Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams),
(gst_rtspsrc_dup_printf):
Give meaningfull error when all streams failed to configure for some
reason.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_loop),
(gst_wavparse_chain):
Don't push EOS from the chain function, the element
driving the pipeline is responsible for this. The bug
this was meant to fix seems to be queue not forwarding
EOS in all cases (see #476514).
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_class_init), (gst_level_start),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Use basetransform segment so that it is correctly managed on flushes and
start/stop.
Report message timestamp as stream time, which is what an application
can understand.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_eos),
(gst_wavparse_loop), (gst_wavparse_chain):
Add EOS logic for the push-based mode too. Fixes#476514.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create):
Make udpsrc timestamp outgoing buffers based on when they were received.
Also make it output a segment in time.
Original commit message from CVS:
Patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
* gst/rtp/gstrtph263pay.c:
Fix up header structure so that compilers don't add padding
between the structure fields, since that would lead to us
sending RTP packets with broken headers (as is currently the
case when compiling with MSVC). Also see similar fixes in
libgstrtp in gst-plugins-base. (#474616; #471194)
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_start),
(gst_spectrum_transform_ip):
Use the correct parameter order for the memset calls.
Thanks to Christian Schaller for noticing.
Original commit message from CVS:
* configure.ac:
* gst/spectrum/Makefile.am:
* gst/spectrum/demo-audiotest.c: (draw_spectrum),
(message_handler), (main):
* gst/spectrum/demo-osssrc.c: (draw_spectrum), (message_handler):
* gst/spectrum/gstspectrum.c: (gst_spectrum_base_init),
(gst_spectrum_class_init), (gst_spectrum_init),
(gst_spectrum_dispose), (gst_spectrum_set_property),
(gst_spectrum_get_property), (gst_spectrum_start),
(gst_spectrum_setup), (gst_spectrum_message_new),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Port GstSpectrum to GstAudioFilter and libgstfft, add support
for int32, float and double, use floats for the message contents,
average all FFTs done in one interval for better results, use
a better windowing function, allow posting the phase in the message
and actually do an FFT with the requested number of bands instead
of interpolating.
* tests/check/elements/spectrum.c: (GST_START_TEST),
(spectrum_suite):
Improve the units tests by checking for a 11025Hz sine wave
and add unit tests for all 4 supported sample types.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
* gst/wavparse/gstwavparse.h:
Implement seek-query. Refactor duration calculations. Appropriate use
of uint64_scale_int and uint64_scale. Move repeadedly calculated stuff
out of loops.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_dup_printf):
Use new basesink async property to make sparse RTCP packet not wait for
preroll.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_dup_printf),
(gst_rtspsrc_get_float), (gst_rtspsrc_play):
Make sure we generate and parse floating point values in the POSIX
locale instead of the current locale.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Fix method detection again.
Keep track of when we must send a Range header.
Use segment values for Range, Speed and Scale headers.
Parse Speed and Scale headers to update the segment values.
Original commit message from CVS:
* gst/audiofx/audiochebyshevfreqband.c:
* gst/audiofx/audiochebyshevfreqlimit.c:
Add small comparision with the windowed sinc filters in the docs.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_loop):
Refactor the udp and interleaved loop function a bit.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_try_send), (gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Protect connection activity with a new lock, avoids deadlocks when going
to PAUSED. Fixes#455808.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_set_property), (bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Use generator macros for the process functions for the different
sample types, add lower upper boundaries for the GObject properties
so automatically generated UIs can use sliders and change frequency
properties to floats to save a bit of memory, even ints would in
theory be enough. Also rename frequency to cutoff for consistency
reasons.
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
Regenerated for the above changes.
Original commit message from CVS:
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_class_init):
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_class_init):
Use generator macros for the process functions for the different
sample types, add lower upper boundaries for the GObject properties
so automatically generated UIs can use sliders and add a note about
the number of poles as a too high number of poles combined with
very low or very high frequencies will produce only noise.
* docs/plugins/gst-plugins-good-plugins.args:
Regenerated for the property changes.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_property),
(gst_rtspsrc_flush), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_parse_range),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Improve timeout handling.
Use the same socket for sending and receiving RTCP packets so that some
servers can track clients better.
Improve connection closed handling. Try to reconnect.
Don't overwrite our content base with NULL.
Improve debugging.
Improve range parsing and handling.
Remove flushing hack now that core does the right thing.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_init), (gst_multiudpsink_set_property),
(gst_multiudpsink_get_property), (gst_multiudpsink_init_send),
(gst_multiudpsink_close), (gst_multiudpsink_add):
* gst/udp/gstmultiudpsink.h:
Add support for getting and setting the socket to use.
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_create), (gst_udpsrc_get_property):
Add support for getting the currently used socket.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (process_32), (process_64),
(bpwsinc_build_kernel), (bpwsinc_push_residue),
(bpwsinc_transform), (bpwsinc_start), (bpwsinc_query),
(bpwsinc_query_type), (bpwsinc_event), (bpwsinc_set_property):
* gst/filter/gstbpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/bpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
* gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
Reset residue length only when actually creating a residue.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Improve UDP performance by avoiding a select() when we have data
available immediatly.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT),
(gst_rtp_dec_class_init):
* gst/rtsp/gstrtpdec.h:
Add (dummy) SSRC management signals.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(find_stream), (gst_rtspsrc_create_stream), (new_session_pad),
(request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc),
(on_timeout), (gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add connection-speed property.
Add find_stream helper functions.
Handle stream EOS based on BYE messages or SSRC timeout.
Returns SUCCESS from the state change function as we hide our async
elements from the parent.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (process_32), (process_64),
(lpwsinc_build_kernel), (lpwsinc_push_residue),
(lpwsinc_transform), (lpwsinc_start), (lpwsinc_query),
(lpwsinc_query_type), (lpwsinc_event), (lpwsinc_set_property):
* gst/filter/gstlpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
Original commit message from CVS:
* gst/debug/Makefile.am:
* gst/debug/breakmydata.c:
* gst/debug/gstdebug.c:
* gst/debug/negotiation.c:
* gst/debug/progressreport.c:
* gst/debug/rndbuffersize.c:
* gst/debug/testplugin.c:
Add new test element and clean-up the others a little.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel):
* gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
Improve debugging a bit.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_start):
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(lpwsinc_start):
Reset the residue in BaseTransform::start to get a clean residue
on stream changes.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (process_32), (process_64):
* gst/filter/gstlpwsinc.c: (process_32), (process_64):
Fix processing with buffer sizes that are larger than the filter
kernel size.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel):
Fix a segfault with more than one channel and don't rebuild
the kernel & residue with every buffer.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_mode_get_type),
(gst_bpwsinc_window_get_type), (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (bpwsinc_build_kernel), (bpwsinc_set_property),
(bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
Add support for a bandreject mode and allow specifying the window
function that should be used.
* gst/filter/gstlpwsinc.c:
And another small formatting fix.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (process_32), (process_64),
(bpwsinc_build_kernel), (bpwsinc_setup), (bpwsinc_get_unit_size),
(bpwsinc_transform), (bpwsinc_set_property),
(bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
Apply the same changes to the bandpass filter:
- Support double input
- Fix processing for input with >1 channels
- Specify frequency in Hz
- Specify actual filter kernel length
- Use transform instead of transform_ip as we're working
out of place anyway
- Factor out filter kernel generation and update the filter
kernel when the properties are set
Fix bandpass filter kernel generation to actually generate
a bandpass filter by creating a highpass instead of a second
lowpass.
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
Small formatting fix.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (process_32), (process_64),
(lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
Specify the actual filter length instead of a weird
2N+1. Setting the property will round to the next odd number.
Also remove now obsolete FIXMEs.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_window_get_type),
(gst_lpwsinc_class_init), (gst_lpwsinc_init),
(lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
Allow choosing between hamming and blackman window. The blackman
window provides a better stopband attenuation but a bit slower
rolloff.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (process_32), (process_64),
(lpwsinc_build_kernel):
Fix processing if the input has more than one channel.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_dispose),
(gst_bpwsinc_init), (bpwsinc_setup), (bpwsinc_transform_ip),
(bpwsinc_set_property), (bpwsinc_get_property):
"this" is a C++ keyword, use "self" instead.
Add TODOs and FIXMEs and remove two wrong FIXMEs.
* gst/filter/gstlpwsinc.c:
Add FIXMEs and a new TODO.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_dispose),
(gst_lpwsinc_class_init), (gst_lpwsinc_init), (process_32),
(process_64), (lpwsinc_build_kernel), (lpwsinc_setup),
(lpwsinc_get_unit_size), (lpwsinc_transform),
(lpwsinc_set_property), (lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
Add double support, replace "this" with "self" as the former
is a C++ keyword.
Implement the frequency property in Hz instead of fraction
of sampling frequency.
Remove some unecessary FIXMEs and add some TODOs, add some
required locking and refactor the kernel generation into a
separate function that is also called when the properties
change now.
And use BaseTransform::transform instead of transform_ip
as the convolution is done out of place anyway. Should
be done in place later.
Original commit message from CVS:
* gst/filter/Makefile.am:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_dispose),
(gst_bpwsinc_base_init), (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (bpwsinc_setup):
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_dispose),
(gst_lpwsinc_base_init), (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (lpwsinc_setup):
* gst/filter/gstlpwsinc.h:
Use GstAudioFilter as base class and don't leak the memory
of the filter kernel and residue.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (get_default_rate_for_pt),
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_udp_sink):
Fix default clock-rate for realmedia.
Fix parsing of transport.
Don't try to link NULL pads.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_movie):
If the buffer was entirely clipped ... don't try sending it :)
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_methods),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_prepare_transports):
If we don't hav a session manager, set the caps on outgoing buffers
ourselves.
Force PAUSE/PLAY methods for now until the extensions can overwrite.
Append final bit of the transport string even when it does not contain a
placeholder.
Original commit message from CVS:
* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_free),
(gst_rtsp_ext_list_connect):
* gst/rtsp/gstrtspext.h:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_send_cb):
Clean up the interface list.
Allow connecting to interface signals for the extensions.
Remove old extension code.
Free list on cleanup.
Allow extensions to send additional RTSP messages.
Original commit message from CVS:
* gst/audiofx/audioamplify.c: (gst_audio_amplify_init),
(gst_audio_amplify_setup), (gst_audio_amplify_transform_ip):
* gst/audiofx/audiodynamic.c:
(gst_audio_dynamic_set_process_function), (gst_audio_dynamic_init),
(gst_audio_dynamic_setup), (gst_audio_dynamic_transform_ip):
* gst/audiofx/audiodynamic.h:
* gst/audiofx/audioinvert.c: (gst_audio_invert_init),
(gst_audio_invert_setup), (gst_audio_invert_transform_ip):
* gst/audiofx/audioinvert.h:
Don't save format information ourselves, this is already saved in
GstAudioFilter.
Original commit message from CVS:
* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
(gst_rtsp_ext_list_stream_select):
* gst/rtsp/gstrtspext.h:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
Use rank to filter out extensions.
Add url to stream_select interface call.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
(gst_qtdemux_clip_buffer), (gst_qtdemux_loop_state_movie),
(qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps):
Clip raw audio and video when we can, keep track of current output
segment.
Don't leak buffers and events when there is no output pad.
Improve debugging here and there.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_child_proxy_get_child_by_index),
(gst_iir_equalizer_child_proxy_get_children_count),
(gst_iir_equalizer_child_proxy_interface_init),
(gst_iir_equalizer_class_init), (arg_to_scale), (setup_filter),
(gst_iir_equalizer_compute_frequencies):
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_class_init):
* gst/equalizer/gstiirequalizer3bands.c:
(gst_iir_equalizer_3bands_class_init):
* gst/equalizer/gstiirequalizernbands.c:
Better algorith for the center frequencies. Subtract band filters from
input for negative gains. Rework the gain mapping.
Original commit message from CVS:
* gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create):
Add example to the docs. Fix buffer-offset-end and add some debug.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
Fix parsing of esds atoms inside mp4a atoms so that we can set correct
codec_info for AAC audio. Fixes#457097 along with a whole other bunch
of qt/aac files.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
Don't return GST_FLOW_ERROR when pushing an event returns FALSE. We
don't have enough granularity to convert that boolean into a
GstFlowReturn.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps):
Set the encoding-name in the rtp caps to all uppercase, as required by
the caps spec.
Some small cleanups in the error paths. Fixes#453037.
Original commit message from CVS:
* gst/multifile/Makefile.am:
* gst/multifile/gstmultifile.c:
* gst/multifile/gstmultifilesink.c:
* gst/multifile/gstmultifilesink.h:
* gst/multifile/gstmultifilesrc.c:
* gst/multifile/gstmultifilesrc.h:
Add .h files to be able to add it to the docs.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(new_session_pad), (gst_rtspsrc_setup_streams):
* gst/rtsp/gstrtspsrc.h:
For container formats we only need to activate one of the streams so
that we correctly signal no-more-pads. Fixes#451015.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_samples),
(qtdemux_video_caps):
* gst/qtdemux/qtdemux_fourcc.h:
Add MJPG to the variants of motion jpeg.
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_close), (rtsp_connection_free):
Use threadsafe inet_ntop to convert an ip number to a string.
Fixes#447961.
Don't leak fd (and ip) when freeing a connection without first closing
it.
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_free):
Revert previous commit again, since we are frozen (sorry).
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_free):
inet_ntoa() uses a static buffer internally, so we need to copy the
returned string if we want to store it for later (#447961).
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect):
Fix the MingW build.
Patch By: Vincent Torri <vtorri at univ-evry dot fr>
Fixes: #446981
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
For AMR-NB streams, export the AMRSpecificBox as codec_data on the
caps.
Fixes#447458
Original commit message from CVS:
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
Make sure we allocate enough memory for the codec_data.
Fixes#447210.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps):
Add missing rate fields to caps. Fixes#441118.
Original commit message from CVS:
* gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420):
Printf fixes in debug statements; use LOG level for debug statements
that are printed for each and every frame; convert c++ comments to
C-style comments; not much point using g_try_malloc() if we then not
even check the return value.
Original commit message from CVS:
* configure.ac:
Bump requirements to released versions (core and base 0.10.13).
* gst/icydemux/gsticydemux.c: (gst_icydemux_unicodify):
Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
own implementation.
Original commit message from CVS:
* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
(gst_multipart_mux_get_mime), (gst_multipart_mux_collected):
Add support for mapping gst structure names to the MIME type equivalent.
Implemented for audio/x-mulaw->audio/basic. Fixes#442874.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
(gst_wavenc_sink_setcaps), (gst_wavenc_format_samples),
(gst_wavenc_chain), (gst_wavenc_change_state):
* gst/wavenc/gstwavenc.h:
Properly write wav files with width!=depth by having the depth most
significant bytes set and all others zero. Fixes#442535.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (add_date_header),
(rtsp_connection_send), (parse_response_status),
(parse_request_line), (parse_line), (rtsp_connection_receive):
* gst/rtsp/rtspdefs.c: (rtsp_version_as_text):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspmessage.c: (key_value_foreach),
(rtsp_message_init_request), (rtsp_message_init_response),
(rtsp_message_remove_header), (rtsp_message_append_headers),
(rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
Improves version checking, allowing an RTSP server to reply with "505
RTSP Version not supported.
Adds a Date header to all messages.
Replies with RTSP_EPARSE rather than RTSP_EINVALID in cases where we
want to be able to send a response even if something in the request was
invalid. EINVAL is only used when passing wrong arguments to functions.
Do not handle an invalid method in parse_request_line(). Defer this to
the caller so it can respond with "405 Method Not Allowed".
Improves parsing of the timeout parameter to the Session header,
allowing whitespace after the semicolon.
Avoids a compiler warning due to variables shadowing a function argument.
Original commit message from CVS:
Based on Patch by: Daniel Charles <dcharles at ti dot com>
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
(gst_rtp_amr_depay_process):
* gst/rtp/gstrtpamrdepay.h:
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_base_init),
(gst_rtp_amr_pay_class_init), (gst_rtp_amr_pay_init),
(gst_rtp_amr_pay_setcaps), (gst_rtp_amr_pay_handle_buffer):
* gst/rtp/gstrtpamrpay.h:
Add support for AMR-WB.
Small cleanups such as using BOILERPLATE.
Original commit message from CVS:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream):
Fix compile warning when debug is disabled as spotted bu Saur on IRC.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_base_init),
(gst_avi_demux_reset), (gst_avi_demux_parse_stream):
* gst/avi/gstavidemux.h:
Parse subtitle text streams instead of erroring out (#442034). Still
needs a parser for the subtitles to actually show up.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_push_event),
(gst_avi_demux_loop):
Make _push_event() return TRUE if the event could be pushed on at
least one pad and not only if it could be pushed on all pads,
otherwise we'll end up posting an error message on EOS if one or
more source pads are not connected.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_reset),
(gst_id3demux_send_new_segment), (gst_id3demux_chain),
(gst_id3demux_sink_event):
* gst/id3demux/gstid3demux.h:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_reset),
(gst_tag_demux_chain), (gst_tag_demux_sink_event),
(gst_tag_demux_send_new_segment):
Handle and adjust new-segment events so that downstream really
sees a stream with the tag pieces stripped off the front and back.
Fixes strangeness in seeking when mp3 decoders use the new-segment
byte position to estimate their current playback position timestamp
and then the arriving buffers don't match up.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_detect):
Don't unnecessarily perform a READY->NULL->READY transition on the
detected audio sink when starting up. Fixes: #440127
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_auth),
(gst_rtspsrc_try_send), (gst_rtspsrc_parse_methods),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_play):
(rtsp_connection_send), (rtsp_connection_receive):
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send):
Fix for new API.
* gst/rtsp/rtspconnection.c: (add_auth_header),
Only add authorisation and session headers when sending messages.
* gst/rtsp/rtspmessage.c: (key_value_foreach), (rtsp_message_init),
(rtsp_message_init_request), (rtsp_message_init_response),
(rtsp_message_unset), (rtsp_message_add_header),
(rtsp_message_remove_header), (rtsp_message_get_header),
(rtsp_message_append_headers), (dump_key_value),
(rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
Add support for multiple headers of the same type by storing the parsed
headers in a GArray instaed of a hashtable.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c (gst_spectrum_set_property,
gst_spectrum_event, gst_spectrum_transform_ip):
Use lock to protect from concurrent access.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_create), (gst_udpsrc_unlock), (gst_udpsrc_unlock_stop):
Since we depend on 0.10.13 -core, override the unlock_stop vmethod for
safer shutdown.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* configure.ac:
* gst/auparse/gstauparse.c: (gst_au_parse_reset),
(gst_au_parse_parse_header), (gst_au_parse_chain):
* gst/auparse/gstauparse.h:
Use audioconvert for converting from non-native endianness floats
in auparse instead of doing it ourself. Fixes#424527.
This needs the audioconvert from plugins-base CVS.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
(gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_init),
(gst_rtp_h263p_pay_set_property), (gst_rtp_h263p_pay_get_property),
(gst_rtp_h263p_pay_flush):
* gst/rtp/gstrtph263ppay.h:
Add new fragmentation mode base on GOB headers. Fixes#438940.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
Don't crash when an unsupported transport error was returned by the
server, just try to configure the next stream. Fixes#439255.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
Add TCP timeout property and use it for all TCP connection.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_write), (rtsp_connection_next_timeout),
(rtsp_connection_reset_timeout):
Make connect and writes cancelable and make them use the timeout.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams):
Refactor timeout handling.
Also send keep-alive when dealing with TCP transport.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_free), (rtsp_connection_next_timeout),
(rtsp_connection_reset_timeout):
* gst/rtsp/rtspconnection.h:
Use a timer to handle the session timeouts, add some methods to deal
with timeouts.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams):
Ignore streams that fail the setup command, we will retry with a
different transport later on.
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_configure_stream):
Fix encoding name case.
Original commit message from CVS:
* gst/rtp/gstrtptheoradepay.c: (decode_base64),
(gst_rtp_theora_depay_parse_configuration):
* gst/rtp/gstrtptheorapay.c: (encode_base64),
(gst_rtp_theora_pay_finish_headers),
(gst_rtp_theora_pay_handle_buffer):
Update theora pay/depayloader in a similar to vorbis.
* gst/rtp/gstrtpvorbisdepay.c:
(gst_rtp_vorbis_depay_parse_configuration):
Update docs.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send):
When we try to execute a method that is not supported by the server,
don't error out but remove the method from the accepted methods so that
we never try to perform this method again.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_range):
Parse range correctly.
* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
The baseurl now always has a '/' at the start.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps),
(gst_rtspsrc_parse_range), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
Factor out caps configuration and configure more stuff such as the time
ranges and speed/scale values.
* gst/rtsp/rtsptransport.c:
Add Copyright after non-trival fixes.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_transform_ip):
Use guint8 * instead of gpointer then vs6 can build
in_data += (filter->width / 8).
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
* gst/rtsp/rtspmessage.c: (rtsp_message_init_data),
(rtsp_message_get_header):
* gst/rtsp/rtspmessage.h:
Make channel guint8 where possible.
Make rtsp_message_init_data() take the channel as a guint8.
* gst/rtsp/rtspdefs.c:
Fixed a typo: Timout -> Timeout
* gst/rtsp/rtspdefs.h:
Make RTSP_CHECK() behave as a statement.
* gst/rtsp/sdpmessage.c:
Avoid a compiler warning in INIT_ARRAY().
Fixes#437692.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free),
(rtsp_url_get_request_uri):
* gst/rtsp/rtspurl.h:
Add support for query parameters to RTSP URLs.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes#437670.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
Skip LIST chunks before the fmt chunk (fixes#437499). Also fix
streaming mode regression for file from #343837 with 'bext' chunk
before the 'fmt' chunk.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
(gst_rtspsrc_handle_src_event),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.h:
Preliminary seek support.
Activate internal pads so that we can receive events on them.
Don't try to parse a range string when it's NULL.
Original commit message from CVS:
* gst/rtp/README:
Update README with new RTP variables that will be used for
synchronisation.
* gst/rtp/gstrtpvorbisdepay.c: (decode_base64),
(gst_rtp_vorbis_depay_parse_configuration),
(gst_rtp_vorbis_depay_process):
* gst/rtp/gstrtpvorbispay.c: (encode_base64),
(gst_rtp_vorbis_pay_finish_headers),
(gst_rtp_vorbis_pay_handle_buffer):
Update vorbis pay and depayloader to draft-04.
