Commit graph

8564 commits

Author SHA1 Message Date
George Kiagiadakis
eafa9f08f7 splitmuxsrc: print potentially negative offset with a sign 2016-01-25 15:36:29 +01:00
Tim-Philipp Müller
5d14746792 taginject: fix sample pipeline in docs
https://bugzilla.gnome.org/show_bug.cgi?id=679571
2016-01-21 15:30:42 +00:00
Tim-Philipp Müller
aeed2e550c rtp: fix compiler warnings with gcc-6
In file included from gstrtpL16depay.h:27:0,
                 from gstrtp.c:73:
gstrtpchannels.h:154:33: error: 'channel_orders' defined but not used [-Werror=unused-const-variable]
 static const GstRTPChannelOrder channel_orders[] =
2016-01-19 13:04:39 +00:00
Sebastian Dröge
7927f49ca0 wavparse: Don't play anything after the end of the data chunk even when seeking
Especially in push mode we would completely ignore the size of the data chunk
when not stop position is given for the seek. Instead make sure that the end
offset is at most the end of the data chunk if known.

Without this we would output anything after the data chunk, possibly causing
loud noises if the media file is followed by an INFO chunk or an ID3 tag.
2016-01-19 14:57:03 +02:00
Sebastian Dröge
322bdf5136 wavparse: Don't do calculations with -1 offsets when handling SEGMENT events
We use that to signal "infinity", taking the difference between that and some
other value is not going to give us any useful result for the end offsets of
segments.
2016-01-19 14:55:57 +02:00
Sebastian Dröge
366bbffcd8 Revert "WIP: rtpjitterbuffer: Add RFC7273 media clock handling"
This reverts commit 271501f657.

It wasn't meant to be pushed yet as the commit message indicates.
2016-01-18 11:30:45 +02:00
Aleix Conchillo Flaqué
665d14a2a0 rtspsrc: handle rtcp/srtcp caps properly when using interleaved data
We check the stream profile and use the proper RTCP caps:
application/x-srtcp if we are using a secure profile and
application/x-rtcp otherwise.

https://bugzilla.gnome.org/show_bug.cgi?id=760556
2016-01-18 11:29:25 +02:00
Sebastian Dröge
271501f657 WIP: rtpjitterbuffer: Add RFC7273 media clock handling 2016-01-18 08:58:59 +02:00
Sebastian Dröge
53c797d604 wavparse: When flushing on EOS, don't process more data than the "data" size
Even if we have more data queued up when flushing than the size of the data
chunk, don't process and output it. If the data size is known, this likely
contains another chunk (e.g. an INFO chunk) or things like ID3 tags. Just
outputting them as if they were data is going to cause unexpected behaviour
and unpleasant audio noises.
2016-01-13 23:42:31 +01:00
Antonio Ospite
bdcc0390af interleave: Fix the example by setting channel-masks in the sink pads
The current example does not work, it fails with:

ERROR: from element /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0: Internal data flow error.
gstwavparse.c(2178): gst_wavparse_loop (): /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0:
streaming task paused, reason not-negotiated (-4)

This is because negotiation with wavenc gets messed up by the missing
channel positions configuration.

The proper way to define the channel layout when using the interleave
element in code would be to set the channel-positions property, but
gst-launch-1.0 does not know how to deal with arrays; so the example
pipeline works around the issue by setting the channel-masks in the sink
pads.

Also fix a repetition in the deinterleave example description

https://bugzilla.gnome.org/show_bug.cgi?id=735673
2016-01-12 22:11:30 +00:00
Tim Sheridan
205565ccd9 sbcparse: Fix frame length calculation
SBC frame length calculation wasn't being rounded up to the nearest byte
(as specified in the A2DP 1.0 specification, section 12.9). This could
cause 'stereo' and 'joint stereo' mode SBC streams to have incorrectly
calculated frame lengths.

Incorrect frame length calculation causes frame coalescing to fail, as
subsequent frames in the stream aren't found in the expected locations.

https://bugzilla.gnome.org/show_bug.cgi?id=742446
2016-01-12 21:52:12 +00:00
Reynaldo H. Verdejo Pinochet
0bb8000874 flacparse: demote warning on wrong reserved value to fixme
We are likely just parsing a backward-compatible stream we
don't fully support.
2016-01-10 22:54:12 -08:00
Thiago Santos
4ac0a49308 imagefreeze: simplify caps selection
The downstream caps query with a filter alraedy gives us the possible
intersection so there is no need to check it again with downstream
if it is supported. Just try to set it directly.
2016-01-08 16:29:29 -03:00
Tim-Philipp Müller
3aa0dd8629 rtph264depay: fix unnecessary sub-buffer creation
We create a sub-buffer just to copy over its metas and then
throw it away immediately, just use the original input buffer
directly.
2016-01-08 16:40:32 +00:00
Tim-Philipp Müller
6171b0a675 rtpdvdepay: fix unnecessary sub-buffer creation
We create a sub-buffer just to copy over its metas and then
throw it away immediately, just use the original input buffer
directly.
2016-01-08 16:40:32 +00:00
Tim-Philipp Müller
c75f94c8f5 rtpamrdepay: fix unnecessary sub-buffer creation
We create a sub-buffer just to copy over its metas and then
throw it away immediately, just use the original input buffer
directly.
2016-01-08 16:40:32 +00:00
Tim-Philipp Müller
a8b8643977 rtpvrawdepay: fix major memory leak and performance issue
We call gst_rtp_buffer_get_payload() which creates a sub-buffer
of each input buffer, just to copy over metas, and then leak it.

https://bugzilla.gnome.org/show_bug.cgi?id=760289
2016-01-08 16:40:28 +00:00
Tim-Philipp Müller
6dab3ece07 flacparse: don't map buffer multiple times when parsing 2016-01-07 16:24:09 +00:00
Steven Hoving
910d75ddaf matroska: Store subtitle stream count in the correct variable
And don't override the video stream count instead.
2016-01-07 18:20:30 +02:00
Sebastian Dröge
4917515342 equalizer: The child-proxy API is GObject based in 1.x
Not GstObject anymore.
2016-01-05 18:59:25 +02:00
Reynaldo H. Verdejo Pinochet
ba094b50e1 flacparse: add debug msg on CRC mismatch while validating frame header 2015-12-31 16:04:15 -08:00
Reynaldo H. Verdejo Pinochet
6b7675e4a2 flacparse: drop unneeded braces at _parse_frame() exit
Additionally, drop redundant comment & line break
2015-12-31 16:04:15 -08:00
Reynaldo H. Verdejo Pinochet
b6ebad0997 flacparse: minor grammar correction 2015-12-31 16:04:15 -08:00
Reynaldo H. Verdejo Pinochet
5234c7c2bd flacparse: update URLs on pointers to online spec 2015-12-31 15:34:57 -08:00
Reynaldo H. Verdejo Pinochet
5f4317843c flacparse: make buffer DTS setting explicitly unconditional
We are setting it to PTS regardless of block_strategy
2015-12-31 14:40:15 -08:00
Reynaldo H. Verdejo Pinochet
2c14f2fff1 flacparse: add actual invalid block type to warning
For someone that read the spec is clear the only *invalid*
data block type is 127. For the rest, its useful information.

Additionally. values 7-126 are currently reserved by the
spec so the situation might change in the future.
2015-12-31 14:21:40 -08:00
Reynaldo H. Verdejo Pinochet
c43f84abf3 flacparse: use shift instead of mask & comp
We are only interested on the first bit of the first
byte of the metadata block header to figure out whether
is marked as the last one. The shift makes it quite
clearer.
2015-12-31 14:12:36 -08:00
Reynaldo H. Verdejo Pinochet
8a745837aa flacparse: warn on wishful parsing of weird headers
If we get anything from 7 to 126 as type when parsing
a metadata block header, we are likely dealing with a
FLAC stream version we don't fully understand. Issue
a warning if so.

Document function assumptions regarding the passed-on
type while at this.
2015-12-31 13:04:23 -08:00
Reynaldo H. Verdejo Pinochet
df6f0bc595 flacparse: show meaningful info on frame CRC check
As CRCs are calculated for the comparition already, we
might as well (cheaply) inform the user how the numbers
differ if a missmatched pair is found.

While at it:

Rephrase candidate-frame message to make more sense
2015-12-31 13:04:23 -08:00
Reynaldo H. Verdejo Pinochet
395afed566 flacparse: drop remaining trailing whitespace 2015-12-31 13:04:23 -08:00
Reynaldo H. Verdejo Pinochet
a086ee6192 flacparse: drop superflous else clauses 2015-12-31 13:04:23 -08:00
Reynaldo H. Verdejo Pinochet
7286aae6e5 flacparse: factor out buffer time and offset resetting
Avoids multiple occurrences of the same resetting pattern
2015-12-31 13:04:23 -08:00
Reynaldo H. Verdejo Pinochet
5bf1f1ec9c flacparse: move block handling by type out of _parse_frame() 2015-12-31 13:04:23 -08:00
Hyunjun Ko
3300039513 rtspsrc: replace duplicated codes to call new base sdp apis
https://bugzilla.gnome.org/show_bug.cgi?id=745880
2015-12-31 17:12:09 +02:00
Reynaldo H. Verdejo Pinochet
eb47176b7c flacparse: drop redundant return statement on _header_is_valid()
Fix the rather vague error message while at it.
2015-12-30 22:33:58 -08:00
Reynaldo H. Verdejo Pinochet
276fcc5916 flacparse: rework gst_flac_parse_frame_is_valid()
drop unnecessary nesting looking for end of frame
2015-12-30 21:43:55 -08:00
Reynaldo H. Verdejo Pinochet
90b62be301 flacparse: factor out context clearing routine 2015-12-30 21:43:45 -08:00
Sebastian Dröge
e618444ca7 matroskademux: Guard against no codec data in prores caps creation
CID 1346532
2015-12-29 18:05:56 +02:00
Sebastian Dröge
903c431d6d scaletempo: Free the various buffers in GstBaseTransform::stop()
Previously we leaked them completely, but as they're specific to the caps
freeing them in stop() instead of finalize() makes most sense.
2015-12-25 11:41:19 +01:00
Thiago Santos
0906d822ad qtdemux: drop flushes from our own offset seek
Prevents downstream from receiving flushes for a seek only in
upstream. Those seeks are only to start reading from the right
offset when skipping or returning to qt atoms.

https://bugzilla.gnome.org/show_bug.cgi?id=758928
2015-12-22 12:33:39 -03:00
Thibault Saunier
7b026e4bc0 matroskademux: Always set the channel mask for PCM streams
Just use the gst_audio_channel_get_fallback_mask function for now as
the specification is too complicated and nobody implements it.
2015-12-21 18:34:42 +01:00
William Manley
77cdb23850 progressreport: add support for using format=buffers with do-query=false
This is useful for investigating and debugging pipelines which are
producing buffers at a slower/faster rate than you would expect.

https://bugzilla.gnome.org/show_bug.cgi?id=759635
2015-12-20 20:28:56 +00:00
Jan Schmidt
774a32ff89 qtmux: Don't write invalid edit list start time.
Avoid writing a negative number as a large positive
integer in an edit list when the first_ts is smaller
than the first_dts - which can happen when the first
packet received has a PTS but no DTS.

https://bugzilla.gnome.org/show_bug.cgi?id=759615
2015-12-19 03:49:28 +11:00
Jan Schmidt
675a4088e5 splitmuxsink: Only update running time when it increases.
Don't increment running time from every buffer. The correct
logic to only increment when running time advances is a
little further down, so delete this left-over line.
2015-12-19 03:49:28 +11:00
Thibault Saunier
10d1ba1477 matroska-mux: Implement prores support
https://bugzilla.gnome.org/show_bug.cgi?id=758258
2015-12-19 03:49:28 +11:00
Jan Schmidt
71d43327a3 matroska-demux: Play ProRes video streams
Generate video/x-prores caps for ProRes video streams.
Every frame needs an 8 byte header prepended, as described in
http://wiki.multimedia.cx/index.php?title=Apple_ProRes#Frame_layout
so do that in a post-processing callback.

https://bugzilla.gnome.org/show_bug.cgi?id=758258
2015-12-19 03:47:49 +11:00
Vincent Dehors
c1b66a63ac rtpj2kdepay: Push one JPEG2000 frame per buffer, not a buffer list with multiple buffers
https://bugzilla.gnome.org/show_bug.cgi?id=758943
2015-12-17 16:04:07 +01:00
Dave Craig
328346ad21 audioparsers: Check for NULL return value of gst_pad_get_current_caps()
https://bugzilla.gnome.org/show_bug.cgi?id=759503
2015-12-16 10:12:44 +01:00
Evan Callaway
4718870959 rtspsrc: Retry connection if tunneling needs authentication
Leverage response from gst_rtsp_connection_connect_with_response to
determine if the connection should be retried using authentication.  If
so, add the appropriate authentication headers based upon the response
and retry the connection.

https://bugzilla.gnome.org/show_bug.cgi?id=749596
2015-12-14 16:42:27 +01:00
Luis de Bethencourt
4735d2a9a5 rtspsrc: check port-range format
The string could exist but with a wrong format, in that case we still want
to reset the values of client_port_range.min and max like we do if there is
no string.

CID 1139593
2015-12-14 14:53:57 +00:00