Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_create_index),
(gst_flv_demux_loop):
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_script),
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
(gst_flv_parse_tag_timestamp):
* gst/flv/gstflvparse.h:
In pull mode we create our own index before doing anything else
and don't use the index provided by some files (which are more than
often incorrect and cause failed seeks).
For push mode we still use the index provided by the file and extend it
while doing the playback.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_push_src_event),
(gst_flv_demux_loop), (gst_flv_demux_handle_seek_pull),
(gst_flv_demux_sink_event):
Instead of using gst_pad_event_default() use a small
gst_pad_push_event() wrapper that only does what we want and is much
more simple.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_change_state),
(gst_flv_demux_set_index), (gst_flv_demux_init):
* gst/flv/gstflvdemux.h:
If our index was created by the element and not provided from the
outside we should destroy it when starting a new stream to get
all old entries removed.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_pull_range):
Improve debugging a bit when pulling a buffer from upstream fails.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
(gst_flv_demux_handle_seek_pull), (gst_flv_demux_dispose):
* gst/flv/gstflvdemux.h:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Close the currently playing segment from the streaming thread
instead of the thread where the seek event is handled.
Original commit message from CVS:
* gst/flv/gstflvmux.c: (gst_flv_mux_audio_pad_setcaps),
(gst_flv_mux_write_buffer):
Don't set video_codec to the value that actually should go
into audio codec, otherwise we create invalid files.
Fixes bug #556564.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Use gst_pad_alloc_buffer_and_set_caps() to make sure we get
a buffer with caps that we can work with (i.e. the pad's caps).
Add non-keyframe video frames to the index too but without the
keyframe flag.
Add audio frames to the index only if we have no video stream.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Create pads from the pad templates, use fixed caps on them
and only activate them after the caps are set.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_loop):
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_timestamp):
* gst/flv/gstflvparse.h:
Get an approximate duration of the file by looking at the timestamp
of the last tag in pull mode. If we get (maybe better) duration from
metadata later we'll use that instead.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_pull_range),
(gst_flv_demux_pull_tag), (gst_flv_demux_pull_header):
Refactor _pull_range() logic with checks into a seperate function
to make things a bit more readable.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_chain),
(gst_flv_demux_base_init):
Use gst_element_class_set_details_simple().
If we get GST_FLOW_NOT_LINKED in the parse loop but at least
one of the pads is linked continue the loop.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate),
(gst_flv_parse_tag_audio), (gst_flv_parse_video_negotiate):
Correct caps for video codec id 5: It's On2 VP6 with alpha channel
which needs a different decoder and has different caps.
Add support for audio codec id 14, which is MP3 with 8kHz sampling
rate.
Fix endianness and signedness for raw audio codec ids.
Add support for alaw and mulaw audio.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_chain):
Go out of the parse loop as soon as we get an error instead
of parsing until the GstAdapter is empty.
Add some explanations about the header and tag size.
Don't print synchronizing message if everything is fine.
Original commit message from CVS:
* gst/flv/Makefile.am:
* gst/flv/gstflvdemux.c: (plugin_init):
* gst/flv/gstflvmux.c: (gst_flv_mux_base_init),
(gst_flv_mux_class_init), (gst_flv_mux_init),
(gst_flv_mux_finalize), (gst_flv_mux_reset),
(gst_flv_mux_handle_src_event), (gst_flv_mux_handle_sink_event),
(gst_flv_mux_video_pad_setcaps), (gst_flv_mux_audio_pad_setcaps),
(gst_flv_mux_request_new_pad), (gst_flv_mux_release_pad),
(gst_flv_mux_write_header), (gst_flv_mux_write_buffer),
(gst_flv_mux_collected), (gst_flv_mux_change_state):
* gst/flv/gstflvmux.h:
Add first version of a FLV muxer. The only missing feature is writing
of stream metadata.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item),
(gst_flv_parse_tag_script):
Handle NULL returns from FLV_GET_STRING() more gracefully. Fixes
crash caused by a strlen on a NULL string (#527622).
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Don't strdup (and thus leak) codec name strings when passing
them to gst_tag_list_add().
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Fix list of supported and known codecs.
Emit tag with the codec name so it gets properly reported in totem and
other applications.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Output segment with proper 'stop' value, makes flvdemux 100% compatible
with gnonlin.
Original commit message from CVS:
* gst/flv/gstflvparse.c:
Add mapping for Nellymoser ASAO audio codec.
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Make sure we
actually have data to read at the end of the tag. This avoids trying
to allocate negative buffers.
Original commit message from CVS:
2007-10-22 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video), (gst_flv_parse_tag_type): Don't
emit no-more-pads for single pad scenarios as the header
is definitely not reliable. We emit them for 2 pads scenarios
though to speed up media discovery.
Original commit message from CVS:
2007-09-27 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video): I got it wrong again, audio rate
was not detected correctly in all cases.
Original commit message from CVS:
2007-09-26 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video): codec_data is needed for every tag
not just the first one. (Fix a stupid bug i introduced without
testing)
Original commit message from CVS:
2007-09-26 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video): Fix bit masks operations to be
sure we detect the codec_tags and sample rates correctly.
Fix raw audio caps generation.
Original commit message from CVS:
2007-08-27 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_handle_seek_pull):
Make sure we initialize the seek result.
Original commit message from CVS:
2007-08-22 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
(gst_flv_demux_pull_tag):
* gst/flv/gstflvdemux.h:
* gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item),
(gst_flv_parse_tag_script), (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video): Handle pixel aspect ratio through
metadata tags like ASF does. Fluendo muxer supports this and
Flash players can support it as well this way.
Original commit message from CVS:
2007-08-22 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_pull_tag):
* gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item),
(gst_flv_parse_tag_script), (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video): Make sure we don't try filling up the
index if no times object was parsed. Fix the way we decide to
push
tags and emit no-more-pads. Fix some printf typing in debugging.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_set_index),
(gst_flv_demux_get_index):
Fix locking and refcounting on the index.
Original commit message from CVS:
2007-08-14 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
(gst_flv_demux_adapter_flush), (gst_flv_demux_chain),
(gst_flv_demux_pull_tag), (gst_flv_demux_do_seek),
(gst_flv_demux_handle_seek), (gst_flv_demux_sink_event),
(gst_flv_demux_src_event), (gst_flv_demux_query),
(gst_flv_demux_change_state), (gst_flv_demux_set_index),
(gst_flv_demux_get_index), (gst_flv_demux_dispose),
(gst_flv_demux_class_init): First method for seeking in pull
mode using the index built step by step or coming from metadata.
* gst/flv/gstflvdemux.h:
* gst/flv/gstflvparse.c: (FLV_GET_STRING),
(gst_flv_parse_metadata_item), (gst_flv_parse_tag_script),
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Parse
more metadata types and keyframes index.
Original commit message from CVS:
2007-07-19 Julien MOUTTE <julien@moutte.net>
* configure.ac:
* gst/flv/Makefile.am:
* gst/flv/gstflvdemux.c: (gst_flv_demux_flush),
(gst_flv_demux_cleanup), (gst_flv_demux_chain),
(gst_flv_demux_pull_tag), (gst_flv_demux_pull_header),
(gst_flv_demux_seek_to_prev_keyframe), (gst_flv_demux_loop),
(gst_flv_demux_sink_activate),
(gst_flv_demux_sink_activate_push),
(gst_flv_demux_sink_activate_pull), (gst_flv_demux_sink_event),
(gst_flv_demux_change_state), (gst_flv_demux_dispose),
(gst_flv_demux_base_init), (gst_flv_demux_class_init),
(gst_flv_demux_init), (plugin_init):
* gst/flv/gstflvdemux.h:
* gst/flv/gstflvparse.c: (FLV_GET_BEUI24), (FLV_GET_STRING),
(gst_flv_demux_query_types), (gst_flv_demux_query),
(gst_flv_parse_metadata_item), (gst_flv_parse_tag_script),
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
(gst_flv_parse_tag_type), (gst_flv_parse_header):
* gst/flv/gstflvparse.h: Adds a first draft of an FLV demuxer.
It does not do seeking yet, it supports pull and push mode so
YES
you can use it to play youtube videos directly from an HTTP uri.
Not so much testing done yet but it parses metadata, reply to
duration queries, etc...
First of all a keyframe seek should be done to the
keyframe right before the requested position and not
to the keyframe that is nearest to the requested position.
Use per track index arrays and use our new binary search function
from core to speed up the search.
Rewrite the quant table parsing to also handle multiple tables in one JPEG HDQ
segment.
Handle more jpeg types by keeping track of the tables used per component and
putting the used ones in the quant headers.
Read the timestamp of the incomming buffer before we push it in the adapter and
flush it out again as the buffer might be unreffed then and we read from invalid
memory.
Fixes#581444.
Don't require width/height on the caps. Use the SOF header to find width/height
and fall back to the caps if there is no SOF. Also use the SOF info to find the
subsampling and quantization tables used. This allows us to set the right type
value in the JPEG rtp header.
Deprecate the quality property, it's unused now and it was used wrongly before.
Always send full quant tables for now until we have some code to detect default
ones.
Fixes#580880
Use the width and the height from the payload headers and set them on the
output caps for added awesomeness.
Fix quant parsing, we need to check the type in the lower 6 bits.
Add first bits of caching quantization tables.
Server eof (e.g. connection closed) is announced as connection closed,
so better record state and act accordingly to prevent (read/write)
errors during subsequent teardown/cleanup sequences. #Fixes 580851.(c).
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.
