Don't use g_assert() for error handling, even if they're highly unlikely.
Either we *know* that something can't happen, in which case we
should just not handle it, or we think something can happen, but it is
very very unlikely that it will ever happen, in which case we should
handle it like any other error instead of asserting.
g_assert() is best left for conditions we have control of, like checking
internal consistency of our code, not checking return values of external
code.
Fixes a bunch of warnings when compiling with -DG_DISABLE_ASSERT:
gstrtpgsmpay.c: In function 'gst_rtp_gsm_pay_handle_buffer':
gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used
gstspeexenc.c: In function 'gst_speex_enc_encode':
gstspeexenc.c:904:19: warning: variable 'written' set but not used
pulsesink.c: In function 'gst_pulsesink_change_state':
pulsesink.c:2725:9: warning: variable 'res' set but not used
pulsesrc.c: In function 'gst_pulsesrc_change_state':
pulsesrc.c:1253:7: warning: variable 'e' set but not used
GCC 4.6.x spits warnings about variables that are unused but set. Such
variables have been removed where trivial but with comments left behind
for informational purposes in some cases.
gst_rtp_session_chain_recv_rtcp () was changed in commit 490113d4
to always return GST_FLOW_OK instead of the return value of
rtp_session_process_rtcp (), so we'll keep it that way.
Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files.
Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git
If we get GAP samples, there is no need to transmitt it.
In some situations, microphone is muted, we can drop net traffick
usage to ~1 kbit/s. Without patch it will stay ~20 kbit/s
Theora can only use the last frame (or the keyframe) as a reference, so in
practice. If we receive a buffer that references an unknown codebook, request
new headers. It probably means that headers were lost.
Improve parsing of the samplerate.
Parse the framelen so that we can calculate timestamps.
When interpollate the incomming timestamp on outgoing buffers when there are
multiple subframes.
fixes#625825
atof() converts strings according to the current locale, but the
framerate string will likely always use a dot as floating point
separator, so use g_ascii_strtod() instead (but also canonicalise
the string before, so we can handle both formats as input).
... thereby (partially) deprecating properties currently controlling whether
or not byte-stream output or NAL/AU alignment (though properties still determine
fallback if nothing specified in caps).
Fixes#606662.
Use 3 adapters, one to accumulate paketization units, another on to accumulate
tiles and a last one to accumulate the final frame.
Don't just blindly flush the adapter on DISCONT but only discard the current
packetization unit.
When we dropped jpeg2000 packets between SOP markers, adjust the SOT header with
the new lenght.
When parsing the bitstream, look for SOP markers because we are allowed to split
packets on those marker boundaries.
Rework the parsing code a little so that we can pack multiple Packetization
units in one RTP packet.
Only set the delta flag when all of the units in the packet are delta units.
Based on patch from Olivier Crête <olivier.crete@collabora.co.uk>
Fixes#632945
Incomming buffer is only pushed on the adapter at the end of the
handle_buffer function. But duration/timestamp of this buffer is already
taken into account for the current data in the adapter. This leads to
wrong rtp timestamps and extra latency.
Put a DISCONT event on the next output buffer when the input buffer had a
DISCONT.
Make sure we clear our adapter and reset our state before going to PAUSED.
Free the qtables.
Fixes#626869
Although the spec says that the clock-rate should always be 90000, some rtsp
servers send different clock-rates so we must accept then in order to handle
those streams too.
When we can't find any channel or encoding-params on the caps for dynamic
payload types, set the default number of channels to 1, as the spec says we
should.
See #623209
When parsing the number of channels, use the encoding-params property from the
RTP caps because that is where we can find the channels according to the spec.
Fall back to the channels property in the caps when needed.
Fixes#623209
G729 packets may only occur intermittently (e.g. cn packets), and as such
do not allow for perfect-rtptime calculating rtp times based on frame or byte
count. In particular, do not use rtp audio base payloader as base class, but
rather base payloader directly.
Even though we don't use delivery-method in our payloader, older versions of
the theora payloader in gstreamer required it. As such we need to keep this
around in the caps for backwards-compatibility.
This reverts part of 49463a37cbFixes#618940
When we calculate the frame duration, we need to use the amount of
frames in the _previous_ packet, not the current packet. The frame duration is
needed to correctly de-interleave interleaved streams. This fixes the case where
there are a variable number of frames in a packet.
Fixes#620494
It probably will not be in the final RFC as it is not in RFC 5215 for Vorbis.
If there is a configuration specified, assume it is in-line and if nothing is
specified, assume it is in-band.
https://bugzilla.gnome.org/show_bug.cgi?id=618386
Don't blindly add the durations of incomming buffers to the total queued
duration because it might be invalid. Mark the total queued duration invalid
when we receive an invalid incomming timestamp because that's when we lose track
of the total queued duration.
Fixes#618324
Add a new config-interval property to instruct the payloader to insert
configuration headers at periodic intervals in the stream
(when a keyframe is countered).
Add a new config-interval property to instruct the payloader to insert
config (VOSH, VOS, etc) at periodic intervals in the stream
(when a GOP or VOP-I is encountered).
Based on patch by <marc.leeman at gmail.com>
Fixes#607452.
... which evidently makes (most) sense if output buffers are
actually frames.
Partially based on a patch by
Miguel Angel Cabrera <mad_aluche at hotmail.com>
Fixes#609658.
So we stop dropping fragments as soon as there is a picture start (code).
In particular, this prevents dropping the first frame following
initial DISCONT.
... rather than falling back to sending the whole frame in one packet
if number of GOB startcodes < maximum.
One might take this further and still perform Mode B/C payloading,
but at least this should cater for decent fragments in typical cases.
Fixes#599585.
in rtph264depay.c, lines 577-576, NALU-type 24 (Single-Time Aggregation
Packet) is handled in fall-through as NALU-type 26 (unhandled).
This leads high quality h264 streams such as:
rtsp://stream.yle.mobi/yle/areena/MEDIA_E0342657_p3.mp4
to fail with "NAL unit type 24 not supported yet" (but it's actually
supported), and thus to close any stream which contains STAPs.
The proposed one-liner patch fixes the issue.
Fixes#615051.
When no constantDuration has been given in the caps, try to derive one from the
timestamp difference between packets. Also keep doing this for each packet
because some broken streams might simply provide wrong timestamps.
The profile-level-id represents restrictions on what can be sent, it does not
describe the stream. So it should be reflected in the sink caps of the
payloader, not the src caps.
https://bugzilla.gnome.org/show_bug.cgi?id=607353
Use GstRTPBaseAudioPayload as the base class. This saves a lot of code and fixes
a bunch of problems that were already solved in the base class.
Fixes#853367
Add a new spspps-interval property to instruct the payloader to insert
SPS and PPS at periodic intervals in the stream.
Rework the SPS/PPS handling so that bytestream and AVC sample code both use the
same code paths to handle sprop-parameter-sets. This also allows to have the AVC
code to insert SPS/PPS like the bytestream code.
Fixes#604913
celtdepay : added default framesize(480) channels(1) and clockrate(32000)
depay_setcaps : now gets channels and framesize from string with default value
depay_process : now adds timestamp to outbuf
Added frame_size to GstRtpCeltDepay
Changed some GST_DEBUG to GST_DEBUG_OBJECT or GST_LOG_OBJECT
celtpay : getcaps : gets channel and framesize and sets caps
Added frame-size to static caps for audio/x-celt
In case of non-interleaved (= sequentially payloaded) streams,
the AU-Index serves little purpose (that is not already covered by
RTP fields). (Broken) Payloaders might consider this field then
to be disregarded and have non spec compliant values, e.g. each
RTP packet having AU-Index 2 (rather than 0). As such, ensure/force
simple sequential sending of non-interleaved streams.
Whenever we see a gap, we flush the temporary packets (but not the adapter). If we
had some data temporarily stored it will be outputted (the sound will sound a bit
garbled... but that's how it sounds on MacOSX :)
Reverse-engineered by comparing:
* A rtp hinted file provided by DarwinStreamingServer
* The output procued by DSS for that same file
Also used various streaming sources available on the internet to fine-tune
the code.
The header/codec_data extraction methods are from FFMpeg (LGPL).
