator has been obsoleted. Having a mixture of two
allocators can lead to hidden issues on systems that use different
implementations. Take the leap and replace everything with g_new/g_free.
Change-Id: I025c8383ef47b2c2472573360407fd6a6ca339b3
In many transcoding scenarios resampling is not actually required. We
can shortcut the operation by just returning the original frame, instead
of a cloned and newly allocated one. We just need to distinguish between
the cases to determine whether the frame returned by resample_frame()
needs to be freed.
Change-Id: I3f36a46bd3b967f140c8353119fdb24ad8363c15
... so that the desired wav channel can be controlled when producing a
mixed audio file
When a mixed wav file is created, the channels in the wav container are
currently allocated in the same order as each SSRC is received, meaning
it is impossible to know which channels have been allocated to the offer
or answer side of the call. Furthermore if there is a reinvite or media
file played, these are also allocated in the order that SSRC is received
- so an "answer" could end up sharing a channel with an "offer" with no
way of knowing this.
This patch allows you to specify how many channel slots should be
allocated within the mixer, and allows you to then specify which slot is
assigned to each media in the call (this will usually be 2 slots in
total, slot 1 for answer, slot 2 for offer or vice versa).
Ported from https://github.com/sipwise/rtpengine/pull/1852Closes#1857Closes#1852
Change-Id: I010208427cabc3a48d6ef7bd3a84e9a5bdcfd492
Obsolete str_init(), rename STR_INIT() to just STR(), and replace all
instances of str_init() with STR().
no-op
Change-Id: I981529063ad2ea26089add467f7a84b638dbf423
Some source strings are `const char *` and we don't have an alternative
`str` type pointing to a const buffer. Strictly speaking some buffers
aren't even char buffers, yet we use the `str` type to manage them. This
eliminates the need to explicitly cast every usage.
Change-Id: Ifd3cdd3d400fc483e6c488a95e9381938e63cc7f
... stream after a reconnect
if we're in the middle of a disconnect then ssrc_tls_state may have destroyed
the streambuf so we need to skip the rest of the function, otherwise the now
null pointer is passed into streambuf_write, which will return immediately and
the metadata will have been marked as sent when it hasn't been
closes#1694
Change-Id: I679fdd0ff9245e4413bc5a5ab40a70ea4f9ef7c0
If the forwarded streams of a call are later combined without the inserted
silence, the resulting mix may end up with them out of sync. This
is already handled when outputting mixed files so the same functionality
from mix.c has been added for tls forwarding
This type is used as const everywhere except internally, so make it part
of the typedef for brevity.
Change-Id: Ic4afe037b392239a991d5380c6708903011da29e
This makes it possible to have codecs running at variable clock rates
that differ from their RTP clock rates.
Change-Id: Ia2f5effb82eefe8c3028573ba0a6697da28473b1
Instead of just having an integer multiplier, support a fractional
factor. This allows us to have the RTP clock rate run faster than the
audio clock rate, and not just slower by an integer factor.
Change-Id: I7681cf369c43d8424ca2d2ebeffe932595d271ec
Parsing out the a=fmtp string has been left up to the codec init
function until now, with the values that resulted from the parsing being
stored only within the codec. Convert this to an explicit method to
parse the a=fmtp string, and introduce a dedicated struct to store the
resulting values.
Functionally this change is a no-op.
Change-Id: Ia84e26d632ed5209b4439fd82c1e4e38850fd024
Provide a standard output format if no other outputs are configured, so
that the decoder has something to work with. Applicable to TLS-send-only
scenarios without recording.
Change-Id: I627bb7af3f3033e1025009c21a4da6991e491dcf
required to distinguish between 20-ms and 30-ms modes, both for encoding
and decoding
add support for the iLBC mode= format parameter and dynamic mode
switching
closes#854
Change-Id: Icb6f0ec80df86d27681c689c168b24f163a2db06
Fixes garbled audio for certain codes (e.g. G.729)
Also adds some additional debug output
Fixes#410
Change-Id: I1dbae2638f1e55bf80bb96549a75b9c4b82d08bf
Richard Fuchs
Peter Kelly
Tom Briden
Frederic-Philippe Metz