Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtsp.c: (plugin_init):
* gst/rtsp/gstrtspgoogle.c:
* gst/rtsp/gstrtspgoogle.h:
Remove google extension again, it's not needed anymore because we never
send multiple transports anymore.
Original commit message from CVS:
Based on patch by: Eric Zhang <chao.zhang at access-company dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_nat_method_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_set_property),
(gst_rtspsrc_get_property), (gst_rtspsrc_create_stream),
(gst_rtspsrc_stream_free),
(gst_rtspsrc_stream_configure_udp_sinks),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_send_dummy_packets),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add property to configure NAT traversal method.
Ignore EOS from the internal sinks.
Implement sending dummy packets as a (simple) method to open up
some firewalls.
Send PLAY request to the server after we started the udp sources.
Fixes#559545.
Original commit message from CVS:
Patch by: Yotam <sh dot yotam at gmail dot com>
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_event):
Flush the remaining frames on EOS. Fixes#560641.
Original commit message from CVS:
* gst/qtdemux/qtdemux.h (struct _GstQTDemux):
* gst/qtdemux/qtdemux.c (gst_qtdemux_do_seek): Queue up new
segment events instead of sending them from the seeking thread.
Fixes#559288.
(gst_qtdemux_push_pending_newsegment): New helper, sends out
queued newsegment events.
(gst_qtdemux_loop_state_movie): Voilà, call it here. Only need to
call it here, as we only seek when looping, and only push in the
movie state.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_tag_add_tmpo),
(qtdemux_tag_add_covr), (qtdemux_parse_udta):
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c:
Add cover and alternative copyright tag, and enhance some existing
ones by marking them as container atoms.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_transports_string),
(gst_rtspsrc_change_state):
Only send one transport at a time for improved compatibility with some
broken servers. See #537832.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
(gst_rtspsrc_perform_seek):
Only pause/play in the seek handler when the source was playing.
Fixes#529379.
Original commit message from CVS:
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_handle_dirac_packet):
Fix muxing of Dirac streams if the input already has the format
we need, i.e. is the output of matroskademux.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak),
(qtdemux_video_caps), (qtdemux_audio_caps):
Refactor some raw audio caps building, and handle >16-bit cases.
Fix/replace building caps from a string description.
Original commit message from CVS:
* gst/audiofx/audiowsincband.c:
* gst/audiofx/audiowsinclimit.c:
* gst/cutter/gstcutter.c:
Make author name consistent with others.
Original commit message from CVS:
Based on patch by: Eric Zhang <chao.zhang at access-company dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_perform_seek),
(gst_rtspsrc_stream_configure_udp_sink):
Pause the RTSP stream before doing a new play request.
Make sure that adding the udpsinks does not cause the rtspsrc to become
a sink. Fixes#559547.
Original commit message from CVS:
* gst/matroska/matroska-ids.h:
* gst/matroska/matroska-mux.c: (gst_matroska_pad_free),
(gst_matroska_mux_handle_dirac_packet),
(gst_matroska_mux_write_data):
Implement Dirac muxing into Matroska comforming to the spec, i.e.
put all Dirac packages up to a picture into a Matroska block.
TODO: Implement writing of the ReferenceBlock Matroska elements,
currently the Dirac muxing is only 100% correct if Matroska version 2
is selected for muxing.
Original commit message from CVS:
* ext/flac/Makefile.am:
Include $(FLAC_CFLAGS) in CFLAGS to make sure to find the FLAC headers.
This fixes compilation if FLAC is installed in an uncommon location
that is not already handled by other CFLAGS. Fixes bug #558711.
Original commit message from CVS:
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_format_get_rank):
Guard more uncommon formats with ifdefs so that we can compile on older
versions.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
(gst_wavenc_sink_setcaps), (gst_wavenc_change_state):
* gst/wavenc/gstwavenc.h:
Add support for float/double as input and remove the (nowadays)
useless parsing of the depth as we require width==depth.
Original commit message from CVS:
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_setcaps):
* gst/rtp/gstrtpmpapay.c:
Narrow down the caps of the mpeg audio pay/depayloaders to only accept
mpeg version 1. Fixes#558427.
Original commit message from CVS:
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_flush),
(gst_rtp_L16_pay_getcaps):
Only put an integral amount of samples in the RTP packet.
Fixes#556641.
Original commit message from CVS:
* gst/rtp/gstrtpchannels.c: (gst_rtp_channels_get_by_index):
* gst/rtp/gstrtpchannels.h:
Add method to get possible channel positions.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
Don't allow width=32,depth=24 as input. WAV requires that the width
is the next integer multiply of 8 from the depth.
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps):
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps),
(gst_rtp_L16_pay_getcaps):
* gst/rtp/gstrtpchannels.c: (check_channels),
(gst_rtp_channels_get_by_pos), (gst_rtp_channels_get_by_order),
(gst_rtp_channels_create_default):
* gst/rtp/gstrtpchannels.h:
Add mappings for multichannel support. Does not completely just work
because the getcaps function does not yet return the allowed channel
mappings. See #556641.
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps),
(gst_rtp_L16_depay_process):
Check if clock-rate and channels are valid.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_setcaps),
(gst_rtp_ac3_depay_process):
Don't ignore the return value of set_caps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
(gst_rtp_amr_depay_process):
* gst/rtp/gstrtpamrdepay.h:
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set output caps on the buffers, the base class does that for
us.
The subclass will make sure we are negotiated.
* gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_setcaps),
(gst_rtp_dv_depay_process), (gst_rtp_dv_depay_reset):
* gst/rtp/gstrtpdvdepay.h:
Clean up caps negotiation.
The subclass will make sure we are negotiated.
* gst/rtp/gstrtpg726depay.c: (gst_rtp_g726_depay_setcaps),
(gst_rtp_g726_depay_process):
Clean up caps negotiation.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_init),
(gst_rtp_g729_depay_setcaps), (gst_rtp_g729_depay_process):
* gst/rtp/gstrtpg729depay.h:
The subclass will make sure we are negotiated.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_setcaps),
(gst_rtp_gsm_depay_process):
Clean up caps negotiation.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_setcaps):
Clean up caps negotiation.
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_setcaps),
(gst_rtp_h263_depay_process):
Clean up caps negotiation.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_setcaps),
(gst_rtp_h263_pay_flush), (gst_rtp_h263_pay_handle_buffer):
* gst/rtp/gstrtph263pay.h:
Don't ignore the return value of set_outcaps.
Do some more timestamps.
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
(gst_rtp_h263p_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init),
(gst_rtp_h263p_pay_setcaps), (gst_rtp_h263p_pay_flush),
(gst_rtp_h263p_pay_handle_buffer):
* gst/rtp/gstrtph263ppay.h:
Don't ignore the return value of set_outcaps.
Do some more timestamps.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps),
(gst_rtp_h264_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
Fix possible caps leak.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_setcaps):
Add some more debug info.
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps),
(gst_rtp_ilbc_depay_process):
Clean up caps negotiation.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_sink_setcaps):
Clean up caps negotiation.
* gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_setcaps),
(gst_rtp_mp1s_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set caps on buffers, subclass does that for us.
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps),
(gst_rtp_mp2t_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set caps on buffers, subclass does that for us.
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps),
(gst_rtp_mp4a_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_new_caps),
(gst_rtp_mp4a_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_setcaps),
(gst_rtp_mp4g_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set caps on buffers, subclass does that for us.
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize),
(gst_rtp_mp4g_pay_new_caps), (gst_rtp_mp4g_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
(gst_rtp_mp4v_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set caps on buffers, subclass does that for us.
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_new_caps),
(gst_rtp_mp4v_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_setcaps),
(gst_rtp_mpa_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_setcaps),
(gst_rtp_mpv_depay_process):
Clean up caps negotiation.
Actually set output caps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtpmpvpay.c: (gst_rtp_mpv_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps),
(gst_rtp_pcma_depay_process):
Clean up caps negotiation.
Set output buffer duration because we can.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps),
(gst_rtp_pcmu_depay_process):
Clean up caps negotiation.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_init),
(gst_rtp_speex_depay_setcaps), (gst_rtp_speex_depay_process):
Clean up caps negotiation.
Set output caps on the pad and header buffers.
Set duration on output buffers because we can.
* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_parse_ident):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_setcaps),
(gst_rtp_sv3v_depay_process):
Clean up caps negotiation.
No need to validate the buffer, the base class does that for us.
No need to set caps out output buffers, subclass does that.
* gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps),
(gst_rtp_theora_depay_process):
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_class_init),
(gst_rtp_theora_pay_flush_packet), (encode_base64),
(gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_parse_id),
(gst_rtp_theora_pay_handle_buffer):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps),
(gst_rtp_vorbis_depay_process):
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpvrawdepay.c: (gst_rtp_vraw_depay_setcaps):
Clean up caps negotiation, don't ignore setcaps return.
* gst/rtp/gstrtpvrawpay.c: (gst_rtp_vraw_pay_setcaps):
Don't ignore the return value of set_outcaps.
Original commit message from CVS:
* tests/check/elements/icydemux.c: (icydemux_found_pad):
Add some refcount check
* tests/check/elements/rtp-payloading.c: (rtp_pipeline_run):
Don't ignore the result of write(), fixes a compiler warning for me.
* tests/icles/videobox-test.c: (main):
Make the output a little more pretty.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.interfaces:
* docs/plugins/inspect/plugin-autodetect.xml:
Add the docs of the new elements.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosrc.c:
(gst_auto_audio_src_class_init):
* gst/autodetect/gstautovideosrc.c:
(gst_auto_video_src_class_init):
Fix "Since" tags in the documentation.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_class_init),
(gst_soup_http_src_set_property), (gst_soup_http_src_get_property):
Add support for souphttpsrc to act as a live source. This makes it
possible to get timestamped buffers in combination with the
"do-timestamp" property. Fixes bug #556019.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/matroska/matroska-mux.c: (gst_matroska_mux_reset),
(gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad):
Fix a memory leak when pads are requested but the pipeline never
goes into PLAYING.
Correctly remove request pads, no matter if they have collected
data or not.
Fixes bug #557710.
Original commit message from CVS:
Patch by: <lrn1986 at gmail dot com>
* gst/udp/gstudpnetutils.h:
Define the correct WINVER so getaddinfo() can be used when using
mingw32. Fixes bug #557294.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c: (update_coefficients):
Don't calculate the filter coefficients for every single buffer
but only when it's needed. Fixes bug #557260.
Original commit message from CVS:
* gst/avi/gstavimux.c:
Fix VPRP chunk setup in avimux.
Fixes: #556010
Patch By: Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
Original commit message from CVS:
* gst/videobox/gstvideobox.c:
support dynamically changing properties in videobox
Fixed: #557085
Patch By: Wim Taymans <wim.taymans@collabora.co.uk>
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_scan):
Skip entries for streams that don't have a output pad yet, thereby
avoiding calling pad functions with a NULL pad.
Fixes#556424
Original commit message from CVS:
* ext/flac/gstflacdec.c (gst_flac_dec_read_stream):
* ext/flac/gstflacenc.c (gst_flac_enc_write_callback):
Cast some size_t arguments to guint to avoid compiler
warnings on 64-bit systems.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event):
Return TRUE instead of FALSE from the event handler when we swallowed the
event.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex),
(gst_avi_demux_parse_stream), (gst_avi_demux_parse_index):
* gst/avi/gstavidemux.h:
For timestamping audio packets we need to take into account the
amount of blocks in one entry using the blockalign. Fixes some sync
issues with zero-padded audio blocks in the beginning of avi files.
Original commit message from CVS:
* gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_class_init),
(gst_multi_file_src_query):
Implement DEFAULT and BUFFER position queries. See #555260.
Original commit message from CVS:
* tests/examples/rtp/client-H263p.sdp:
* tests/examples/rtp/client-H263p.sh:
* tests/examples/rtp/server-VTS-H263p.sh:
Add some more H263p server and client examples.
Original commit message from CVS:
* ext/pulse/pulsesink.c: (gst_pulsesink_write):
* ext/pulse/pulsesrc.c: (gst_pulsesrc_read):
Return -1 instead of 0 in error cases. Fixes#554771.
Original commit message from CVS:
* sys/ximage/gstximagesrc.c: (gst_ximage_src_start),
(gst_ximage_src_stop), (gst_ximage_src_ximage_get):
Stop leaking the cursor image.
Unref the last_ximage and the cached cursor image on shutdown.
Fixes#551570.
Original commit message from CVS:
* sys/v4l2/gstv4l2object.h:
Getting the Class from an instance is not just a matter of casting it to
the class struct but it involves calling G_OBJECT_GET_CLASS on the
instance. Fixes#549784.
Original commit message from CVS:
* configure.ac:
Fix libs for linking directsound.
* sys/directsound/gstdirectsoundsink.c:
Fix buffer sizing to prevent racing the ringbuffer at startup.
Add volume property.
Original commit message from CVS:
* ext/pulse/pulsesink.c:
Fix problems with pulsesink randomly erroring with code 'OK' after a
format change on the stream by waiting when disconnecting the stream.
Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init),
(gst_rtp_amr_depay_process):
Mark DISCONT on output buffers when the marker bit signals a new talk
spurt.
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
Set the marker bit for buffers with a DISCONT flag to signal a talk
spurt.
Original commit message from CVS:
* gst/videomixer/videomixer.c: (gst_videomixer_fill_queues),
(gst_videomixer_sink_event):
Handle segments a little better. Fixes#537361.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods):
Don't assume the server supports PAUSE by default. Fixes#551048.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_set_uri), (gst_udpsrc_start):
Switch on the socket family to get the addrlen size right.
Original commit message from CVS:
Patch by: Daniel Franke <df at dfranke dot us>
* gst/udp/gstudpsrc.c: (gst_udpsrc_create), (gst_udpsrc_start):
OS X's bind() implementation is picky about its addrlen parameter and
fails with EINVAL if it is larger than expected for the socket's address
family. Set the length to the expected length instead. Fixes#553191.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
Handle the case where we cannot do desribe or when the describe result
does not contain a valid SDP message.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_header),
(gst_qtdemux_chain):
Some 'broken' files out there have atom lengths of zero...
which basically results in qtdemux consuming that atom again and again
until the *end of night* !
Detect that and emits an adequate element error message.
Original commit message from CVS:
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_init),
(gst_rtp_mp4g_depay_finalize), (gst_rtp_mp4g_depay_setcaps),
(gst_rtp_mp4g_depay_clear_queue), (gst_rtp_mp4g_depay_flush_queue),
(gst_rtp_mp4g_depay_queue), (gst_rtp_mp4g_depay_process),
(gst_rtp_mp4g_depay_change_state):
* gst/rtp/gstrtpmp4gdepay.h:
Handle interleaved streams by reordering AU in a queue.
Original commit message from CVS:
* gst/rtp/gstrtpmp4gdepay.c: (gst_bs_parse_init),
(gst_bs_parse_read), (gst_rtp_mp4g_depay_process):
Change some of the ranges in the caps, mostly for the amount of bits we
can use.
Added a little bitstream parse and use it to parse the AU header fields.
Check for malformed and wrongly sized packets better.
Implement more header field parsing.
Handle the size of fragmented packets correctly.
Original commit message from CVS:
Patch by: Jonathan Matthew <notverysmart@gmail.com>
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
Add mapping for 'tiff' => image/tiff
Fixes#552213
Original commit message from CVS:
* ext/raw1394/gstdv1394src.c: (SEND_COMMAND):
* ext/raw1394/gsthdv1394src.c: (SEND_COMMAND):
Pretend to care about the result of write() which works around
compiler warnings.
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_class_init):
Make sure the desired default values are actually set, not only
registered as defaults (actual problem is that the stereo-specific
values are only updated if channels==2, which is not the case yet
when the object is created, so the default values for the
mid-side-stereo and loose-mid-side-stereo settings are never
set in _update_quality()). Makes flacenc create smaller files by
default (for stereo input), and fixes#550791.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_state_header), (qtdemux_parse_node),
(qtdemux_parse_trak), (qtdemux_video_caps):
* gst/qtdemux/qtdemux.h:
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c:
Add support for video/mj2 mime-type and its additional atoms/boxes.
Fixes#550646.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
Add mapping for IMA Loki SDL MJPEG ADPCM codec.
Add some alternative byteswapped mappings that seem to pop up sometimes.
Fixes#550288.
Original commit message from CVS:
* po/LINGUAS:
* po/POTFILES.in:
* po/POTFILES.skip:
Add 'ca' to LINGUAS; add some more files with translations and some
files which should be ignored by translation tools.
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data):
* ext/speex/gstspeexdec.h:
* ext/speex/gstspeexenc.c: (gst_speex_enc_encode):
* ext/speex/gstspeexenc.h:
Use integer encoding and decoding functions instead of converting
the integer input to float in the element. The libspeex integer
functions are doing this for us already or, if libspeex was compiled
in integer mode, they're doing everything using integer arithmetics.
Also saves some copying around.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset),
(gst_wavpack_enc_push_block), (gst_wavpack_enc_chain):
* ext/wavpack/gstwavpackenc.h:
Handle non-zero start timestamps and stream discontinuities
correctly. This only has an effect if we're muxing into
a container format as the raw WavPack stream must contain
continous sample numbers.
Original commit message from CVS:
* ext/speex/gstspeexenc.c: (gst_speex_enc_encode):
Correct the timestamp and granulepos calculation by one Speex
frame.
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data):
Correctly take the granulepos from upstream if possible and
correctly handle the granulepos in various calculations: the
granulepos is the sample number of the _last_ sample in a frame, not
the first.
* ext/speex/gstspeexenc.c: (gst_speex_enc_sinkevent),
(gst_speex_enc_encode), (gst_speex_enc_chain),
(gst_speex_enc_change_state):
* ext/speex/gstspeexenc.h:
Handle non-zero start timestamps in the encoder and detect/handle
stream discontinuities. Fixes bug #547075.
Original commit message from CVS:
Patch by: Craig Keogh <cskeogh at adam dot com dot au>
* ext/annodex/gstcmmlparser.c: (gst_cmml_parser_parse_chunk):
Fix compiler warnings caused by passing a string as format string
instead of "%s" and then the string. This is only exposed by -Wformat=2
as used by default on Ubuntu. Fixes bug #550015.
Original commit message from CVS:
Patch by: Mersad Jelacic <mersad at axis dot com>
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartmux.c: (gst_multipart_mux_get_mime):
Convert audio/x-adpcm to and from the audio/G726-X in the muxer and
demuxer. Fixes#549551.
Original commit message from CVS:
* sys/osxaudio/gstosxaudiosink.c:
(gst_osx_audio_sink_select_device):
* sys/osxaudio/gstosxaudiosrc.c:
(gst_osx_audio_src_create_ringbuffer),
(gst_osx_audio_src_select_device):
* sys/osxaudio/gstosxringbuffer.c: (gst_osx_ring_buffer_acquire):
Fix the build on macosx.
Original commit message from CVS:
* gst/icydemux/gsticydemux.c:
Small docs fix: in the example pipeline, we need to pass
iradio-mode=true to the source, so the server actually sends
an ICY stream.
Original commit message from CVS:
* sys/osxaudio/Makefile.am:
* sys/osxaudio/gstosxaudio.c:
* sys/osxaudio/gstosxaudiosink.c:
* sys/osxaudio/gstosxaudiosink.h:
* sys/osxaudio/gstosxaudiosrc.c:
* sys/osxaudio/gstosxaudiosrc.h:
* sys/osxaudio/gstosxringbuffer.c:
* sys/osxaudio/gstosxringbuffer.h:
Rewrite caps setting and ring buffer initialisation.
Previously we never told CoreAudio what format we were going to send it,
so it only worked due to luck, and not at all on some hardware.
Now we explicitly advertise what formats the hardware supports, and then
configure the selected one correctly.
Original commit message from CVS:
* sys/v4l2/gstv4l2object.c:
* sys/v4l2/gstv4l2src.c:
* sys/v4l2/gstv4l2src.h:
* sys/v4l2/v4l2_calls.c:
* sys/v4l2/v4l2src_calls.c:
Fix memory leaks. Small code cleanups : No need for empty _init(). No
need to memset instance structures. Some more FIXME's.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_send_event),
(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_video_pad_setcaps),
(gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_finish):
Add Real[Audio|Video] support to Matroska containers.
It works fine for:
* decoding real audio/video streams contained in mkv
* 'transmuxing' real (.rm) files into .mkv files
It will not work though for encoding real[audio/video] streams that
don't contain the 'mdpr_data' extra data on the caps.
The reason why this will not work is because I never intended to
duplicate virtually all the 'mdpr' block creation into mkvmux.
Fixes#536067
Original commit message from CVS:
* gst/law/alaw-encode.c: (gst_alaw_enc_init), (gst_alaw_enc_chain):
* gst/law/mulaw-conversion.c:
* gst/law/mulaw-encode.c: (gst_mulawenc_init),
(gst_mulawenc_chain):
The encoder can't really renegotiate at the time they perform a
pad-alloc so make the srcpads use fixed caps.
Check the buffer size after a pad-alloc because the returned size might
not be right when the downstream element does not know the size of the
new buffer (capsfilter). Fixes#549073.
Original commit message from CVS:
Patch by: Filippo Argiolas <filippo dot argiolas at gmail dot com>
* sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_set_norm_and_notify):
v4l2src doesn't have a property named "norm" so don't try to notify
about changes to that property. The "norm" property and related
code are commented out currently. Fixes bug #549090.
Original commit message from CVS:
Patch by: Mike Ruprecht <cmaiku at gmail dot com>
* sys/v4l2/gstv4l2object.c: (gst_v4l2_class_probe_devices):
Reprobe devices again instead of taking a cached list as new
devices could've been plugged in. Fixes bug #549062.
Original commit message from CVS:
* gst/autodetect/Makefile.am:
Don't link the autodetect plugin with GConf as it doesn't
use GConf. Fixes bug #545463.
Original commit message from CVS:
* gst/matroska/ebml-read.c: (gst_ebml_read_element_id),
(gst_ebml_read_element_length), (gst_ebml_read_uint),
(gst_ebml_read_sint), (gst_ebml_read_float),
(gst_ebml_read_header):
Change some GST_ELEMENT_ERRORs to GST_ERROR_OBJECT to make it
possible to ignore errors and not post any ERROR messages on
the bus.
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_contents):
Ignore any errors and not just EOS when parsing the contents of
a SeekHead. Errors here are usually caused by truncated files
and playback of the file works fine. Fixes playback of the
audio_only_chapter_seekbroken.mka file from the MPlayer samples
archive.
Original commit message from CVS:
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartmux.c:
Conform to RFC2046. audio/basic is mulaw 8000Hz mono.
Original commit message from CVS:
* sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_buffer_alloc,
gst_directdraw_sink_bufferpool_clear):
Fix two more buffer ref leaks.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas at tandberg com>
* sys/directdraw/gstdirectdrawsink.c:
(gst_directdraw_sink_show_frame):
Fix buffer ref leak.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
Revert the last commit. wavenc still supports width!=depth for 32 bit
width. Thanks Tim.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup_or_simpleblock):
If the duration of a block is unknown only use the timestamp for the
first lace and use GST_CLOCK_TIME_NONE as duration for the following
laces. Otherwise every lace has the same timestamp which leads to
various problems. Really fixes bug #548831.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
If we're not allowing width!=depth in wavenc we should also disable
the code that was added to support width!=depth.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
Don't calculate the default duration of a frame from the audio sampling
rate. This only works for raw audio if every frame contains a single
sample and results in broken buffer durations for other formats
if no specified default duration is given or the blocks have no
duration. Fixes bug #548831.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup_or_simpleblock):
Allow zero sized blocks instead of returning GST_FLOW_OK. Such blocks
are used for text/plain subtitles as a gap-filler in some files.
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_structure),
(gst_v4l2_get_caps_info):
Add S910 and PWC formats with a low priority.
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_format_get_rank),
(gst_v4l2src_probe_caps_for_format):
Add more debugging.
Original commit message from CVS:
* gst/rtsp/gstrtspgoogle.c:
Things that can happen when your brain is in google mode trying to
deal with their google rtsp server extensions and trying to type your
google mail account.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtsp.c: (plugin_init):
* gst/rtsp/gstrtspgoogle.c: (gst_rtsp_google_before_send),
(gst_rtsp_google_after_send), (gst_rtsp_google_get_transports),
(_do_init), (gst_rtsp_google_base_init),
(gst_rtsp_google_class_init), (gst_rtsp_google_init),
(gst_rtsp_google_finalize), (gst_rtsp_google_change_state),
(gst_rtsp_google_extension_init):
* gst/rtsp/gstrtspgoogle.h:
Add google RTSP extension, it can only handle udp and responds with
unsupported if we do anything else. Fixes#546465.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_pause):
Make transport setup code a bit better using GString.
Add some more debug.
Check for closed connections before doing anything on them.
Original commit message from CVS:
* ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init),
(gst_pulsesrc_create_stream), (gst_pulsesrc_negotiate),
(gst_pulsesrc_prepare):
* ext/pulse/pulseutil.c: (gst_pulse_gst_to_channel_map),
(gst_pulse_channel_map_to_gst):
* ext/pulse/pulseutil.h:
If downstream provides no channel layout and >2 channels should be
used use the default layout that pulseaudio chooses and also
add this layout to the caps. Fixes bug #547258.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink):
Don't try to configure RTCP back to the server when the server did not
give us a valid port number.
Original commit message from CVS:
* gst/videobox/gstvideobox.c: (gst_video_box_set_property):
Use new basetransform method to renegotiate. Fixes#544956.
* tests/icles/Makefile.am:
* tests/icles/videobox-test.c: (make_pipeline), (main):
Add videobox renegotiation example.
Original commit message from CVS:
* ext/pulse/pulsesink.c: (gst_pulsesink_prepare):
* ext/pulse/pulsesrc.c: (gst_pulsesrc_prepare):
The bytes_per_sample and silence_sample fields of the GstRingBufferSpec
are already filled with the correct values by
gst_ring_buffer_parse_caps() so there's no need to set them again
with wrong values.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
(gst_avi_demux_read_subindexes_push):
Some AVI 2.0 (ODML) files don't respect the 'specifications' completely
and instead of using the 'ix##' nomenclature, use '##ix'.
They're still valid though, this fixes the duration and indexes for
virtually all the ODML files I have.
Original commit message from CVS:
* ext/pulse/pulsesink.c: (gst_pulsesink_class_init),
(gst_pulsesink_init), (gst_pulsesink_finalize),
(gst_pulsesink_set_volume), (gst_pulsesink_get_volume),
(gst_pulsesink_set_property), (gst_pulsesink_get_property),
(gst_pulsesink_prepare), (gst_pulsesink_change_state):
* ext/pulse/pulsesink.h:
Add "device-name" property to pulsesink too and currently commented
out and not working support for a "volume" property.
Original commit message from CVS:
Patch by: Laszlo Pandy <laszlok2 at gmail dot com>
* ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init),
(gst_pulsesrc_get_property):
Add "device-name" property, which provides a human readable string
for the audio device, to make it more consisten with other audio
sources. Fixes bug #547519.
Original commit message from CVS:
* ext/pulse/pulsemixer.c: (gst_pulsemixer_change_state):
* ext/pulse/pulsemixerctrl.c: (gst_pulsemixer_ctrl_subscribe_cb),
(gst_pulsemixer_ctrl_open), (gst_pulsemixer_ctrl_new),
(gst_pulsemixer_ctrl_free), (gst_pulsemixer_ctrl_timeout_event):
* ext/pulse/pulsemixerctrl.h:
* ext/pulse/pulseprobe.c: (gst_pulseprobe_open),
(gst_pulseprobe_enumerate), (gst_pulseprobe_new),
(gst_pulseprobe_free), (gst_pulseprobe_needs_probe),
(gst_pulseprobe_probe_property), (gst_pulseprobe_get_values):
* ext/pulse/pulseprobe.h:
* ext/pulse/pulsesink.c: (gst_pulsesink_init):
* ext/pulse/pulsesrc.c: (gst_pulsesrc_init), (gst_pulsesrc_delay),
(gst_pulsesrc_change_state):
Improve debugging a bit by including the parent object in pulsemixerctrl
and pulseprobe objects and using GST_WARNING_OBJECT instead of
GST_WARNING.
Use the parent GObject subclass instead of a random struct as GObject
parameter for G_OBJECT_WARN_INVALID_PROPERTY_ID. This fixes a crash
when probing for another property than "device".
Original commit message from CVS:
Patch by: Laszlo Pandy <laszlok2 at gmail dot com>
* ext/pulse/pulsemixer.c: (gst_pulsemixer_set_property):
Fix property probing after the device property is set by calling
set_server when the server property changes. Fixes bug #547518.
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_query):
Properly set the maximum latency value, in the same way it is done in
v4lsrc.
* sys/v4l2/v4l2src_calls.c:
Simplify fraction equality check, no need to use GValues for this.
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_query):
Add warning messages stating exactly why the latency query failed.
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_capture):
In some cases, the negotiated framerate might be the default one which
is already set internally. But we still need to mark it down in fps_n
and fps_d so that the latency query can happen properly.
Original commit message from CVS:
* docs/plugins/inspect/plugin-1394.xml:
Whoops, forgot one doc file for people who can't/don't build the
raw1394 plugin.
