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Author SHA1 Message Date
Wim Taymans
63dbc75734 gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug and comments. 
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Add some debug and comments.
Fix double unref() in error cases.
 2009-08-11 02:30:24 +01:00
Wim Taymans
9bfc641f0d gst/rtpmanager/gstrtpbin.*: Add debugging category. 
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
(create_session), (find_stream_by_ssrc), (create_stream),
(gst_rtp_bin_class_init), (new_payload_found),
(new_ssrc_pad_found), (create_recv_rtp), (create_recv_rtcp),
(create_send_rtp), (create_rtcp):
* gst/rtpmanager/gstrtpbin.h:
Add debugging category.
Added RTPStream to manage stream per SSRC, each with its own
jitterbuffer and ptdemux.
Added SSRCDemux.
Connect to various SSRC and PT signals and create ghostpads, link stuff.
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
Added rtpbin to elements.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Fix caps and forward GstFlowReturn
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
(gst_rtp_session_request_new_pad):
Add debug category.
Add event handling
* gst/rtpmanager/gstrtpssrcdemux.c: (find_rtp_pad_for_ssrc),
(create_rtp_pad_for_ssrc), (gst_rtp_ssrc_demux_class_init),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.h:
Add debug category.
Add new-pt-pad signal.
 2009-08-11 02:30:24 +01:00
Wim Taymans
a9d14ed310 gst/rtpmanager/: Added simple SSRC demuxer. 
Original commit message from CVS:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpssrcdemux.c: (find_pad_for_ssrc),
(create_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init),
(gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_init),
(gst_rtp_ssrc_demux_finalize), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_src_event),
(gst_rtp_ssrc_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.h:
Added simple SSRC demuxer.
 2009-08-11 02:30:23 +01:00
Wim Taymans
5351f0cb51 gst/rtpmanager/: Some more ghostpad magic. 
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
(create_session), (gst_rtp_bin_base_init), (create_recv_rtp),
(create_recv_rtcp), (create_send_rtp), (create_rtcp),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
Some more ghostpad magic.
 2009-08-11 02:30:23 +01:00
Wim Taymans
fdae491de7 gst/rtpmanager/Makefile.am: Add .h file so it can be disted properly. 
Original commit message from CVS:
* gst/rtpmanager/Makefile.am:
Add .h file so it can be disted properly.
 2009-08-11 02:30:23 +01:00
Wim Taymans
f0d1ab1c1f Add RTP session management elements. Still in progress. 
Original commit message from CVS:
* configure.ac:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_new),
(signal_waiting_threads), (async_jitter_queue_ref),
(async_jitter_queue_ref_unlocked),
(async_jitter_queue_set_low_threshold),
(async_jitter_queue_set_high_threshold),
(async_jitter_queue_set_max_queue_length),
(async_jitter_queue_get_g_queue), (calculate_ts_diff),
(async_jitter_queue_length_ts_units_unlocked),
(async_jitter_queue_unref_and_unlock), (async_jitter_queue_unref),
(async_jitter_queue_lock), (async_jitter_queue_unlock),
(async_jitter_queue_push), (async_jitter_queue_push_unlocked),
(async_jitter_queue_push_sorted),
(async_jitter_queue_push_sorted_unlocked),
(async_jitter_queue_insert_after_unlocked),
(async_jitter_queue_pop_intern_unlocked), (async_jitter_queue_pop),
(async_jitter_queue_pop_unlocked), (async_jitter_queue_length),
(async_jitter_queue_length_unlocked),
(async_jitter_queue_set_flushing_unlocked),
(async_jitter_queue_unset_flushing_unlocked),
(async_jitter_queue_set_blocking_unlocked):
* gst/rtpmanager/async_jitter_queue.h:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(gst_rtp_bin_class_init), (gst_rtp_bin_init),
(gst_rtp_bin_finalize), (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property), (gst_rtp_bin_change_state),
(gst_rtp_bin_request_new_pad), (gst_rtp_bin_release_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c: (new_pad), (create_stream),
(free_stream), (find_stream_by_ssrc), (gst_rtp_client_base_init),
(gst_rtp_client_class_init), (gst_rtp_client_init),
(gst_rtp_client_finalize), (gst_rtp_client_set_property),
(gst_rtp_client_get_property), (gst_rtp_client_change_state),
(gst_rtp_client_request_new_pad), (gst_rtp_client_release_pad):
* gst/rtpmanager/gstrtpclient.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_base_init),
(gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
(gst_rtp_jitter_buffer_dispose), (gst_rtp_jitter_buffer_getcaps),
(gst_jitter_buffer_sink_setcaps), (free_func),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_src_activate_push),
(gst_rtp_jitter_buffer_change_state), (priv_compare_rtp_seq_lt),
(compare_rtp_buffers_seq_num), (gst_rtp_jitter_buffer_sink_event),
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_base_init),
(gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_init),
(gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_chain),
(gst_rtp_pt_demux_getcaps), (find_pad_for_pt),
(gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
(gst_rtp_pt_demux_change_state):
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (gst_rtp_session_change_state),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
(gst_rtp_session_request_new_pad), (gst_rtp_session_release_pad):
* gst/rtpmanager/gstrtpsession.h:
Add RTP session management elements. Still in progress.
 2009-08-11 02:30:23 +01:00
Mark Nauwelaerts
96e72522fc avidemux: push mode; cater for chunk padding 2009-08-10 14:41:52 +02:00
Mark Nauwelaerts
f67db2a089 avidemux: only use stream's pad after having checked it exists 2009-08-10 14:41:34 +02:00
Mark Nauwelaerts
4249f52c6c avidemux: sprinkle some more GST_DEBUG_FUNCPTR 2009-08-10 14:41:29 +02:00
Mark Nauwelaerts
6d26594eef avidemux: post error message if no pads to push EOS event on 2009-08-10 14:41:27 +02:00
Mark Nauwelaerts
b0a0c06155 avidemux: fix typo in warning message 2009-08-10 14:41:23 +02:00
Mark Nauwelaerts
7750173244 avidemux: fix some buffer ref handling 2009-08-10 14:41:19 +02:00
Mark Nauwelaerts
5b0f7f04e7 avidemux: do not exceed maximum number of supported streams 2009-08-10 14:41:16 +02:00
Mark Nauwelaerts
effa7b4660 avidemux: prevent double unref; gst_avi_demux_parse_avih already unrefs 2009-08-10 14:41:14 +02:00
Mark Nauwelaerts
42bc085d95 avidemux: verify size of INFO LIST to satisfy subsequent expectations 2009-08-10 14:41:12 +02:00
Mark Nauwelaerts
f4f8e8532c avidemux: check video stream framerate against avi header frame duration 
The former might be bogus in silly cases, and the latter seems to
carry more weight.
 2009-08-10 14:41:09 +02:00
Mark Nauwelaerts
3863871100 avidemux: streamline stream duration calculation 2009-08-10 14:41:07 +02:00
Edward Hervey
d522f94f98 dv1394src: Fix element for live usage... which has been broken for 2 years :( 
This is a live source, therefore:
* Use GST_FORMAT_TIME as the default format
* set_timestamp to True
* properly implement query latency.

This allows expected live usage like : playbin2 uri=dv://
 2009-08-10 09:58:34 +02:00
Edward Hervey
3fd4cdcc43 raw1394: Remove unneeded variable 2009-08-10 09:58:34 +02:00
Edward Hervey
d29ba8d48f matroska: remove dead assignments 2009-08-10 09:58:33 +02:00
Edward Hervey
0d6f0801f5 rtp: Remove dead assignments and resulting unneeded variables. 2009-08-10 09:58:33 +02:00
Sebastian Dröge
153ae910a0 wavpack: Use GLib GChecksum instead of our own MD5 implementation 
This requires GLib 2.16 but that version is already required by core anyway.
 2009-08-10 09:54:16 +02:00
Thiago Santos
08862850a7 matroska: Adds support to muxing/demuxing WMA 
Adds support for muxing wma audio family and fixes
demuxing of wma family in matroskademux. matroskademux
was broken because it missed codec_data.
 2009-08-09 20:34:05 -03:00
Thiago Santos
df442b4727 matroskamux: adds support for wmv family 
Adds support to WMV1, WMV2, WMV3 and other family formats that
are signaled by the 'format' field in the caps (i.e. WVC1).
Partially fixes #576378
 2009-08-09 20:34:04 -03:00
Tim-Philipp Müller
8c8e6af45b v4l2src: if max == min width/height put an int in the probed caps, not an int range 
Fixes #560033.
 2009-08-09 14:19:42 +01:00
Tim-Philipp Müller
6df8fb76ef osxaudiosrc: if max_channels == min_channels, use an int instead of an int range in the caps 2009-08-09 13:58:07 +01:00
LoneStar
494555cecd id3demux: Try GST_*_TAG_ENCODING and locale encoding if tags are not UTF8 
Fixes bug #499242.
 2009-08-09 12:52:17 +02:00
Tim-Philipp Müller
78626d4db2 configure: bump core/base requirements to latest release 
To avoid confusion.
 2009-08-09 01:30:51 +01:00
Tim-Philipp Müller
04efc92897 check: fix flvmux unit test on big endian machines 
flvmux only accepts raw audio in little endian, but audiotestsrc
produces audio in the native endianness, which makes linking
between audiotestsrc and flvmux fail on big endian machines. Add
an audioconvert element in between the two to fix this.
 2009-08-09 01:28:40 +01:00
Vincent Penquerc'h
19b7001bf9 matroska: add kate subtitle support to matroska muxer and demuxer 
See #525743.
 2009-08-08 12:54:48 +01:00
Tim-Philipp Müller
b0bcb27517 id3demux: add ID3 v2.3 spec as well 2009-08-07 16:51:45 +01:00
Tim-Philipp Müller
0417283077 id3demux: sizes in ID3 v2.3 are unlikely to be sync-safe integers 
In ID3 v2.3 compressed frames will have a 4-byte data length indicator
after the frame header to indicate the size of the decompressed data.
This integer is unlikely to be a sync-safe integer for v2.3 tags,
only in v2.4 it's sync-safe.
 2009-08-07 16:42:39 +01:00
Tim-Philipp Müller
3ec0a31d64 id3demux: fix typo in debug message 2009-08-07 16:36:55 +01:00
Tim-Philipp Müller
2e05af3876 id3demux: fix parsing of unsync'ed ID3 v2.4 tags and frames 
Reversing the unsynchronisation seems to work slightly differently
for ID3 v2.3 tags and v2.4 tags: v2.3 tags don't have syncsafe frame
sizes in the frame header, so the unsynchronisation is applied to
the whole frame data including all the frame headers. v2.4 frames
have sync-safe sizes, however, so the unsynchronisation only needs
to be applied to the actual frame data, and it seems that's what's
being done as well. So we need to undo the unsynchronisation on a
per-frame basis for v2.4 tags for things to work properly.

Fixes extraction of coverart/images from APIC frames in ID3 v2.4
tags (#588148).

Add unit test for this as well.
 2009-08-07 16:02:23 +01:00
Sebastian Dröge
c42f0ad5b6 souphttpsrc: Use SOUP_METHOD_GET instead of "GET" string 
Fixes bug #590970.
 2009-08-06 21:24:14 +02:00
Wim Taymans
b32ef1d51e pulsesrc: set the default slave method to skew 
Set the default slave method to the much better skew algorithm. This is the
default in the new base class but we override this here as well for the
upcomming release.
 2009-08-06 13:03:13 +02:00
Tim-Philipp Müller
1425c46e20 pulsesrc: fix compilation with --disable-gst-debug 2009-08-06 10:21:38 +01:00
Wim Taymans
ddfa9961c6 rtph264pay: use array instead of queue 2009-08-06 11:14:44 +02:00
Mark Nauwelaerts
2bfb42c5f8 rtph264pay: push NALs only after SPS/PPS 
parse complete (bytestream) buffer for SPS/PPS before pushing NALs.

Fixes #564501.
 2009-08-06 11:14:44 +02:00
Sebastian Dröge
198604c108 v4l2: Directly use GST_PTR_FORMAT for printing caps with the LOG_CAPS macro 2009-08-04 14:45:31 +02:00
Edward Hervey
d65d542e9d rtpqdm2depay: Fix debug statement. 2009-08-04 11:17:17 +02:00
Sebastian Dröge
23dbb15ff5 v4l2: Remove some OMAP specific hacks 
They require special build flags and are not useful in general.
 2009-08-04 09:32:07 +02:00
Rob Clark
99e2ac121d v4l2sink: change where buffers get dequeued 
It seems to cause strange occasional high latencies (almost 200ms) when dequeuing buffers from _buffer_alloc().  It is simpler and seems to work much better to dqbuf from the same thread that is queuing the next buffer.
 2009-08-04 09:22:29 +02:00
Rob Clark
f19cfbda96 v4l2: Add v4l2sink element 
This also does the following changes:
(1) pull the bufferpool code out into gstv4l2bufferpool.c, and make a
    bit more generic so it can be used both for v4l2src and v4l2sink
(2) move some of the device probing/configuration/caps stuff into
    gstv4l2object.c so it does not have to be duplicated between
    v4l2src and v4l2sink

Fixes bug #590280.
 2009-08-04 09:16:56 +02:00
Sebastian Dröge
56850099a6 flvmux: Enable unit test now that it passes 2009-08-04 07:08:45 +02:00
Edward Hervey
20c7977b9b rtpqdm2depay,rtpsv3vdepay: Add debugging category. 2009-08-03 21:26:31 +02:00
Edward Hervey
25c5514fab rtpqdm2depay: Handle gaps in incoming packets. 
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 :)
 2009-08-03 21:26:30 +02:00
Edward Hervey
6aff520a24 rtpqdmdepay: Fix CRC calculation and remove commented code. 2009-08-03 21:26:30 +02:00
Edward Hervey
d39c057e42 rtp: New QDM2 rtp depayloader. 
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).
 2009-08-03 21:26:30 +02:00
Edward Hervey
e2b3665ae6 rtpsv3vdepay: Properly fill codec_data and cleanup code a bite more. 2009-08-03 21:26:30 +02:00