We didn't handle unsynchronization at all up to now, which might have
caused frames to not be extracted - esp. frames after an APIC picture
frame. Fixes#577468.
If the codec is actually something else (e.g. mjpeg) change the caps to
match when parsing the ESDS atom.
Also, for AAC, override rate and channels with correct values read from
ESDS, since the rate/channels values elsewhere are often wrong.
We implemented the AAL2 packing, add the encoding-name for those to the caps and
a property to force AAL2 decoding (always TRUE for now).
Implement RFC3551 unpacking for regular G726.
See #567140.
In streaming mode, avidemux is not supposed to send an EOS event downstream but
it is supposed to return UNEXPECTED from the chain function instead so that
upstream can do the right EOS handling.
Fix the duration query so that it also works with formats other than
TIME, such as DEFAULT to get the number of frames.
Add a convert function.
Fixes#578052.
In the sequence of header lengths, for headers >127 bytes, we use
multiple bytes to encode the length. Bytes other than the last must have
the top (flag) bit set.
The on-npt-stop signals was added only recently to rtpjitterbuffer in
-bad, so check if the signal exists before g_signal_connect()ing to
it, to avoid warnings.
Some clips (trailers) may have (length-wise) unbalanced streams,
which stalls the pipeline if seeking into that region.
Additional stream synchronization can handle this, as well as
sparse (subtitle) streams (at some later time ?)
Try to post a decent error message when it looks like we're failing
because the Real RTSP extension plugin is missing. Also add i18n
bits for rtspsrc so our error messages get translated.
Cater for DELTA_UNIT flag on buffers, keep track of current
position, remove and warn about edit lists if any (as those
as are de facto discarded anyway), add some debug statements
and indent fixes.
The audioMuxVersion structure is packed in such a way that the codec
data does not start byte-aligned, which means there's an extra bit of
padding at the end. We don't want that bit in the codec data, since
some decoders seem get confused when they're fed with an extra codec
data byte (also it's just not right of course).
Add network interface selection when joining multicast groups.
Useful when using the udpsrc on multihomed hosts.
Fixes#575234.
API: GstUDPSrc::multicast-iface
Non-ok flow returns may happen for a variety of perfectly legitimate and expected reasons
(temporarily not linked, seeking, pipeline shutdown), so we really shouldn't spew ERROR
debug messages to stderr in those cases. Fixes#570781. (Seems like someone already took
care of some of these.)
Standard pull mode loop based SEEK handling fails in push mode,
so convert the SEEK event appropriately and dispatch to upstream.
Also cater for NEWSEGMENT event handling, and properly inform
downstream and application of SEEKABLE capabilities, depending
on scheduling mode and upstream.
Previously the sockaddr length used for recvfrom() was calculated as
sizeof (struct sockaddr). However, this is too little to hold an IPv6
address, so the full size of the gst_sockaddr union should be used
instead.
MS RTSP spec states that the UDP port pair used in subsequent SETUP
requests for various streams must be identical (since there will actually
be only 1 stream of muxed asf packets). Following traditional specs and
using different port pairs in the SETUPs for separate streams will result
in all but the first one failing and only one stream being streamed.
So, in appropriate circumstances, retry UDP SETUP using previously used
port pair. Fixes#552650.
When we are dealing with connected sockets shared between a udpsrc and a udpsink
we might receive ICMP connection refused error messages in udpsrc that will
cause it to go into a bursty loop because the poll returns right away without a
message to read.
Instead of looping, read the error message from the error queue in udpsrc.
Fixes#567857.
Reading integers from random memory addresses will result
in SIGBUS on some architectures if the memory address
is not correctly aligned. This can happen at two
places in avidemux so we should use GST_READ_UINT32_LE
and friends here. Fixes bug #572256.
stps atoms contain "partial sync" information, which means that it's
a sync point where pts != dts. This is needed to properly handle
MPEG2, H.264, Dirac, etc., in quicktime.
Not all Matroska files have a Tags element which contains
information about the title among other things. Most video
Matroska files only contain the Title element so we
should parse this too. Fixes bug #570435.
Move reallocating the history buffer out of _compute_frequencies() and call the
right function as needed. Add some logging and tweak the formatting of existing
logging. Simplify setting need_new_coefficients when changing properties.
Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered. Fix warnings that gtk-doc points out.
Introduce a new max-delay property that can only
be set before going to PLAYING or PAUSED. This
is used to limit the maximum delay and is set
to the current delay by default.
Using this will make sure that we have enough data
in our internal ringbuffer for the echo. With dynamic
reallocation of the ringbuffer as used before silence
could've been used as the echo directly after setting
a new delay.
Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered.
Save some allocations if the echo delay is increased often
during playback by always allocating enough memory to hold
data up to the next complete second, i.e. in the worst case
allocate memory for one additional second.
Add a note to the docs that audioecho's reverb will
sound metallic. This happens because for a real
reverb filter additional filtering is necessary.
Also note which values should be used for the delay
property to get an echo effect.
The element can add an echo and a simple reverb effect to
an audio stream but for a real reverb filter it would need
some additional filtering to prevent a metallic-sounding
result.
Original commit message from CVS:
Patch by: Luotao Fu <l dot fu at pengutronix dot de>
* gst/videocrop/gstvideocrop.c:
(gst_video_crop_get_image_details_from_caps):
Add 8bit grayscale support to videocrop plugin. Fixes#567952.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Implement a simple compensation algorithm for rounding errors.
This makes sure that a spectrum message is posted on the bus
every interval nanoseconds. Fixes bug #567955.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_segments):
Catch invalid and commonly wrong playback rates in the elst atoms.
Fixes#567800.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state):
Don't call gst_fft_f32_free() with NULL to prevent a
crash. Fixes bug #567642.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Use correct types for frame/fft counters and some minor
cleanup.
Original commit message from CVS:
* gst/spectrum/Makefile.am:
* gst/spectrum/README:
* gst/spectrum/gstspectrum.c: (gst_spectrum_base_init),
(gst_spectrum_class_init), (gst_spectrum_init),
(gst_spectrum_reset_state), (gst_spectrum_finalize),
(gst_spectrum_set_property), (gst_spectrum_start),
(gst_spectrum_stop), (gst_spectrum_setup),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Post a spectrum message on the bus for every interval, even
if the interval is small than the length of the FFT.
Fixes bug #567642.
Major cleanup of the spectrum element.
Original commit message from CVS:
Patch by: Thiago Sousa Santos <thiagoss@embedded.ufcg.edu.br>
* gst/qtdemux/qtdemux.c:
Fix format string for guint64.
Original commit message from CVS:
* gst/audiofx/audiochebband.c: (gst_audio_cheb_band_class_init),
(gst_audio_cheb_band_init), (gst_audio_cheb_band_finalize),
(gst_audio_cheb_band_set_property):
* gst/audiofx/audiochebband.h:
* gst/audiofx/audiocheblimit.c: (gst_audio_cheb_limit_class_init),
(gst_audio_cheb_limit_init), (gst_audio_cheb_limit_finalize),
(gst_audio_cheb_limit_set_property):
* gst/audiofx/audiocheblimit.h:
* gst/audiofx/audiowsincband.c: (gst_audio_wsincband_class_init),
(gst_audio_wsincband_init), (gst_audio_wsincband_finalize),
(gst_audio_wsincband_set_property):
* gst/audiofx/audiowsincband.h:
* gst/audiofx/audiowsinclimit.c: (gst_audio_wsinclimit_class_init),
(gst_audio_wsinclimit_init), (gst_audio_wsinclimit_finalize),
(gst_audio_wsinclimit_set_property):
* gst/audiofx/audiowsinclimit.h:
Use a custom mutex for protecting the instance fields instead of
the GstObject lock. Using the latter can lead to deadlocks, especially
with the FIR filters when updating the latency.
Original commit message from CVS:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofxbasefirfilter.c:
(gst_audio_fx_base_fir_filter_dispose),
(gst_audio_fx_base_fir_filter_base_init),
(gst_audio_fx_base_fir_filter_class_init),
(gst_audio_fx_base_fir_filter_init),
(gst_audio_fx_base_fir_filter_push_residue),
(gst_audio_fx_base_fir_filter_setup),
(gst_audio_fx_base_fir_filter_transform),
(gst_audio_fx_base_fir_filter_start),
(gst_audio_fx_base_fir_filter_stop),
(gst_audio_fx_base_fir_filter_query),
(gst_audio_fx_base_fir_filter_query_type),
(gst_audio_fx_base_fir_filter_event),
(gst_audio_fx_base_fir_filter_set_kernel):
* gst/audiofx/audiofxbasefirfilter.h:
* gst/audiofx/audiofxbaseiirfilter.c:
Implement a base class for generic audio FIR filters.
* gst/audiofx/audiowsincband.c:
(gst_gst_audio_wsincband_mode_get_type),
(gst_gst_audio_wsincband_window_get_type),
(gst_audio_wsincband_base_init), (gst_audio_wsincband_class_init),
(gst_audio_wsincband_init), (gst_audio_wsincband_build_kernel),
(gst_audio_wsincband_setup), (gst_audio_wsincband_set_property),
(gst_audio_wsincband_get_property):
* gst/audiofx/audiowsincband.h:
* gst/audiofx/audiowsinclimit.c:
(gst_audio_wsinclimit_mode_get_type),
(gst_audio_wsinclimit_window_get_type),
(gst_audio_wsinclimit_base_init),
(gst_audio_wsinclimit_class_init), (gst_audio_wsinclimit_init),
(gst_audio_wsinclimit_build_kernel), (gst_audio_wsinclimit_setup),
(gst_audio_wsinclimit_set_property),
(gst_audio_wsinclimit_get_property):
* gst/audiofx/audiowsinclimit.h:
* tests/check/elements/audiowsincband.c: (GST_START_TEST):
* tests/check/elements/audiowsinclimit.c: (GST_START_TEST):
Use this new base class for audiowsincband and audiowsinclimit.
Also cleanup both elements.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
In push mode, error out if we get EOS before we've created any srcpads.
Handle (in pull mode) some files that have a truncated moov atom where
the final sub-atom is a 'free' atom and the contents of that are not
present in the file.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps):
Some cleanups, refactoring and minor enhancements in caps handling.
* gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init),
(gst_matroska_mux_init), (gst_matroska_pad_reset),
(gst_matroska_pad_free), (gst_matroska_mux_reset),
(gst_matroska_mux_video_pad_setcaps),
(gst_matroska_mux_request_new_pad):
* tests/check/elements/matroskamux.c: (teardown_src_pad):
Only remove, release or reset what is appropriate upon state change.
Original commit message from CVS:
* gst/matroska/matroska-mux.c: (gst_matroska_mux_reset),
(gst_matroska_mux_handle_sink_event), (gst_matroska_mux_finish):
* gst/matroska/matroska-mux.h:
Remove internal taglist and fully use tagsetter interface.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_reset),
(gst_avi_mux_riff_get_avi_header):
* gst/avi/gstavimux.h:
Ensure header size invariance during subsequent rewrite by using
tags snapshot.
Original commit message from CVS:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofxbaseiirfilter.c:
(gst_audio_fx_base_iir_filter_base_init),
(gst_audio_fx_base_iir_filter_dispose),
(gst_audio_fx_base_iir_filter_class_init),
(gst_audio_fx_base_iir_filter_init),
(gst_audio_fx_base_iir_filter_calculate_gain),
(gst_audio_fx_base_iir_filter_set_coefficients),
(gst_audio_fx_base_iir_filter_setup), (process),
(gst_audio_fx_base_iir_filter_transform_ip),
(gst_audio_fx_base_iir_filter_stop):
* gst/audiofx/audiofxbaseiirfilter.h:
Implement a base class for IIR filters.
* gst/audiofx/audiochebband.c: (gst_audio_cheb_band_base_init),
(gst_audio_cheb_band_class_init), (gst_audio_cheb_band_init),
(generate_coefficients), (gst_audio_cheb_band_set_property),
(gst_audio_cheb_band_setup):
* gst/audiofx/audiochebband.h:
* gst/audiofx/audiocheblimit.c: (gst_audio_cheb_limit_base_init),
(gst_audio_cheb_limit_class_init), (gst_audio_cheb_limit_init),
(generate_coefficients), (gst_audio_cheb_limit_set_property),
(gst_audio_cheb_limit_setup):
* gst/audiofx/audiocheblimit.h:
Use the IIR filter base class for the chebyshev filters.
Original commit message from CVS:
Patch by: j^ <j at oil21.org>
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps),
(qtdemux_audio_caps):
Add codec mapping for xvid, fmp4 and ac3 tracks.
Fixes#565850
Original commit message from CVS:
* ext/pulse/pulsemixerctrl.c:
And remove temporary comment pointing to the bug ticket.
* gst/avi/gstavimux.c:
Move reoccuring logging to LOG and log instance too.
Original commit message from CVS:
* gst/avi/gstavimux.c:
Instead of filtering wrongly just use the mergemode. Applications is
use KEEP_ALL if they want to supress tag-events. Fixes#563221 for
avi for real (I hope). Everyone chime in, before I fix the others.
Original commit message from CVS:
* gst/avi/gstavidemux.c:
More logging.
* gst/avi/gstavimux.c:
Handle more metadata fields. Better estimate of metadata size. Don't
merge received tags, if application has specified tags using
GST_TAG_MERGE_REPLACE_ALL. Fixes#563221 for avi.
Original commit message from CVS:
* gst/rtp/gstrtpjpegdepay.c: (gst_rtp_jpeg_depay_process):
Add an EOI marker at the end of the jpeg frame when it's missing.
Fixes#563056.
Original commit message from CVS:
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush),
(gst_rtp_mp4v_pay_event):
Don't try to push packets before we could find a valid config
startcode. Fixes#563509.
