Commit graph

10459 commits

Author SHA1 Message Date
Tim-Philipp Müller
ad245a0dc2 matroska-demux: don't leak audio codec_data buffer 2011-10-10 19:02:58 +01:00
Stefan Sauer
4b06e3095a tests: add missing PLUGIN_ASE_LIBS to LDADD 2011-10-10 13:20:04 +02:00
Mark Nauwelaerts
00a91fc061 speexenc: only push header buffers following initial events 2011-10-09 21:32:32 +02:00
Thiago Santos
ca417fd376 qtmux: Fix memory leak on atoms recovery function
Remember to free the ftyp data after writing it to a file.

Fixes #660969
2011-10-09 11:18:18 -03:00
Vincent Penquerc'h
be82dd8e3a matroskademux: improve segment handling with non-zero starting timestamp
... as well as related items, such as seeking and position reporting.

https://bugzilla.gnome.org/show_bug.cgi?id=659808
2011-10-05 14:34:55 +02:00
Stas Sergeev
73fac4e5bc v4l2, ximagesrc: fix some printf format compiler warnings
https://bugzilla.gnome.org/show_bug.cgi?id=660150
2011-09-30 18:05:32 +01:00
Thiago Santos
a4154e9db2 tests: qtmux: Refactor bitrate check test
Refactor bitrate check test to accomodate multiple tests
for bitrate
2011-09-30 13:05:24 -03:00
Thiago Santos
535f92a0a4 qtmux: update esds atom under wave atom for aac bitrates
AAC in mov format puts an ESDS atom inside of a WAVE atom in
STSD atom, we need to update the bitrate on this ESDS. This patch
fixes it.
2011-09-30 13:05:24 -03:00
Thiago Santos
31acc88b39 qtmux: Also update btrt atom
When rewriting bitrates, also update the btrt atom under stsd
2011-09-30 13:05:24 -03:00
Thiago Santos
e58b0466ec tests: qtmux: add tests for bitrate average calculation
Adds tests to make sure qtmux/mp4mux sets average bitrate
correctly
2011-09-30 13:05:20 -03:00
Thiago Santos
7a143ea94f qtmux: Calculate average bitrate for streams
Calculate and use average bitrate for streams when no
bitrate tag was received
2011-09-30 12:43:13 -03:00
Thiago Santos
4737090594 qtmux: Avoid a buffer metadata copy if possible
If first_ts is 0 there is no need to subtract, so we might
skip some copying to make the buffer metadata writable.
2011-09-30 12:43:13 -03:00
Tim-Philipp Müller
ca77c96c51 speexenc: initialise variable before adding to it 2011-09-29 23:21:46 +01:00
Mark Nauwelaerts
c5354bee04 speexdec: port to audiodecoder 2011-09-29 17:33:25 +02:00
Mark Nauwelaerts
53476c1580 speexenc: clean up some unused remnants 2011-09-29 17:33:23 +02:00
Mark Nauwelaerts
c1909c32c5 speexenc: port to audioencoder 2011-09-29 17:33:21 +02:00
Mark Nauwelaerts
e8bcd41d73 flacenc: port to audioencoder 2011-09-28 16:14:46 +02:00
Vincent Penquerc'h
671b56f9da matroskademux: ensure minimal alignment for audio/x-raw-* buffers
Since matroskademux will attempt to push unaligned buffers,
downstream might have trouble with those, especially if downstream
uses ORC, such as audioconvert.

Ensure we push buffers aligned to the basic type at least for
those raw buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=659798
2011-09-28 12:49:42 +02:00
Raimo Järvi
827c3aa14b goom2k1: Fix compiler warnings on 64 bit mingw-w64
Fixes bug #660294.
2011-09-28 00:18:15 +01:00
Tim-Philipp Müller
3828537857 soup: rename souphttpsink to souphttpclientsink
To avoid confusion, and because we might want a server
sink at some point too.

https://bugzilla.gnome.org/show_bug.cgi?id=659947
2011-09-25 15:13:39 +01:00
Tim-Philipp Müller
be7cbd4c21 souphttpsink: don't create unused second sink pad object
The base class will create the sink pad.
2011-09-23 16:39:46 +01:00
Julien Isorce
2131a3b7f8 ac3parse: correctly check for ac3/e-ac3 switch
https://bugzilla.gnome.org/show_bug.cgi?id=659943
2011-09-23 16:26:50 +01:00
Mark Nauwelaerts
fd757890eb rtph264depay: improve downstream flow return feedback to upstream
... although basertpdepay does not really make it easy/possible to do so
all the way.
2011-09-20 14:14:39 +02:00
Vincent Penquerc'h
82927d6bdd ximagesrc: add xid and xname properties to allow capturing a particular window
A particular window may be selected using the new xid (X-Window
XID, eg a pointer) and xname (window title) properties. If both
are specified, the XID is used in preference, falling back to
xname if not found.

Default (if none of xid and xname are specified, or if no such
window is found) is to capture the root window.

https://bugzilla.gnome.org/show_bug.cgi?id=546932
2011-09-20 13:09:35 +01:00
Tim-Philipp Müller
b6b072e948 tests: add unit test to make sure encodebin picks mp4mux for variant=iso
https://bugzilla.gnome.org/show_bug.cgi?id=651496
2011-09-20 12:55:31 +01:00
Ha Nguyen
931020158e rtpbin: Fix a leaked clock for each buffering message
Fixes bug #659237.
2011-09-19 14:05:26 +02:00
Mark Nauwelaerts
d959bb6041 qtdemux: parse embedded ID32 tags 2011-09-19 12:11:45 +02:00
Mark Nauwelaerts
e2179cbb74 rtpsession: avoid source premature timing out
Use slightly adjusted sender interval to determine sender timeout rather than
our own sender side interval (which may have been forced small).
2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
f65d4c8300 rtpsession: avoid timing out source too quickly
... following a PAUSE/PLAY cycle, particularly applicable when operating
with a short RTCP interval (possibly forced so server-side).
2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
77ebd33991 rtpjitterbuffer/rtpbin: relax dropping rtcp packets
... to at least having it trigger a/v synchronization, possibly without
using provided values which are still not considered sane
(as previously dropped).
2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
adfe7d0467 rtpjitterbuffer: some more reset when clearing pt map
... which in particular caters for some more reset following a possible
rtsp PLAY.
2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
81fc784163 rtspsrc: do not set elements to PLAYING when doing seek in PAUSED 2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
915db26029 rtpjitterbuffer: only reset skew on gap if input ts available 2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
1e17e10f75 rtpjitterbuffer: check some more for possible rtp timestamp discontinuity
... when operating in non slave mode, and reset if detected.
This should avoid some (large) bogus outgoing timestamp due to jumps
in rtp time, as result of PAUSE/PLAY or seek or ...
2011-09-19 11:56:40 +02:00
Mark Nauwelaerts
8599801cae rtspsrc: switch to rtp time based syncing when guessed appropriate 2011-09-19 11:52:08 +02:00
Mark Nauwelaerts
9c95072048 rtpbin: alternative inter-stream syncing methods
... at least if not syncing to NPT time:
* either sync using RTCP SR data (as currently)
* only perform the above once using initial RTCP SR packets
* discard RTCP and sync by equating provided stream's clock-base rtptime,
  as provided by jitterbuffer (typically obtained from RTP-Info in RTSP).
2011-09-19 11:52:03 +02:00
Mark Nauwelaerts
4b7301e4d1 rtpjitterbuffer: also provide clock-base to sync signal 2011-09-19 11:52:00 +02:00
Mark Nauwelaerts
f29c253934 rtpbin: allow configurable rtcp stream syncing interval
... rather than necessarily syncing at each RTCP SR.
2011-09-19 11:51:57 +02:00
Mark Nauwelaerts
afd26f0078 rtpsession: trigger reconsideration if rtcp interval set 2011-09-19 11:51:50 +02:00
Mark Nauwelaerts
3e33a7a09f rtspsrc: configure rtcp interval if provided
... in PLAY response.
2011-09-19 11:51:47 +02:00
Lasse Laukkanen
056e9188b1 isomp4: Fix allowing zero duration tracks
https://bugzilla.gnome.org/show_bug.cgi?id=637486
2011-09-19 11:18:27 +02:00
Vincent Penquerc'h
3319737e5c udpsrc: error out when no protocol is specified in the uri
It is certainly better than to crash.

https://bugzilla.gnome.org/show_bug.cgi?id=658178
2011-09-19 10:16:38 +02:00
Vincent Penquerc'h
7e4574e968 speexenc: do not use invalid buffer timestamps 2011-09-19 09:37:58 +02:00
Arun Raghavan
8ca420f547 pulse: New pulseaudiosink element to handle format changes
This introduces a new bin which wraps around pulsesink and depending on
the formats supported by the sink, plugs in/out a decodebin2 as
required. This allows users to switch sinks on the stream and adapts
accordingly (for example, you could watch a movie in passthrough mode on
your receiver which supports AC3 decode, then plug out and switch to a
non-digital profile to continue uninterrupted on analog output).

The bin is required because doing the same with playbin2/playsink will
require API changes that cannot be made in 0.10. With 0.11/1.0, we
should be able to ask for upstream caps renegotiation to deal with all
this.

https://bugzilla.gnome.org/show_bug.cgi?id=657179
2011-09-19 07:43:04 +05:30
Branko Subasic
11b0a0effc matroskademux: Avoid sending EOS when in paused state
Changed the ebml reader's gst_ebml_peek_id_length() function so
that it returns the actual reason for why the peek failed, instead
of (almost) always returning GST_FLOW_UNEXPECTED. This prevents
the pulling task from sending EOS when doing a flushing seek.
2011-09-16 15:18:48 +02:00
Vincent Penquerc'h
26ae233035 matroskademux: fix stuttering A/V
Someone got had by implicit promotion to unsigned in ops with
a signed and an unsigned value.

https://bugzilla.gnome.org/show_bug.cgi?id=659153
2011-09-15 17:29:00 +01:00
Vincent Penquerc'h
352bab2ef7 navseek: toggle pause/play on space bar
A useful thing to have.

https://bugzilla.gnome.org/show_bug.cgi?id=659065
2011-09-14 21:32:42 +01:00
David Svensson Fors
682ae32f6f matroskademux: configurable timestamp gap handling
matroskademux performs segment tricks to skip gaps in streams,
notably at start for non 0 based files.  There may however be
cases when full presentation (including intermediate gaps) is
desired, so a property allows to configure as of which gap
to act (or not at all).

API: GstMatroskaDemux::max-gap-time

Fixes #659009.
2011-09-14 14:49:36 +02:00
Thiago Santos
c78048bc25 tests: flvmux: Fix flvmux's tests after fix for request pads handling
Now that flvmux doesn't release its request pads on PAUSED->READY the
test doesn't need to re-request them for every reuse test start.
2011-09-12 09:21:47 -03:00
Thiago Santos
261d11a6d7 qtmux: Fix ctts generation for streams that don't start at 0 timestamps
Subtract the first timestamp of a stream from all input buffers to
get 0-based timestamps for creating a sane ctts table. Without this
patch the ctts could have larger values than needed, causing the
playback to have a delay at startup.

As the first timestamp is only found after a few buffers are queued
(due to possible reordered buffers), once we find the first timestamp
we subtract it from all buffers on the queue, from that point on,
all buffers have their timestamps subtract when they are collected.

https://bugzilla.gnome.org/show_bug.cgi?id=658659
2011-09-12 07:37:10 -03:00