pulseaudiosink breaks visualisations in its current form, so let's
prevent it from being autoplugged for the time being.
The best we can hope to do in the 0.10 series is query the list of
available sinks and their formats, and expose these as the bin's sinkpad
caps. While this is not a comprehensive solution, it will make sure that
we're only trying to support compressed formats if we're certain that
one exists.
The long-term fix for this will be in the form of proper upstream
renegotiation support in the 0.11/1.0 series.
https://bugzilla.gnome.org/show_bug.cgi?id=666361
pa_stream_* functions return negative on error, despite the defines
for error codes being positive.
I only got to repro the error twice, so I'm not sure 100% sure this
fixes the issue (the negative var being uninitialized after returning
from pa_stream_get_latency).
Clients can configure the desired behaviour via "transport" property. The
default behaviour is ignoring the transport state. Other modes are master and
slave.
This is causing us lots of headaches in 0.10 and needs to be done
differently and properly in 0.11. playbin or decodebin should
reconfigure themselves based on reconfigure events, for example.
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.
Fix crash or hang in generic/states unit test when doing stop()
right after start(). Create main loop in the start function already
and not just in the thread function, so that stop() always has a
valid main loop to quit on. Also, calling g_main_loop_quit() before
g_main_loop_run() won't work and result in the stop function waiting
for the thread to join forever. Therefore, wait for the thread to
be ready and get the main loop running in the start() function, to
be sure stop() always works.
Conflicts:
ext/pulse/pulseaudiosink.c
ext/pulse/pulsesrc.c
gst/audioparsers/gstaacparse.c
gst/audioparsers/gstamrparse.c
gst/audioparsers/gstdcaparse.c
gst/audioparsers/gstflacparse.c
gst/effectv/gstradioac.c
gst/effectv/gstradioac.h
gst/effectv/gstripple.c
Some possible FIXMEs remaining in the audio parser getcaps functions.
PulseAudio 1.0 supports per-source-output volumes, and this exposes the
functionality via the GstStreamVolume interface.
When compiled against pre-1.0 PulseAudio, the interface is not
implemented, and the "volume" or "mute" properties are not available.
This bit of ugliness will go away when we can depend on PulseAudio 1.0
or greater.
https://bugzilla.gnome.org/show_bug.cgi?id=595055
We need to tell the base class that we're dropping buffers,
so it drops the input timestamps corresponding to these.
Otherwise, the first actual audio buffers we output will be
stamped with those - GST_CLOCK_TIMESTAMP_NONE. That mismatch
between input buffer count and output buffer count will stay
while playing. With enough headers and long enough buffer
durations, the sink will have played enough before receiving
the first valid timestamp (usually 0), and will trigger an
audible discontinuity.
Apparently libpng can technically do up to 2^31-1 rows and columns. However it
imposes an (arbitrary) default limit of 1 million (that could theoretically be
lifted by using some additional API).
Moved array allocation to the heap now.
A setcaps function needs to actually verify the caps carefully. In this case,
it was possible to e.g. link a video decoder with YUV+RGB template caps to
pngenc. That would cause a crash when the decoder pushes a YUV buffer. Same
thing when pushing a valid buffer that exceeds the resolution limits.
Also, missing framerate caps field would cause a glib critical warning due to
invalid GValue. This fails hard now.
This introduces a new bin which wraps around pulsesink and depending on
the formats supported by the sink, plugs in/out a decodebin2 as
required. This allows users to switch sinks on the stream and adapts
accordingly (for example, you could watch a movie in passthrough mode on
your receiver which supports AC3 decode, then plug out and switch to a
non-digital profile to continue uninterrupted on analog output).
The bin is required because doing the same with playbin2/playsink will
require API changes that cannot be made in 0.10. With 0.11/1.0, we
should be able to ask for upstream caps renegotiation to deal with all
this.
https://bugzilla.gnome.org/show_bug.cgi?id=657179
flacdec converts the src timestamp to a sample number, uses that internally, then reconverts the sample number to a timestamp for the output buffer. Unfortunately, sample numbers can't be represented in an integer number of nanoseconds, and the conversion process was truncating rather than rounding, resulting in sample numbers and output timestamps that were often off by a full sample.
This corrects the time->sample convesion
There's no use in splitting the incoming data down to the segsize
limit - by writing as much as possible in one chunk, we increase
performance and avoid PulseAudio unnecessary rewinds.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
The libFLAC API is callback based, and we must only call it to
output data when we know we have enough input data. For this
reason, a single processing step is done when receiving a buffer.
However, if there were metadata buffers still pending, a step
intended for the first audio frame might end up writing that
leftover metadata. Since a single step is done per buffer, this
will cause every buffer to be written one step late.
This would add some latency (a bufferfull's worth), possibly
lose a buffer when seeking or the like, and also cause timestamp
and offset to be applied to the wrong buffer, as updates to
the "current" segment last_stop (from incoming buffer timestamp)
will be applied to an output buffer originating from the previous
incoming buffer.
This fixes the issue by ensuring that, upon receiving the first
audio frame, processing is done till all metadata is processed,
so the next "single step" done will be for the audio frame. After
this, we should keep to 1 input buffer -> 1 output buffer and so
avoid getting out of sync.
https://bugzilla.gnome.org/show_bug.cgi?id=650960
Sine the base class now does the negotiation from the streaming thread we have
to be careful and check if the stream is ready before changing its corked state.
This adds support for various compressed formats (AC3, E-AC3, DTS and
MP3) payloaded in IEC 61937 format (used for transmission over S/PDIF,
HDMI and Bluetooth).
The acceptcaps() function allows bins to probe for what formats the sink
being connected to support. This only works after the element is set to
at least READY.
If the underlying sink changes and the format we are streaming is not
available, we emit a message that will allow upstream elements/bins to
block and renegotiate a new format.
This exposes the source output index of the record stream that we open
so that clients can use this with the introspection if they want (to
move the stream, for example).
Since commit 8bfd80, gst_pulseringbuffer_stop doesn't wait for the
deferred call to be run before returning. This causes a race when
READY->NULL is executed shortly after, which stops the mainloop. This
leaks the element reference which is passed as userdata for the callback
(introduced in commit 7cf996, bug #614765).
The correct fix is to wait in READY->NULL for all outstanding calls to
be fired (since libpulse doesn't provide a DestroyNotify for the
userdata). We get rid of the reference passing from 7cf996 altogether,
since finalization from the callback would anyways lead to a deadlock.
Re-fixes bug #614765.
So that pulsesrc/pulsesink get chosen over other possible PRIMARY
src/sinks by autoaudiosink. Presumably, if pulse is available, it
is always preferred over another src/sink.
Fixes: #647540.
For some reason, in code dating to 2001, encoded jpeg buffers were
rounded up to multiples of 4 bytes. With the added bonus that the
extra bytes are unwritten, causing valgrind issues. Oops. I can't
think of any reason why JPEG buffers need to be multiples of 4 bytes,
so I removed the padding. There might be some code somewhere that
depends on this behavior, so if this needs to be reverted, please fix
the valgrind issues.
This drops support fof PulseAudio versions prior to 0.9.16, which was
released about 1.5 years ago. Testing with very old versions is not
feasible and we don't want to maintain 2 independent code-paths.
Not doing so will cause buffers to be received by downstream without
a time base set.
We use the same method avimux uses to get access to the event when
collectpads got the sink event function.
https://bugzilla.gnome.org/show_bug.cgi?id=640323
Don't use g_assert() for error handling, even if they're highly unlikely.
Either we *know* that something can't happen, in which case we
should just not handle it, or we think something can happen, but it is
very very unlikely that it will ever happen, in which case we should
handle it like any other error instead of asserting.
g_assert() is best left for conditions we have control of, like checking
internal consistency of our code, not checking return values of external
code.
Fixes a bunch of warnings when compiling with -DG_DISABLE_ASSERT:
gstrtpgsmpay.c: In function 'gst_rtp_gsm_pay_handle_buffer':
gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used
gstspeexenc.c: In function 'gst_speex_enc_encode':
gstspeexenc.c:904:19: warning: variable 'written' set but not used
pulsesink.c: In function 'gst_pulsesink_change_state':
pulsesink.c:2725:9: warning: variable 'res' set but not used
pulsesrc.c: In function 'gst_pulsesrc_change_state':
pulsesrc.c:1253:7: warning: variable 'e' set but not used
GCC 4.6.x spits warnings about such usage of variables. The variables in
raw1394 were marked with G_GNUC_UNUSED as this seemed omre appropriate.
The others were removed.
Instead only store them inside the flac metadata. There's
no point in storing them twice and the flac metadata is
still the official way to store image tags inside flac.
Speex has build in silence detection. If speex_encode_int returns 0,
than there is silence and sample do not need to be transmitted.
This work only if vbr=1 and dtx=1 optionas are enabled.
So if we get 0, we add GAP flag to the sample.
Pulsesink was recently changed to defer uncorking until there is data
to write. This condition will however never occur when EOS in being
rendered (since that marks the end of data). Changing to PAUSED state
while EOS is being waited on results in a hang: pausing corks the
stream, which will never be undone since there is no more data when
going back to PLAYING. If pulsesink is the clock provider, deadlock
ensues since time doesn't continue in corked state and the clock id
for EOS wait never fires.
Fixes#645961.
If we did not know how many frames to expect, then we get an unexpected
end of stream when trying to decode more frames that are there, if there
are leftover bits to pad to the next byte
Looking at the remaining bits in the bitstream after decoding a
single frame can't be used as loop condition. The remaining
bits might not give a complete frame and the speex decoder will
then output nothing but access uninitialized memory, which leads
to valgrind warnings.
Fixes bug #644669.
Allows applications to connect to the "draw" signal of
the element and do their custom drawing there.
Includes an example application demonstrating usage.
Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=595520
Not doing so can result in a deadlock when two threads enter
gst_pulseringbuffer_open_device at the same time, as
pa_threaded_mainloop_wait releases the mainloop lock while waiting,
allowing another thread to take it, resulting in a deadlock as two
threads waits for the lock the other is holding.
https://bugzilla.gnome.org/show_bug.cgi?id=643087
The element downstream of mp3enc might only accept certain sample rates or channels,
make sure we relay any restrictions that do exist to upstream when it does a
get_caps() on the sink pad. That way upstream elements like audioresample or
audioconvert can pick a sample rate / channel configuration that will be accepted,
instead of just negotiating to the highest, which might then be rejected.
https://bugzilla.gnome.org/show_bug.cgi?id=641151
By allowing larger chunks to be sent, PulseAudio will have a
lower CPU usage. This is especially important on low-end machines,
where PulseAudio can crash if packets are coming in at a higher
rate than PulseAudio can process them.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
After starting the ringbuffer, we wait for enough data to arrive before
uncorking the stream. This will cause the pipeline to stall if we get an
EOS (or otherwise need to flush the stream) before sufficient data
becomes available. This patch makes sure that the stream is uncorked
while flushing to avoid this problem.
Fixes issue with a webkit unit test testing reverse playback of
an MP4 H.264/AAC file.
https://bugzilla.gnome.org/show_bug.cgi?id=639740
This makes the call to pa_stream_cork() during ringbuffer pause()
synchronous, which makes sure that the clock does not advance after we
take a snapshot for start_time.
https://bugzilla.gnome.org/show_bug.cgi?id=639240
Error out when we are asked to read a different size that what was configured as
the jack period size because that would mean something else is wrong.
Fixes#618409
Not only adjust buffer-time and avoid segtotal=0, but instead ensure segtotal is
atleast 2. Do same change on jacksrc. We could also check the latency and buffer
time configured by the client and adjust buffer-time so that we get to the same
number of segments.
Jack overrides user-specified latency-time with the one it gets from jack
itself. It also needs to adjust buffer-time somewhat to avoid segtotal being 0
The gst_jack_audio_client_set_active() flags the port as deactivating and uses
a GCond to wait until the jack_process_cb() has run once more and cleared the
flag. This way the client zero's the buffer. This happens if one manyally go
to PAUSED and then to READY, while leting the mainloop run inbetween.
Add a new connection mode to jacksrc and jacksink. In this new auto-force
connection mode jack will create as many ports as requested/needed in the
pipeline and will then connect as many physical ports as possible, possibly
leaving some ports unconnected.
Also get rid of some leftover g_print.
Fixes#575284.
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c:
* ext/jack/gstjackaudiosrc.c:
Query port latencies for sink/src delays.
* ext/jack/gstjackbin.c:
No printf please.
Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c:
(gst_jack_audio_sink_allocate_channels):
Include the element name in the port name to avoid duplicate port names.
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (gst_jack_audio_sink_class_init):
Work around missing bits of thread-safety on older GLibs some
more to avoid assertions when starting up multiple playbin
objects concurrently (see #512382).
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_acquire):
Add stdlib include here too.
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_acquire):
Try t better name clients. properly handle return codes when re-
establishing links.
Original commit message from CVS:
Based on patch by: Paul Davis <paul at linuxaudiosystems dot com>
* ext/jack/gstjackaudioclient.c: (gst_jack_audio_unref_connection):
Don't need to take the connection lock, it will not be used and could
cause deadlocks.
Original commit message from CVS:
Includes patch by: Paul Davis <paul at linuxaudiosystems dot com>
* ext/jack/Makefile.am:
* ext/jack/gstjackaudioclient.c: (gst_jack_audio_client_init),
(jack_process_cb), (jack_sample_rate_cb), (jack_buffer_size_cb),
(jack_shutdown_cb), (connection_find),
(gst_jack_audio_make_connection), (gst_jack_audio_get_connection),
(gst_jack_audio_unref_connection),
(gst_jack_audio_connection_add_client),
(gst_jack_audio_connection_remove_client),
(gst_jack_audio_client_new), (gst_jack_audio_client_free),
(gst_jack_audio_client_get_client),
(gst_jack_audio_client_set_active):
* ext/jack/gstjackaudioclient.h:
Make an object to manage client connections to the jack server which we
will use in the future to run selected jack elements with the same jack
connection.
Make some stuff a bit more threadsafe.
Activate the jack client ASAP.
* ext/jack/gstjackaudiosink.c:
(gst_jack_audio_sink_allocate_channels),
(gst_jack_audio_sink_free_channels), (jack_process_cb),
(gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_close_device),
(gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release),
(gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init),
(gst_jack_audio_sink_getcaps):
* ext/jack/gstjackaudiosink.h:
Use new client object to manage connections.
Don't remove and recreate all ports, try to reuse them.
Original commit message from CVS:
second batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
(in gst-plugins/ext/ this time)
Original commit message from CVS:
Remove all usage of gst_pad_get_caps(), and replace it with
gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap().
Original commit message from CVS:
Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files
Original commit message from CVS:
merge TYPEFIND branch. Major changes:
- totally reworked type(find) system
- all typefind functions are in gst/typefind now
- more typefind functions then before
- some plugins might fail to compile now because I don't have them installed and they
a) require bytestream or
b) haven't had their typefind fixed.
Please fix those plugins and put the typefind functions into gst/typefind if they don't have dependencies
Original commit message from CVS:
New typefind system:
* bytestream is now part of the core
* all plugins have been modified to use this new typefind system
* asf typefinding added
* mpeg video stream typefiding removed because it's broken
* duplicate typefind entries removed
* extra id3 typefinding added, because we've seen 4 types of files
(riff/wav, flac, vorbis, mp3) with id3 headers and each of these needs
to work. Instead, I've added an id3 element and let it redo typefiding
after the id3 header. this needs a hack because spider only typefinds
once. We can remove this hack once spider supports multiple typefinds.
* with all this, mp3 typefinding is semi-rewritten
* id3 typefinding in flac/vorbis is removed, it's no longer needed
* fixed spider and gst-typefind to use this, too.
* Other general cleanups
Original commit message from CVS:
New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs
Original commit message from CVS:
compatibility fix for new GST_DEBUG stuff.
Includes fixes for missing includes for config.h and unistd.h
I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately.
Original commit message from CVS:
another batch of connect->link fixes
please let me know about issues
and please refrain of making them yourself, so that I don't spend double
the time resolving conflicts
Original commit message from CVS:
some jack fixes, alsa touchups, and add rtp by default to the build
if there are any problems building rtp, we're moving it back to experimental ;)
Original commit message from CVS:
* a hack to work around intltool's brokenness
* a current check for mpeg2dec
* details->klass reorganizations
* an element browser that uses details->klass
* separated cdxa parse out from the avi directory
Original commit message from CVS:
Finally we're on to a proper jack setup, with a specialized bin and elements
that can only go in a jack bin. I had to fix the parser first to do this, but
to run it, the syntax is like so:
gst-launch jackbin.( filesrc ! mad ! jacksink )
But of course it's not fully functional yet. Sigh.
Original commit message from CVS:
* add notify back to filesrc, it's needed for MVC applications
* remove notify printouts from gst-launch
* cleanup in gst-plugins configure.ac
* some jack updates
* remove SELF_ITERATING flag in favor of SEF_SCHEDULABLE (not a clear name,
but it's what we have for the moment)
* improve parsing of request pad names, no more sscanf
* fixes to the fastscheduler Makefile.am
Original commit message from CVS:
* removal of //-style comments
* don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct,
and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory.
Original commit message from CVS:
s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/
@-substitued variables variables are defined as make variables automagically,
and this gives the user the freedom to say make GST_PLUGIN_LDFLAGS=-myflag
Original commit message from CVS:
* s/gst_element_install_std_props/gst_element_class_install_std_props/ -- it just makes more sense that way
* added jack element, doesn't quite work right yet but i didn't want to lose the work -- it does build, register,
and attempt to run though
* imposed some restrictions on the naming of request pads to better allow for reverse parsing
* added '%s' to reverse parsing
* added new bin flag to indicate that it is self-iterating, and some lame code in gst-launch to test it out
* fixen on launch-gui
* added pkg-config stuff for the editor's libs
* ext/pulse/pulsesrc.c (gst_pulsesrc_class_init, gst_pulsesrc_init)
(gst_pulsesrc_set_property, gst_pulsesrc_get_property)
(gst_pulsesrc_open): Add a "client" property, as in pulsesink.
Fixes#634914
Instead of using get_allowed_caps on the srcpad, the sinkpad getcaps
should use the getcaps of the srcpad's peer. This way the srcpad
can keep using fixed_caps and sinkpad getcaps exposes all caps
that can be negotiated
https://bugzilla.gnome.org/show_bug.cgi?id=637686
Add property to ignore decoding errors. Default is to ignore a few
decoding errors if the input is packetized, but error out immediately
if the input is not packetized.
Ignoring errors for packetized input most likely doesn't work
properly yet, so don't do that for now.
https://bugzilla.gnome.org/show_bug.cgi?id=623063
Don't uncork in the _start method just yet but wait until we have written some
samples to pulseaudio. This avoid underruns on pulseaudio and less crackling
noises when starting.
Using the symbols for the different Speex modes results
in crashes when using MSVC. Use the library functions to
get the modes instead.
Fixes bug #630378.
Make the is_dead check more clear and add an option to check for the status of
the stream in addition to the context.
We don't need a stream to get the device_description string.
Fixes#630317
We also need to share the main-loop threads as this owns the context. Thus have
a class wide main-loop thread. From this we create a context per client-name.
Instead of always looking up the context, we keep this with the instance. The
reverse mapping is only needed in pulse singal handlers. This saves a lot of
locking. Also one signal handler becomes simpler as ther eis only one mainloop
to notify.
Now valgind happy - no leaks, no bad reads/writes.
This reverts major parts of commit 69a397c32f.
Fixes#628996
Error messages should be translated. URIs and filenames should not
be part of the error message string that's shown to the user.
soup_message->reason_phrase is not translated and not suitable as
error message for users (see libsoup documentation). Also fix up
error codes a bit, as far as possible with the existing codes.
Use g_slist_prepend as we don't care about the order. Check for list == NULL
instead of iterating the list to see if it is empty. Move ctx allocation down
to prevent leak in case of failure.
Don't leak the pulsesink element by having the clock keep a ref to the sink.
Create the clock only once in the constructor and use the baseaudiosink clock
cleanup code.
And as a result, don't ignore WRONG_STATE and NOT_LINKED.
Both mean that it's a good idea to pass them upstream instead
of pretending that everything is good.
Allows the application to modify the client name used to connect when
connecting to the PulseAudio daemon. Note however that updating the
property after the element reached the READY state will have no
effect until the next NULL->READY transition.
Fixes bug #627174.
Before they contained the URL before the actual failure. The other
way around makes more sense and we do the same in other elements
like filesrc.
Fixes bug #627289.
Avoid to create a new PA context for each new client by using a hash
table containing the list of ring-buffers and the shared PA context
for each client. Doing this will improve application memory usage in
the cases where multiple pipelines involving multiple pulsesink
elements are used.
Fixes bug #624338.
If the application requests a state-change and pulsesink fails to open
the ring_buffer device the mainloop attribute of the sink should be
cleaned up to avoid future state-change (NULL->READY) failures.
The existing get_type() implementation is racy, and the
g_type_class_ref() workaround didn't actually work because
it was in the wrong function. Since class creation in GObject
is thread-safe these days (since 2.16), the class_ref workaround
is no longer needed and it is sufficient to ensure the _get_type()
function is thread-safe, which G_TYPE_DEFINE does.
https://bugzilla.gnome.org/show_bug.cgi?id=624338
The cycleCount register is 13 bits long and the cycleOffset one
is 12 bits long. To read the cycleCount register we need to shift
12 bits and not 13. Fixes#615461
When reseting, keep 'row' variables at a sane state after
freeing to avoid it being freed again on _resync realloc
when the element is reused.
Fixes#619943
Specifically, verify input components / colour space is as code
subsequently expects, thereby avoiding crashes or otherwise bogus output.
Presently, that means 3 components YCbCr colour space, and somewhat
limited sampling factors.
Fixes#600553.
Unlike filesrc, flacenc outputs the flac blocks neatly aligned one in
each buffer. This means that when we switch from metadata mode to
audio data passthrough mode, there's no data left in the adapter to
push out at this point, so check if there's data in the adapter
before requesting buffers from it (also needed in case we get input
buffers of 0 size).
Fixes#615793.
In fact, due to signedness issues, a negative delay would be changed to
an almost infinite wait causing shout2send to "lock up".
Reported by Christopher Montgomery.
When creating the caps allowed to upstream using downstream
restrictions, use gst_pad_get_allowed_caps as that has the
usable formats and puts into it the width, height and framerate
fields. This avoids getting errors about getcaps returning
non subset caps of its pad template.
This error showed up on the metadata plugin unit test in -bad.
Don't send another newsegment event if the upstream muxer/parser has already
sent one (otherwise the sink will wait for $duration before starting playback).
Fixes long delay until playback starts with flac-in-ogg files.
Fixes#610959.
when we are shutting down, we might still receive state updates from pulseaudio
but since we are unparented we should not do anything with the NULL parent
anymore.
If the FLAC decoder is flushed, its state will be set to frame-sync mode,
which will sync to the next *audio* frame and makes it ignore all headers.
This prevented tags and everything else to show up when using flacdec
in push mode.
Fixes bug #608843.
png_set_gray_1_2_4_to_8() has been deprecated for a while and was
finally removed in libpng 1.4.x. Use png_set_expand_gray_1_2_4_to_8()
instead.
Fixes#608629.
A seek in multi-sink pipeline typically leads to several seek events in a row,
which could lead to sending several newsegments in a row without intermediate
flushing. These would then accumulate, distort rendering times and as such
lead to 'hanging'.
lame's 'lgv' variable is only initialized when the caps
is negotiated, whenever a seek happens before that, it would
attempt to call a function on an empty pointer, causing the crash.
Fixes#603515
Reset the segment info after a flush. We use the segment for handling QoS and if
we don't reset the segment, QoS is basically disabled after a flushing seek.
Use the acquired field of the ringbuffer in get_time to know when we are in an
invalid state. We don't clear the rate flag when releasing the ringbuffer so
this values is not usable.
Avoids some error messages being posted because the pulseaudio connection is
down.
When we can't decode directly into the output buffer, make our temp buffers
only as big as needed instead of allocating for the worst case scenario (well,
we still alloc more than strictly needed for some cases, but significantly
less than before).
Generally decisions on the volume of the stream should be done inside of
PA, not inside of Gst. Only PA knows how volumes translate between
devices and s on.
This patch makes sure that all volumes set via the volume property are
only applied *once* to the underlying stream. After applying them the
client side will not store them anymore. This should make sure that
really only user-triggered volume changes are forwarded to server, but
the client never tries to save/restore the volume internally.
Fixes bug #595231.
pthread does not guarantee that there are no spurious condition variable
wakeups, neither does pa_threaded_mainloop_xxx() which is a wrapper
around it. So we need to loop around the _wait() function to make sure
we get the right wakeup.
Also, unify the order of the wait loops across the file.
If we let the daemon decide freely by passing -1, we end up always getting 20ms.
We want to set this value because in some cases we want to select a higher
latency-time in order to save power.
Fixes#597601
The very first buffer should always have the DISCONT flag set, no
need to warn about that. Only warn if we get a DISCONT buffer in
non-packetised mode and we already have some data.
In case that the pulse daemon runs the source device at a relatively low fixed
fragment size compared to the requested latency-time, configure the ring buffer
segsize to the largest integer multiple of the fragment size that is still
smaller than or equal to the requested latency-time.
Fixes bug #597463.
Don't store the same values in the GstDvDemux. This
fixes a bug where dvdemux would detect a stream as PAL
instead of NTSC, and silently parse it wrong.
There are 5 or 6 AAUX source control packs in a frame, and any
of them could have REC_ST cleared, indicating a recording start
point. libdv only checks the first.
Remove the code to deal with a ringbuffer reset as this code is now in the base
class.
Bump the -base requirement as we need the new baseaudiosink code to function
properly.
This is a live source, therefore:
* Use GST_FORMAT_TIME as the default format
* set_timestamp to True
* properly implement query latency.
This allows expected live usage like : playbin2 uri=dv://
Set the default slave method to the much better skew algorithm. This is the
default in the new base class but we override this here as well for the
upcomming release.
Otherwise that code will just be expanded to nothing when compiled
-DG_DISABLE_ASSERT (PS: why is mainloop_start() called in the init
function and not when changing state to READY?)
For some reason flac doesn't call our metadata callback when we operate
in push mode with unframed input, but that's where we set up the
newsegment event (since that's where we'd get the duration from the
stream info header), so we didn't send a newsegment event at all in this
case. Hack around this by storing a generic newsegment event for now
which will be used if we don't replace it with a better one that
includes the duration.
gst_adapter_peek() will merge buffers as needed, which we can avoid
here since we're doing a memcpy anyway and then flush the copied
data from the adapter right away.
Keep track of the paused state of the source and leave the read function when
paused.
don't wait for a latency update when the delay is not yet known but simply
return 0 instead of blocking.
Keep track of the corked state of the stream.
Fix the state changes.
When seeking in a local flac file (ie. operating pull-based), the decoder
would often just error out after the loop function sees a DECODER_ABORTED
status. This, however, is the read callback's way of telling our loop
function that pull_range failed and streaming should stop, in this case
because of the flush-start event that the seek handler pushed upstream
from the seeking thread. Handle this slightly better by storing the last
flow return from pull_range, so the loop function can evaluate it properly
when it encounters a DECODER_ABORTED and take the right action.
Fixes#578612.
Remove some disabled code in encoder. Try #if 0'ed code and add comments about
why it is disabled. Move idct-method enum to jpeg.c and use in both encoder and
decoder. Add idct-method property to encoder.
We can't wait for the ENTER/LEAVE messages to be be posted because the base
class sometimes calls the start method with the object lock, which would block
the message posting.
Instead, just assume that the message will be posted soon and continue. We'll
have to fix this in the base class.
Previously seekability way always assumed until the first seek actually
failed. Now we assume that all servers are not seekable unless they provide
a Content-Length header. If a seek fails after that we continue to
assume no seekability. Fixes bug #585576.
Emit stream-status messages for the pulse thread.
Don't use our own GCond for signaling but simply use the pulse mainloop
mechanisms for synchronisation.
See #587695
Upper volume limmit was 1000. That appear unneceasrily high. It would also cause
sever distortion if accidentialy used. Now its 10 (~ +15db) which is also in
sync with volume and playbin2.
Since we map the ringbuffer to the pulseaudio internal ringbuffer, flush the
pulseaudio buffer when we are asked to clear the ringbuffer.
This avoids some leftover audio after a seek.
Query the audio format, esp. dvdemux->num_channels, before we use that
variable to allocate the initial buffer. That way we don't accidentally
push a zero-sized buffer as first audio buffer.
Hack around thread-safety issues in GObject and our racy _get_type()
functions (we could easily fix the _get_type() functions, but we still
need to hack around the GObject class races until we require a newer
GLib version, I think).
Some have been replaced by newer ones, others are demoing elements that
don't exist any longer (not in -good anyway), and others have not been
touched in many years and it seem pointless to keep them around.
Removing these files makes sure we don't have any code in our repository
that uses Gtk+ symbols which are to be removed for GNOME3, and as such
will make some script that greps for this kind of stuff give us a clean
bill of code health. Fixes#585757.
A malformed (or simply huge) PNG file can lead to integer overflow in
calculating the size of the output buffer, leading to crashes or buffer
overflows later. Fixes SA35205 security advisory.
Let's be paranoid and make sure we never pass a number that takes up
more than 36 bits to _set_total_samples_estimate(), since libFLAC
expects all the other bits to be zero, and if this is not the case
neighbouring fields in the global stream info header may get messed
up inadvertently, so that flac -d refuses to decode the stream.
See #584455.
When "Content-Type" header is "audio/L16", we need to set the caps on the
outgoing buffers so that downstream elements can have means to detect the
stream type and handle it appropriately. Tested with HTTP stream provided
by pulse-audio's http module (git master).
It was previously sending the bogus buffer which was returned from
the bufferalloc (required for reverse negotiation apparently) instead
of the pending buffer.
This allows to set the Referer header among other things by
adding a "extra-headers" property that takes a GstStructure
with field=string pairs.
Fixes bug #581806.
Store the offset and caps when allocating a buffer during seeking, and then
allocate a new buffer with buffer_alloc before we push it out. This ensures
that in all respects the first buffer decoded during seeking behaves like
all other buffers, including allowing downstream re-negotiation.
lame_set_VBR_quality(), which takes a floating point value for the
quality, has been added only in v3.98. Use lame_set_VBR_q(), which
takes quality as an integer, for older LAME versions.
Fixes#581341.
The libjpeg api says that we need to set the colorspace before we call
_set_defaults(). Indeed, if we don't do that we end up with some very freaky
non-standard quant table and huffman table indexes.
Don't pass a 0 divisor to gst_util_uint64_scale(), or it will complain
in the single image case where fps=0/1 (are we supposed to differentiate
between no fps=still image and fps=0/1=variable rate here btw?)
First we ignore request to fill the ringbuffer which are less then a segment.
The small request where causing stutter.
Then we disable flushing the stream when running against pa 0.9.12 as this
triggers an assertiong in the sound server and terminates it. It does not happen
with 0.9.10 and 0.9.14.
We can use prebuf = 0 to instruct pulse to not pause the stream on underflows.
This way we can remove the underflow callback. We however have to manually
uncork the stream now when we have no available space in the buffer or when we
are writing too far away from the current read_index.
when we switch streams, the clock will reset to 0. Make sure that the provided
clock doesn't get stuck when this happens by keeping an initial offset. We also
need to make sure that we subtract this offset in samples when writing to the
ringbuffer.
If the caps changes, the sink is reset without transitioning through
a PAUSED->PLAYING state change, resulting in a corked stream. This avoids
the problem by checking that the stream is uncorked when writing samples
to it.
When trying to write out a segment, wait until there is enough free space
for the entire segment. This helps to reduce ripple in the clock reporting,
where the app might query the playback position while only half a segment
has been written (and is therefore reported by _delay(), even though
the ring buffer has not yet been advanced)
In the event handler, gst_flac_dec_sink_event(), two functions are called on
the FLAC stream without checking if it has been initialized:
FLAC__stream_decoder_flush()
FLAC__stream_decoder_process_until_end_of_stream()
Both these FLAC__*() functions modify the internal state of the FLAC stream.
Later, when the buffers start flowing, gst_flac_dec_chain() tries to initialize
the stream. the FLAC__stream_decoder_init_stream() call will fail because the
previous calls to FLAC__*() changed the stream state so it is no longer in the
initialized state.
Original commit message from CVS:
* ext/resindvd/plugin.c: (plugin_init):
* ext/resindvd/resindvdsrc.c:
* ext/twolame/gsttwolame.c: (plugin_init):
* gst/aiffparse/aiffparse.c: (plugin_init):
Enable/fix up translations for these plugins.
* po/LINGUAS:
Add 'ca' to LINGUAS.
* po/POTFILES.in:
* po/POTFILES.skip:
Add more files for translation and more files which tools
should skip.
Original commit message from CVS:
* ext/twolame/gsttwolame.c: (gst_two_lame_sink_setcaps),
(gst_two_lame_chain):
* ext/twolame/gsttwolame.h:
Allow raw float samples as input for encoding.
The flacdec API calls the write callback when performing a seek. We cannot yet
push out a buffer at that time so we must keep it and push it out later.
Flush out the upstream part of the pipeline when doing a seek.
Fixes#574275.
g_atomic_int_(get|set) only work on ints and the flags are
an enum (which on most architectures is stored as an int).
Also the way the flags were accessed atomically would still
leave a possible race condition and we don't do it in any
other mixer track implementation, let alone at any other
place where an integer could be changed from different
threads. Removing the g_atomic_int_(get|set) will only
introduce a new race condition on architectures where
integers could be half-written while reading them
which shouldn't be the case for any modern architecture
and if we really care about this we need to use
g_atomic_int_(get|set) at many other places too.
Apart from that g_atomic_int_(set|get) will result in
aliasing warnings if their argument is explicitely
casted to an int *. Fixes bug #571153.
rather than PA thread.
pa_threaded_mainloop_lock() (a.o.) and by extension get_property should
not be done from a PA thread, but the latter may occur as a result of a
property change notification. Fixes#571204 (though current situation
not ideal, e.g. post message rather than signal).
newer pulseaudio.
Fixes: #567794
* Hook pulsesink's volume property up with the stream volume -- not the
sink volume in PA.
* Read the device description directly from the sink instead of going
via the mixer.
* Properly implement _reset() methods for both sink and source to avoid
deadlocks when shutting down a pipeline.
* Replace all simple pa_threaded_mainloop_wait() by proper loops to
guarantee that we wait for the right event in case multiple events are
fired. While this is not strictly necessary in many cases it
certainly is more correct and makes me sleep better at night.
* Replace CHECK_DEAD_GOTO macros with proper functions
* Extend the number of supported channels to 32 since that is the actual
limit in PA.
* Get rid of _dispose() methods since we don't need them.
* Increase the volume property upper limit of the sink to 1000.
* Reset function pointers after we disconnect a stream/context. Better
fix for bug 556986.
* Reset the state of the element properly if open/prepare fails
* Cork the PA stream when the pipeline is paused. This allows the PA
* daemon to
close audio device on pause and thus save a bit of power.
* Set PA stream properties based on GST tags such as GST_TAG_TITLE,
GST_TAG_ARTIST, and so on.
Signed-off-by: Lennart Poettering <lennart@poettering.net>
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.
If libsoup-gnome is found use this as it will give us
the GNOME proxy configuration. Otherwise use normal
libsoup.
The GNOME proxy configuration will only be used if
the proxy properties are not set on souphttpsrc
and if the http_proxy environment variable is not
set.
Fixes bug #552140.
Original commit message from CVS:
Patch by: Lennart Poettering <lennart at poettering dot net>
* ext/pulse/pulseprobe.c: (gst_pulseprobe_new),
(gst_pulseprobe_free):
Fix refcount loop, resulting in a thread leak. Fixes bug #567746.
Original commit message from CVS:
* ext/pulse/pulsesink.c:
* ext/pulse/pulsesink.h:
Use a mutex to protect the current stream pointer, and ignore
callbacks for stream objects that have been destroyed already.
Fixes problems with unprepare/prepare cycles caused by the input
caps changing, without reintroducing bug #556986.
Original commit message from CVS:
* ext/pulse/pulsesink.c: (gst_pulsesink_destroy_stream):
Don't wait for the pulse mainloop when destroying the stream.
Fixes a deadlock when the pulsedaemon goes away while pulsesink
is PLAYING. Fixes bug #556986.
Original commit message from CVS:
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_init),
(gst_smokeenc_getcaps), (gst_smokeenc_setcaps),
(gst_smokeenc_chain), (gst_smokeenc_change_state):
* ext/jpeg/gstsmokeenc.h:
Implement getcaps function.
Set caps on the pad and on all outgoing buffers.
Fixes#565441.
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:
* ext/dv/gstdvdemux.c: (gst_dvdemux_handle_src_event):
Restore previous behaviour of not passing QoS and navigation
events upstream, which presumably wasn't meant to be changed.
Original commit message from CVS:
* ext/dv/gstdvdemux.c: (gst_dvdemux_add_video_pad),
(gst_dvdemux_add_audio_pad), (gst_dvdemux_remove_pads),
(gst_dvdemux_demux_audio), (gst_dvdemux_demux_video),
(gst_dvdemux_chain), (gst_dvdemux_loop),
(gst_dvdemux_change_state):
Add srcpads only when needed and remove them again when going
back to READY. This prevents stalled pipelines if there's no
audio inside the DV stream, which happens for many MXF files.
Original commit message from CVS:
* ext/dv/gstdvdemux.c: (gst_dvdemux_handle_src_event):
Forward all events upstream unless it's something we really
don't handle. This fixes latency configuration of pipelines.
Original commit message from CVS:
* ext/dv/gstdv.c: (plugin_init):
* ext/dv/gstdvdec.c: (gst_dvdec_class_init):
* ext/dv/gstdvdemux.c: (gst_dvdemux_class_init):
Really call dv_init() exactly one time, not one time for
the demuxer and one time for the decoder.
Original commit message from CVS:
Patch by: Zeeshan Ali <zeeshanak at gnome dot org>
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_build_message):
Add transferMode.dnla.org header to HTTP requests as this is
required by the DLNA specs and doesn't hurt in other situations.
Fixes bug #561802.
Original commit message from CVS:
* ext/libpng/gstpngenc.c:
Don't flush downstream after every buffer - that's not what
this libpng callback is for at all!
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:
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:
* 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:
* 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:
* 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:
* 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:
* 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:
* 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:
* 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:
* 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:
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:
* ext/flac/gstflacenc.c: (gst_flac_enc_check_discont):
Actually provide the variables required for the format string.
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_write_callback),
(gst_flac_enc_check_discont), (gst_flac_enc_chain),
(gst_flac_enc_change_state):
* ext/flac/gstflacenc.h:
Handle non-zero start timestamps correctly, mark header packets as
IN_CAPS and print a warning and suggest using audiorate if stream
discontinuities are detected. When FLAC supports flushing the encoder
somehow this should be done for discontinuities instead.
Remove some unused variables from the instance struct.
Original commit message from CVS:
Patch by: Frederic Crozat <fcrozat@mandriva.org>
* ext/dvdread/dvdreadsrc.c: (plugin_init):
* ext/lame/gstlame.c: (plugin_init):
* gst/asfdemux/gstasf.c: (plugin_init):
Make sure gettext returns translations in UTF-8 encoding rather
than in the current locale encoding (#546822).
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_seek_callback):
If seeking failed return the appropiate return value to FLAC.
Otherwise it thinks seeking was successfull and tries to rewrite
parts of the headers which then get appended to the output.
Original commit message from CVS:
Patch by: Frederic Crozat <fcrozat@mandriva.org>
* ext/esd/gstesd.c: (plugin_init):
* ext/flac/gstflac.c: (plugin_init):
* ext/shout2/gstshout2.c: (plugin_init):
* ext/wavpack/gstwavpack.c: (plugin_init):
* sys/oss/gstossaudio.c: (plugin_init):
* sys/v4l2/gstv4l2.c: (plugin_init):
Make sure gettext returns translations in UTF-8 encoding rather
than in the current locale encoding (#546822).
Original commit message from CVS:
* ext/flac/gstflacdec.c:
Add FIXME for 0.11 to simply output everything with width=32 as given
by FLAC and let audioconvert handle the conversions instead of doing
them in flacdec.
Original commit message from CVS:
Based on a patch by: Jonathan Matthew <notverysmart at gmail dot com>
* ext/flac/Makefile.am:
* ext/flac/gstflac.c: (plugin_init):
* ext/flac/gstflactag.c: (gst_flac_tag_setup_interfaces),
(gst_flac_tag_base_init), (gst_flac_tag_class_init),
(gst_flac_tag_dispose), (gst_flac_tag_init),
(gst_flac_tag_sink_setcaps), (gst_flac_tag_chain),
(gst_flac_tag_change_state):
* ext/flac/gstflactag.h:
Port flactag to 0.10, add documentation for it and clean it up a bit.
Fixes bug #413841.
* 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.hierarchy:
* docs/plugins/gst-plugins-good-plugins.interfaces:
* docs/plugins/gst-plugins-good-plugins.prerequisites:
* docs/plugins/inspect/plugin-flac.xml:
* ext/flac/gstflacdec.c: (gst_flac_dec_base_init):
* ext/flac/gstflacdec.h:
* ext/flac/gstflacenc.c: (gst_flac_enc_base_init):
* ext/flac/gstflacenc.h:
Add flactag and flacenc to the documentation and mark
the private parts of the flacdec instance structure as private.
Also use gst_element_class_set_details_simple() in flacdec and
flacenc.
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_query_peer_total_samples),
(gst_flac_enc_sink_setcaps), (gst_flac_enc_write_callback):
Set an estimate for the total number of samples that will be encoded
if possible to help decoders if the streaminfo can't be rewritten
later (like when muxing into Ogg containers).
Add a warning if we get header packets after data packets as those
will get lost when muxing into Ogg, i.e. rewriting the headers doesn't
work.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_dec_metadata_callback),
(gst_flac_dec_write):
Support decoding of all depths between 4 and 32 bits and read the
depth from the streaminfo header if needed. Also support all sampling
rates between 1 and 655350 Hz.
* ext/flac/gstflacenc.c:
(gst_flac_enc_caps_append_structure_with_widths),
(gst_flac_enc_sink_getcaps), (gst_flac_enc_sink_setcaps),
(gst_flac_enc_chain):
* ext/flac/gstflacenc.h:
Support encoding in all bit depths supported by the streamable
subformat (i.e. 8, 12, 16, 20 and 24 bits) and all sampling rates
between 1 Hz and 655350 Hz.
Original commit message from CVS:
* ext/soup/gstsouphttpsrc.c:
* ext/soup/gstsouphttpsrc.h:
Fix seeking race condition in #540300
Patch By: Wouter Cloetens <wouter at mind be>
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_dec_setup_seekable_decoder),
(gst_flac_dec_setup_stream_decoder),
(gst_flac_dec_update_metadata):
Always post the audio-codec tag, not only if other tags are present.
Original commit message from CVS:
* ext/soup/gstsouphttpsrc.c:
Don't throw an error when soup completes a msg with status
'cancelled', as that indicates we cancelled a request while
shutting down or seeking, and it's not an error.
Fixes: #540300 again.
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_class_init),
(gst_lame_get_default_settings):
Use the default for the strict-iso property too.
Allow a bitrate setting of 0, which lets lame choose the default value
and which makes it possible to set the compression-ratio property.
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_init),
(gst_lame_chain), (gst_lame_get_default_settings), (plugin_init):
* ext/lame/gstlame.h:
Get the defaults settings of LAME in the plugin initialization
function and return FALSE here if something goes wrong. This removes
the hacky failing instance init function.
Use LAMEs default value for all settings instead of overwriting some
of them. Overwriting some of them gives unexpected results if one only
sets a preset. Fixes bug #498004.
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_init):
Use LAME's default for the min/max/mean VBR bitrate. Setting our own
defaults will restrict the bitrate when using the presets in a bad way.
Fixes bug #498004.
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_sink_setcaps):
* gst/mpegaudioparse/gstmpegaudioparse.c:
(mp3_type_frame_length_from_header), (mp3_caps_create),
(gst_mp3parse_chain):
Put the MPEG audio version into the caps as "mpegaudioversion".
This is different from "mpegversion".
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_init),
(gst_lame_set_property), (gst_lame_get_property), (gst_lame_setup):
* ext/lame/gstlame.h:
Fix build with lame >= 3.97. The padding type and cwlimit settings
are deprecated now and the function declarations are hidden in the
headers so deprecate the GObject properties for them and remove them
in 0.11. Fixes bug #544039.
Original commit message from CVS:
* ext/Makefile.am:
Finish hooking up pulseaudio plugin to the build.
* ext/pulse/pulsemixerctrl.c:
Fix compilation error.
Original commit message from CVS:
* configure.ac::
* ext/taglib/Makefile.am::
Only use -Wno-attributes (which is there to work around a
bug in the taglib 1.5 headers) if the c++ compiler actually
supports it (#543255).
Original commit message from CVS:
* ext/dv/gstdv.c: (plugin_init):
Marking rank of dvdec as GST_RANK_MARGINAL since it's the slowest
DV decoder available.
Fixes#532393
Original commit message from CVS:
Patch by: Jason Donenfeld <BugZilla at zx2c4 dot com>
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_got_headers_cb):
Fix HTTP auth support with user/password passed via the URI.
Fixes bug #540067.
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
(gst_wavpack_parse_src_query), (gst_wavpack_parse_create_src_pad):
Use G_GINT64_CONSTANT, this fixes the duration query on files without
known length.
Original commit message from CVS:
* docs/plugins/gst-plugins-ugly-plugins-docs.sgml:
* docs/plugins/gst-plugins-ugly-plugins-sections.txt:
* ext/a52dec/gsta52dec.c:
* ext/amrnb/amrnbdec.c:
* ext/amrnb/amrnbenc.c:
* ext/amrnb/amrnbparse.c:
* ext/lame/gstlame.c:
* ext/mad/gstmad.c:
* ext/sidplay/gstsiddec.cc:
* gst/asfdemux/gstrtspwms.c:
* gst/mpegaudioparse/gstxingmux.c:
* gst/realmedia/rademux.c:
* gst/realmedia/rdtmanager.c:
* gst/realmedia/rtspreal.c:
* gst/synaesthesia/gstsynaesthesia.c:
Add missing elements to docs. Restore alphabetical order in section
file. Document mad (it was included in docs already).
Fix doc-markup: use convinience syntax for examples
(produces valid docbook), add several refsec2 when we have several
titles. Fix some types.
Original commit message from CVS:
* ext/lame/gstlame.c:
* ext/sidplay/gstsiddec.cc:
* gst/mpegaudioparse/gstxingmux.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Original commit message from CVS:
* ext/pulse/pulsemixer.c: (gst_pulsemixer_base_init),
(gst_pulsemixer_class_init):
* ext/pulse/pulsesink.c: (gst_pulsesink_base_init),
(gst_pulsesink_class_init), (gst_pulsesink_prepare):
* ext/pulse/pulsesrc.c: (gst_pulsesrc_interface_supported),
(gst_pulsesrc_base_init), (gst_pulsesrc_class_init),
(gst_pulsesrc_prepare):
Some smaller cleanup. Use G_PARAM_STATIC_STRINGS,
gst_element_class_set_details_simple() and fix coding style a bit
more.
Original commit message from CVS:
Patch by: Benjamin Kampmann <benjamin at fluendo dot com>
* ext/cdio/gstcdio.c: (gst_cdio_get_cdtext),
(gst_cdio_add_cdtext_album_tags):
* ext/cdio/gstcdio.h:
* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open):
Also extract album title and album genre from CD-TEXT if
available (#537021).
Original commit message from CVS:
* ext/taglib/Makefile.am::
Add -Wno-attributes to CXXFLAGS to suppress warning caused by
taglib headers (with gcc 4.3.1).
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_extract_picture_buffer):
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
Use new utility functions in libgsttag to process coverart (#512333).
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_dec_write):
We actually support left/side, right/side and mid/side files. The
conversion to normal, interleaved stereo is done by libflac.
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:
* ext/raw1394/gstdv1394src.c:
Document whats first shown on the fdo plugin docs page :)
Original commit message from CVS:
* ext/flac/Makefile.am:
* ext/flac/gstflacdec.c: (gst_flac_dec_write):
Set the channel layout when decoding FLAC files with more than 2
channels as defined by the FLAC spec. Fixes bug #534570.
Also don't try to decode left/side, right/side and mid/side files
as we don't support this at all.
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_write):
When we post an error, we must return -1 to let the parent know that we
cannot write the segment else it will loop and continue to call us again
forever. Patch by Michael Meeks.
Original commit message from CVS:
* ext/wavpack/gstwavpackstreamreader.c:
* tests/examples/spectrum/demo-audiotest.c:
* tests/examples/spectrum/demo-osssrc.c:
Fix some compiler warnings.
Original commit message from CVS:
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_reset),
(gst_gconf_audio_src_change_state):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset),
(gst_gconf_video_sink_change_state):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_reset),
(gst_gconf_video_src_change_state):
* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset),
(gst_switch_commit_new_kid), (gst_switch_sink_change_state):
When we can't create a fakesink/fakesrc complain instead of unreffing
NULL pointers and crashing later. See bug #530535.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* ext/speex/gstspeexenc.c: (gst_speex_enc_sink_getcaps),
(gst_speex_enc_init), (gst_speex_enc_chain):
Add negotiation for the speex channels and rate. Fixes#465146.
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (speex_dec_sink_event),
(speex_dec_chain_parse_data):
Produce concealment data when time progresses in a segment update.
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data),
(speex_dec_chain):
Try to preserve input timestamps when we can.
Do beginnings of error concealment.
Original commit message from CVS:
* ext/soup/gstsouphttpsrc.c: (plugin_init):
Give souphttpsrc GST_RANK_PRIMARY to make it the default HTTP source
over gnome-vfs and everything else. Fixes bug #527848.
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_header):
Fix bounds checking of mode in Speex header, which may
produce negative numbers in speex < 1.1.12
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_response_cb):
Only ignore actual redirects not all responses when in state
GST_SOUP_HTTP_SRC_SESSION_IO_STATUS_RUNNING. Fixes bug #526337.
Original commit message from CVS:
* ext/hal/hal.c: (gst_hal_get_alsa_element):
Don't munge device string to 'default:x' for capture devices.
Fixes#525833.
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_index_entry_free):
Always use GSlice as we actually depend on GLib 2.12 already.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init):
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_base_init):
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_index_entry_new),
(gst_wavpack_parse_index_entry_free),
(gst_wavpack_parse_base_init),
(gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset):
Use GSlice for allocating index entries and use
gst_element_class_set_details_simple().
Original commit message from CVS:
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_got_headers_cb),
(gst_soup_http_src_chunk_allocator),
(gst_soup_http_src_got_chunk_cb),
(gst_soup_http_src_uri_get_protocols):
Don't autoplug souphttpsrc for dav/davs. This is better handled by
GIO and GnomeVFS as they provide authentication.
Don't leak the icy caps if we already set them and get a new
icy-metaint header.
Try harder to set the icy caps on the output buffer to have correct
caps for the first buffer already.
* tests/check/elements/souphttpsrc.c: (got_buffer),
(GST_START_TEST):
Check that we get a buffer with application/x-icy caps if iradio-mode
is enabled and we have an icecast URL.
Original commit message from CVS:
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_chunk_allocator):
Actually set the icy caps on our src pad if we have icecast data.
Fixes bug #523854.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_init),
(gst_soup_http_src_finished_cb), (gst_soup_http_src_response_cb),
(gst_soup_http_src_build_message), (gst_soup_http_src_create):
* ext/soup/gstsouphttpsrc.h:
Try to resume on server disconnect. Fixes bug #522134.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* ext/speex/gstspeexenc.c: (gst_speex_enc_chain):
Unref the buffers only once when handling not-negotiated errors.
Fixes bug #520764.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_class_init),
(gst_soup_http_src_init), (gst_soup_http_src_dispose),
(gst_soup_http_src_set_property), (gst_soup_http_src_get_property),
(gst_soup_http_src_create):
* ext/soup/gstsouphttpsrc.h:
* tests/check/elements/souphttpsrc.c: (run_test), (GST_START_TEST),
(souphttpsrc_suite):
Add support for specifying a list of cookies to be passed in
the HTTP request. Fixes bug #518722.
Original commit message from CVS:
* configure.ac:
* ext/taglib/Makefile.am:
Check for and define ERROR_CXXFLAGS and use them when building
C++ code (#516509).
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_sink_setcaps):
Fix broken GST_ELEMENT_ERROR macro, fixes compile with the Sun
Workshop 12 compiler, but probably also crashes (#517985).
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* configure.ac:
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_cancel_message),
(gst_soup_http_src_finished_cb), (gst_soup_http_src_chunk_free),
(gst_soup_http_src_chunk_allocator),
(gst_soup_http_src_got_chunk_cb), (gst_soup_http_src_create),
(gst_soup_http_src_start), (gst_soup_http_src_set_proxy):
* ext/soup/gstsouphttpsrc.h:
Implement zero-copy and make the buffer size configurable.
Prefix proxy URIs with "http://" if they don't start with it
already and catch errors earlier, fixes hanging in some situations.
Fixes bug #514948.
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query),
(gst_wavpack_parse_create_src_pad):
* ext/wavpack/gstwavpackparse.h:
Always report the duration if we know it in push mode and don't
return 0 just to make totem believe we can't seek in push mode.
Newer totem version use the SEEKING query which properly reports
if we can seek or not.
Original commit message from CVS:
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
* tests/check/Makefile.am:
* tests/check/gst-plugins-good.supp:
Add a few libjpeg suppressions and initialize a variable to
make smokeenc valgrind clean. Fixes bug #515701.
Original commit message from CVS:
* ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
Use and unset the GError when pipeline creation fails instead of
simply leaking it. Fixes bug #515704.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_got_chunk_cb),
(gst_soup_http_src_create):
Fix memory leak and improve debugging a bit.
Original commit message from CVS:
Patch by: John Millikin <jmillikin at gmail dot com>
* ext/flac/gstflacdec.c: (gst_flac_dec_scan_for_last_block),
(gst_flac_extract_picture_buffer), (gst_flac_dec_metadata_callback):
Fix extraction of picture blocks with newer libflac versions again:
FLAC__METADATA_TYPE_PICTURE is an enum, not a define (#513628).
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_init):
Let the proxy property default to the content of the $http_proxy
environment variable.
Original commit message from CVS:
Patch by: Alessandro Decina <alessandro at nnva dot org>
* ext/libpng/gstpngenc.c: (user_write_data), (gst_pngenc_chain):
* ext/libpng/gstpngenc.h:
Preallocate the output buffer so that g_memdup() and
gst_buffer_merge() aren't needed anymore. This greatly improves
performances and fixes#512544.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (soup_got_headers):
Report the size of the stream as the total size instead of
the remaining Content-Length, which is wrong after a seek.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (soup_got_headers):
Correctly set duration on the GstBaseSrc segment when we know it
to fix failing the duration query.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_class_init),
(gst_souphttp_src_init), (gst_souphttp_src_create),
(gst_souphttp_src_is_seekable), (gst_souphttp_src_do_seek),
(soup_add_range_header), (soup_got_headers), (soup_got_chunk):
* ext/soup/gstsouphttpsrc.h:
Add support for seeking to souphttpsrc. Fixes bug #502335.
Original commit message from CVS:
* ext/flac/gstflacdec.c:
Fix compilation against flac 1.1.2 (as on debian stable), where
the picture metadata defines and structs don't exist yet.
Fixes#509301.
Original commit message from CVS:
* ext/lame/gstlame.c:
* ext/lame/gstlame.h:
Fix the case where you initially have stereo input, and so lame's
mode is not set to mono, and then you get input with mono audio and
soon after you get stereo input again. What happened before this
commit is that it would keep the encoding mode as mono. It should
change it back to the one requested by the app (or the default one)
if not requested.
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_sink_event):
Use gst_util_uint64_scale instead of gst_util_uint64_scale_int
as 8 * GST_SECOND is too large for int.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at syknet dot be>
* ext/lame/gstlame.c: (gst_lame_sink_setcaps),
(gst_lame_sink_event):
Correctly set number of channels when using mono-encoding mode
and fix the duration calculation of the EOS buffer.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_extract_picture_buffer):
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
Generate the image-type values correctly. Leave them out of the caps
when outputting a "preview image" tag, since it only makes sense
to have one of those - the type is irrelevant.
* sys/sunaudio/gstsunaudiomixerctrl.c:
(gst_sunaudiomixer_ctrl_open):
If we can, mark the mixer multiple open when we use it, in case
(for some reason) the process wants to open it again elsewhere.
Original commit message from CVS:
* ext/taglib/gstapev2mux.h:
* ext/taglib/gstid3v2mux.h:
Remove useless typedefs without new type name. Fixes a warning with
gcc 4.3.
Original commit message from CVS:
Patch by: John Millikin <jmillikin at gmail dot com>
* ext/flac/gstflacdec.c: (gst_flac_dec_setup_seekable_decoder),
(gst_flac_dec_setup_stream_decoder),
(gst_flac_normalize_picture_mime_type),
(gst_flac_extract_picture_buffer),
(gst_flac_dec_metadata_callback):
Emit metadata messages when a PICTURE block is encountered.
Fixes#506715.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_dec_write):
Fix 'xyz may be used uninitialized' compiler warnings caused
by broken g_assert_not_reached() macro in GLib-2.15.x and don't
abort() in any case but properly report the error.
Original commit message from CVS:
* ext/soup/Makefile.am:
* ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_get_property),
(gst_souphttp_src_unicodify), (soup_got_headers):
Use gst_tag_freeform_string_to_utf8() and post radio station
info as tags on the bus.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_dec_reset_decoders),
(gst_flac_dec_update_metadata), (gst_flac_dec_metadata_callback),
(gst_flac_dec_write):
* ext/flac/gstflacdec.h:
Remove some unused vars.
Do more cleanup of leftover events and tags.
Output tags after the segment event. Fixes#504018.
Original commit message from CVS:
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
Actually drop the buffers which are outside the currently configured
segment instead of just emitting a WARNING.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_dec_metadata_callback),
(gst_flac_dec_write):
* ext/flac/gstflacdec.h:
Send segments from the streaming thread. Fixes#502187.
Fix segment seeking and a bunch of other seeking cases.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (_do_init),
(gst_souphttp_src_class_init), (gst_souphttp_src_init),
(gst_souphttp_src_dispose), (gst_souphttp_src_set_property),
(gst_souphttp_src_get_property), (unicodify),
(gst_souphttp_src_unicodify), (gst_souphttp_src_create),
(gst_souphttp_src_start), (gst_souphttp_src_stop),
(gst_souphttp_src_unlock), (gst_souphttp_src_unlock_stop),
(gst_souphttp_src_get_size), (gst_souphttp_src_is_seekable),
(soup_got_headers), (soup_got_body), (soup_finished),
(soup_got_chunk), (soup_response), (soup_parse_status),
(gst_souphttp_src_uri_get_type),
(gst_souphttp_src_uri_get_protocols),
(gst_souphttp_src_uri_get_uri), (gst_souphttp_src_uri_set_uri),
(gst_souphttp_src_uri_handler_init):
* ext/soup/gstsouphttpsrc.h:
Do not try to unpause I/O in the "queued" state.
Reorganise a bunch of things and cleanups.
Uses G_GUINT64_FORMAT instead of hard-coding %llu.
See #502335.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Also print a useful error message with the old Wavpack API
if possible.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c:
More build fixes for old libwavpack versions: include config.h so
that WAVPACK_OLD_API is actually defined as detected; only use
WavpackGetErrorMessage if it is available. This fixes the build
on debian stable for me.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_create_src_pad):
Workaround the non-existance of WavpackGetChannelMask in Wavpack
versions below 4.40.0.
Original commit message from CVS:
Based on a patch by: Kwang Yul Seo <kwangyul dot seo at gmail dot com>
* configure.ac:
* ext/cairo/gsttimeoverlay.c:
(gst_cairo_time_overlay_print_smpte_time):
Fix compilation with MSVC by using gst_util_guint64_to_gdouble()
and checking for rint() and implementing it ourself if it doesn't
exist.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* configure.ac:
Bump libsoup requirement as libsoup does not support async client
operation prior to version 2.2.104 and it has some leaks.
* ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_class_init),
(gst_souphttp_src_init), (gst_souphttp_src_dispose),
(gst_souphttp_src_set_property), (gst_souphttp_src_create),
(gst_souphttp_src_start), (gst_souphttp_src_stop),
(gst_souphttp_src_unlock), (gst_souphttp_src_unlock_stop),
(gst_souphttp_src_get_size), (soup_got_headers), (soup_got_body),
(soup_finished), (soup_got_chunk), (soup_response),
(soup_session_close):
* ext/soup/gstsouphttpsrc.h:
Implement unlock().
Picks up the size from the Content-Length header and emit a duration
message.
Don't leak the GMainContext object.
Fixes#500099.
Original commit message from CVS:
* ext/wavpack/gstwavpackcommon.c: (gst_wavpack_set_channel_layout):
Also set the channel layout on the Wavpack caps if we're having
a mono layout. Of course only do it for "audio/x-wavpack".
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* ext/libpng/gstpngenc.c:
Don't leak buffer data memory. Fixes#498395.
Original commit message from CVS:
2007-11-20 Julien MOUTTE <julien@moutte.net>
* ext/taglib/gsttaglibmux.c: (gst_tag_lib_mux_render_tag),
(gst_tag_lib_mux_adjust_event_offsets):
* gst/qtdemux/qtdemux.c: (qtdemux_parse_theora_extension):
* sys/osxaudio/Makefile.am:
* sys/osxvideo/cocoawindow.h:
* sys/osxvideo/cocoawindow.m: Fix build on Mac OS X 10.5
Original commit message from CVS:
* ext/taglib/gstid3v2mux.cc: (add_musicbrainz_tag), (add_funcs):
Write GST_TAG_MUSICBRAINZ_DISCID and GST_TAG_CDDA_CDDB_DISCID
into ID3v2 TXXX frames (fixes#347848).
Original commit message from CVS:
Patch by: Julien Puydt <julien dot puydt at laposte net>
* ext/raw1394/Makefile.am:
* ext/raw1394/gst1394probe.c: (gst_1394_get_guid_array),
(gst_1394_property_probe_get_properties),
(gst_1394_property_probe_probe_property),
(gst_1394_property_probe_needs_probe),
(gst_1394_property_probe_get_values),
(gst_1394_property_probe_interface_init),
(gst_1394_type_add_property_probe_interface):
* ext/raw1394/gst1394probe.h: (GST_1394_PROBE_H):
* ext/raw1394/gstdv1394src.c: (_do_init), (gst_dv1394src_class_init),
(gst_dv1394src_init), (gst_dv1394src_dispose),
(gst_dv1394src_set_property), (gst_dv1394src_get_property),
(gst_dv1394src_discover_avc_node), (gst_dv1394src_query),
(gst_dv1394src_update_device_name):
* ext/raw1394/gstdv1394src.h:
Implement GstPropertyProbe interface and add "device-name" property,
so applications can use this to probe for available devices in the
same way they can already with v4lsrc and v4l2src (however horrible
this property probe interface may be). Fixes#358841.
Original commit message from CVS:
* ext/cairo/gsttextoverlay.c: (gst_text_overlay_font_init):
Implement minimal parsing of the passed pango font description
string, so passing a font size works the same as with the
pango textoverlay plugin; fixes#455086.
(Maybe we could just use pangocairo here at some point).
Original commit message from CVS:
* ext/taglib/gstid3v2mux.cc:
Add support for license/copyright URI tags (ID3v2 WCOP frame).
Prerequisite for #447000.
Original commit message from CVS:
* ext/flac/gstflacenc.c:
* ext/flac/gstflacenc.h:
Save the flow return from the last gst_pad_push() and
make sure we pass the right flow return value upstream
in the case of failure; minor clean-ups.
Original commit message from CVS:
* ext/taglib/gstapev2mux.cc:
* ext/taglib/gstid3v2mux.cc:
* gst/apetag/gstapedemux.c:
Add support for the new GST_TAG_COMPOSER (#459809).
Original commit message from CVS:
* ext/lame/gstlame.c:
Allow fixing the sample rate lame converts to by negotiating fixed
sample rate on the src pad caps.
Add docs for it.
* tests/check/Makefile.am:
* tests/check/pipelines/lame.c:
Add a check for it.
Original commit message from CVS:
* ext/cairo/gsttextoverlay.c:
Add info about static leak.
* tests/check/Makefile.am:
* tests/check/generic/states.c:
Improved state change unit test.
Original commit message from CVS:
* ext/taglib/gstapev2mux.cc:
* ext/taglib/gstid3v2mux.cc:
Work around compiler warnings with g++-4.2 when assigning a
string constant to a gchar * (partially fixes#478092).
Original commit message from CVS:
* ext/taglib/gstapev2mux.cc:
* ext/taglib/gstapev2mux.h:
* ext/taglib/gsttaglibmux.c:
* tests/check/elements/apev2mux.c:
Update my mail address.
Original commit message from CVS:
* ext/gconf/gstgconfaudiosink.c:
Fix warning when building without debug.
* sys/oss/gstossmixertrack.c:
Use const like in alsamixertrack.c (fixes warnings).
Original commit message from CVS:
* ext/gconf/gstswitchsink.c:
If the new kid element fails to change state for some reason
(e.g. esdsink not being able to connect to the sound server),
forward the error message it posted on the bus instead of just
posting a generic 'Internal state change error: please file a
bug' error message. Fixes#471364.
Original commit message from CVS:
* ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
Handle a NULL gconf key gracefully by rendering the default element.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Don't unref the outgoing buffer twice when dropping it because it's
outside of the segment.
Original commit message from CVS:
* configure.ac:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
(gst_wavpack_dec_chain), (gst_wavpack_dec_sink_event):
Use the new buffer clipping function from gstaudio here and
require gst-plugins-base CVS.
* tests/check/elements/wavpackdec.c: (GST_START_TEST):
For framed Wavpack buffers we require a valid timestamp.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c:
(gst_wavpack_dec_clip_outgoing_buffer):
Fix buffer clipping to correctly clip to the segment stop.
Original commit message from CVS:
* ext/libpng/gstpngdec.c: (gst_pngdec_caps_create_and_set):
Remove endianness-flipping hack that seems to have been required
only because of a bug in ffmpegcolorspace.
Partially Fixes: #451908
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_index_get_last_entry),
(gst_wavpack_parse_index_get_entry_from_sample),
(gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset),
(gst_wavpack_parse_scan_to_find_sample):
* ext/wavpack/gstwavpackparse.h:
Use a GSList for the GArray that is used like a list anyway.
Original commit message from CVS:
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),
(gst_gdk_pixbuf_class_init), (gst_gdk_pixbuf_flush),
(gst_gdk_pixbuf_sink_event), (gst_gdk_pixbuf_change_state):
Add state change function where we set 0/1 as default framerate in
case our setcaps function isn't called, like it might not in a
filesrc ! gdkpixbufdec scenario. Fixes assertion triggered by
gdkpixbufdec trying to create caps with a 0/0 framerate.
Also post an error message on the bus if gst_pad_push() fails when
called from our sink event handler (+1 for flow returns for event
functions in 0.11) instead of failing silently.
Original commit message from CVS:
* ext/gconf/gconf.h:
Make the prototype of gst_gconf_get_key_for_sink_profile
match the implementation.
Patch by: Damien Carbery <damien dot carbery at sun dot com>
Fixes: #449747
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_sink_set_caps):
Remove workaround for bug #421543. This is fixed in core 0.10.13 and
not necessary anymore as we need at least that core version.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
(gst_wavpack_dec_chain):
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
(gst_wavpack_parse_push_buffer):
* ext/wavpack/gstwavpackparse.h:
Improve discont handling by checking if the next Wavpack block has
the expected, following block index.
Original commit message from CVS:
* ext/dv/gstdvdemux.c: (gst_dvdemux_loop):
* ext/libpng/gstpngdec.c: (user_read_data), (gst_pngdec_task):
When operating in pull mode, error out correct on not-linked.
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data):
Use different variables for nested for loops so that the outer loop
functions properly and speex files with multiple frames per buffer work
properly.
Fixes#441408.
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_init),
(notgst_value_array_append_buffer),
(gst_flac_enc_process_stream_headers),
(gst_flac_enc_write_callback), (gst_flac_enc_chain),
(gst_flac_enc_change_state):
* ext/flac/gstflacenc.h:
Collect headers, add "streamheader" field to output caps and set
BUFFER_IN_CAPS flag on pushed header buffers. That way oggmux
produces output according to the official FLAC-to-Ogg mapping
instead of completely broken files. Fixes#426044.
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_sink_setcaps),
(gst_flac_enc_chain):
Don't crash in chain function if setcaps hasn't been called.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
(gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property):
Specify and use properties as unsigned int that are an unsigned int.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
(gst_wavpack_enc_init), (gst_wavpack_enc_set_wp_config),
(gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property):
* ext/wavpack/gstwavpackenc.h:
Fixup docs, make the bitrate property an int as it should be and
allow to set the different extra processing modes instead of only
allowing none and the default one.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c:
Add missing audioconverts in the example pipelines of wavpackenc. As
the wavpack stuff now needs input with 32 bit width (and random depth)
this is needed now. The example pipelines for the parser and decoder
are still fine.
Original commit message from CVS:
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_uri_set_uri):
Replace direct comparison of a string with the string literal "" with
a comparison of the first character with '\0'. Fixes#438926.
Original commit message from CVS:
* ext/wavpack/gstwavpack.c: (plugin_init):
Call bindtextdomain() to get localized strings.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
(gst_wavpack_parse_handle_seek_event),
(gst_wavpack_parse_push_buffer), (gst_wavpack_parse_chain):
* ext/wavpack/gstwavpackparse.h:
Handle DISCONT buffers by correctly setting the DISCONT flag
on outgoing buffers when necessary.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_handle_seek_event)
Send newsegment from the streaming thread.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_dec_loop):
Correctly post an error on the bus if something went wrong in the loop
function. This fixes a few cases where the task was paused and nothing
happened anymore.
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_handle_seek_event):
Remove old workaround that was needed when seeking after the last
sample. With the fixed error handling this works now as expected
without pushing the last sample although it wasn't requested.
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_handle_seek_event):
Handle segment seeks in the seek event handler, correctly work with
stop position == -1 and instead of stopping the task on seek just
pause it.
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_pull_buffer),
(gst_wavpack_parse_create_src_pad),
(gst_wavpack_parse_resync_loop), (gst_wavpack_parse_loop),
(gst_wavpack_parse_chain):
Correctly handle errors, especially in the loop function. Before it
was easy to get the task paused but no error being posted on the bus.
Original commit message from CVS:
* ext/libpng/gstpngdec.c: (gst_pngdec_task):
If we get a fatal flow return in the loop function, first post the
error message and only then send the EOS event downstream, otherwise
applications might get an eos message before the error message and
think everything was ok (related to #429319).
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
(gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps),
(gst_wavpack_dec_clip_outgoing_buffer),
(gst_wavpack_dec_post_tags), (gst_wavpack_dec_chain):
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset),
(gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config),
(gst_wavpack_enc_chain):
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.c:
Don't play audioconvert. As wavpack wants/outputs all samples with
width==32 and depth=[1,32] accept this and let audioconvert convert
to accepted formats instead of doing it in the element for n*8 depths.
This also adds support for non-n*8 depths and prevents some useless
memory allocations. Fixes#421598
Also add a workaround for bug #421542 in wavpackenc for now...
* tests/check/elements/wavpackdec.c: (GST_START_TEST):
* tests/check/elements/wavpackenc.c: (GST_START_TEST):
* tests/check/elements/wavpackparse.c: (GST_START_TEST):
Consider the change above in the unit tests and test if the correct
caps are accepted and set. Also check for GST_BUFFER_OFFSET_END in
the wavpackparse unit test.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init),
(gst_wavpack_dec_sink_set_caps):
Set caps on the src pad as soon as possible.
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackcommon.h:
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.h:
Fix indention. gst-indent is now called by cicl.
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Revert last commit, preventing infinite plugging loops with ranks
is no clean solution and in general there's no reason why one wants
to parse framed wavpack data again.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_push_block):
Send the new segment event in time format instead of bytes. This
allows "wavpackenc ! wavpackdec ! someaudiosink" pipelines.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Accept framed and non-framed input, wavpackparse doesn't care. To
prevent "wavpackparse ! wavpackparse ! ..." pipelines lower the
rank of wavpackparse by one. This allows "wavpackenc ! wavpackparse !
..." pipelines.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Revert to use gst_pad_alloc_buffer() here. We can and should use it.
Thanks to Jan and Mike for noticing my mistake.
Original commit message from CVS:
Patch by: Christophe Dehais <christophe dot dehais at gmail dot com>
* ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
Accept complex pipeline descriptions as an audio profile instead of just
a single element. Fixes#420658.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
(gst_wavpack_enc_init), (gst_wavpack_enc_chain),
(gst_wavpack_enc_rewrite_first_block):
* ext/wavpack/gstwavpackenc.h:
Put the write helpers into the GstWavpackEnc struct directly and not
as a pointer to save two small, but useless mallocs. This also makes
it possible to drop the finalize method.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_push_buffer):
For consistency reasons also set GST_BUFFER_OFFSET_END on the outgoing
buffers the same way wavpackenc does it.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Don't use gst_pad_alloc_buffer() as we might clip the buffer later and
BaseTransform-based elements will likely break because of wrong
unit-size. Also plug a possible memleak that happens when decoding
fails for some reason.
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_setup):
Disable the bitrate checking when the user has requested
Free Format mode, as all bitrates less than the maximum
are valid then.
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_setcaps),
(gst_dvdec_src_negotiate), (gst_dvdec_chain),
(gst_dvdec_change_state):
* ext/dv/gstdvdec.h:
Infer pixel-aspect-ratio from the video frame format if it isn't
provided by the container, as happens when playing DV from AVI
or Quicktime containers.
Patch by: Wim Taymans <wim@fluendo.com>
Fixes#380944
Original commit message from CVS:
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_start):
Free handles that we allocated when exiting via the error paths.
Original commit message from CVS:
* ext/wavpack/gstwavpack.c: (plugin_init):
* ext/wavpack/gstwavpackcommon.c:
Use a general wavpack debug category for common code.
* ext/wavpack/gstwavpackstreamreader.c:
(gst_wavpack_stream_reader_set_pos_abs),
(gst_wavpack_stream_reader_set_pos_rel),
(gst_wavpack_stream_reader_write_bytes):
Use the general wavpack debug category here too and add debug
output to the functions that should not be called at all by
the wavpack library.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_plugin_init):
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_plugin_init):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Change debugging category names to conform to the conventions.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
(gst_switch_commit_new_kid):
Fix up the dispose logic so it doesn't leak, and fix setting of
the child state so that we don't set a child to our current state
just as we are changing it to something else.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
(gst_switch_commit_new_kid):
Fix up the reference counting of the child elements.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset),
(gst_switch_commit_new_kid), (gst_switch_sink_set_child):
Install fakesink in NULL by fixing some broken logic. This obviates
the need to manually set _IS_SINK.
Add some comments and remove a little cruft while I'm at it.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset):
Mark us as a sink when we have no fakesink in NULL. Fixes#414887.
Original commit message from CVS:
* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open),
(gst_cdio_cdda_src_finalize):
Make sure we always destroy our libcdio handle.
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_init):
Use gst_pad_new_from_static_template instead of
static_pad_template_get+pad_new.
Original commit message from CVS:
Patch by: Loïc Minier <lool+gnome at via ecp fr>
* ext/libcaca/Makefile.am:
* gst/debug/Makefile.am:
Don't mix tabs and spaces (#414168).
Original commit message from CVS:
* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_probe_devices),
(gst_cdio_cdda_src_read_sector), (gst_cdio_cdda_src_open),
(gst_cdio_cdda_src_finalize):
Small code cleanups.
Don't use pad_alloc as the base class cannot deal with the error codes.
Original commit message from CVS:
* ext/hal/gsthalaudiosink.c: (do_toggle_element):
* ext/hal/gsthalaudiosrc.c: (do_toggle_element):
Having NULL as UDI previously selected the default sink/src. Change
this back but mention it in the debug output.
* ext/hal/hal.c: (gst_hal_get_alsa_element),
(gst_hal_get_oss_element), (gst_hal_get_string),
(gst_hal_render_bin_from_udi), (gst_hal_get_audio_sink),
(gst_hal_get_audio_src):
* ext/hal/hal.h:
Refactor a bit, check all error conditions, greatly improve debugging
and fix some possible memory leaks. Also implement OSS support
and allow specifying an UDI that points to a real device. For this the
child device which supports ALSA (preferred) or OSS is used.
As a side effect this makes it impossible now to get a alsasink in
halaudiosrc and a alsasrc in halaudiosink.
Original commit message from CVS:
* ext/hal/gsthalaudiosink.c: (do_toggle_element):
* ext/hal/gsthalaudiosrc.c: (do_toggle_element):
Check if the device UDI is set before trying to query HAL
about it and give a useful error message if it wasn't set.
* ext/hal/hal.c: (gst_hal_get_string):
Don't query HAL for NULL UDIs. Passing NULL as UDI to HAL
gives an assertion failure in D-Bus when running with
DBUS_FATAL_WARNINGS=1.
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_init),
(gst_lame_setup):
Display sensible defaults and limits for the
vbr-min/max/mean properties. Fix the 'hard-limit' VBR min
property - it's supposed to be a boolean 0/1 value.
Original commit message from CVS:
* ext/lame/gstlame.c:
Initialise the variables so gcc doesn't complain about possibly
uninitialised uses, even though they can't actually happen.
Original commit message from CVS:
* ext/shout2/gstshout2.c: (gst_shout2send_class_init),
(gst_shout2send_init), (gst_shout2send_start),
(gst_shout2send_set_property), (gst_shout2send_get_property):
* ext/shout2/gstshout2.h:
Add a property for username.
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_sink_setcaps),
(gst_lame_set_property), (gst_lame_setup):
Fix up bitrate checking macro. Make it give us a
GST_ELEMENT_WARNING message so the application has a chance of
reporting this to the user. Move the checking to _setup, so we
are sure it runs in the READY state, when we hope to have a pipeline
and a bus that is not flushing.
This fixes e.g. using 96 kbit/sec as a bitrate.
Original commit message from CVS:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset),
(do_change_child):
Don't reset the profile when going switching states, as it makes
the element non-reusable.
Original commit message from CVS:
* ext/gconf/Makefile.am:
* ext/gconf/gconf.c: (gst_gconf_get_string),
(gst_gconf_get_key_for_sink_profile), (gst_gconf_set_string),
(gst_gconf_render_bin_with_default):
* ext/gconf/gconf.h:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init),
(gst_gconf_audio_sink_reset), (gst_gconf_audio_sink_init),
(gst_gconf_audio_sink_dispose), (do_change_child),
(gst_gconf_switch_profile), (gst_gconf_audio_sink_set_property),
(cb_change_child), (gst_gconf_audio_sink_change_state):
* ext/gconf/gstgconfaudiosink.h:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_base_init),
(gst_switch_sink_class_init), (gst_switch_sink_reset),
(gst_switch_sink_init), (gst_switch_sink_dispose),
(gst_switch_commit_new_kid), (gst_switch_sink_set_child),
(gst_switch_sink_set_property), (gst_switch_sink_handle_event),
(gst_switch_sink_get_property), (gst_switch_sink_change_state):
* ext/gconf/gstswitchsink.h:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
(gst_auto_audio_sink_clear_kid), (gst_auto_audio_sink_reset),
(gst_auto_audio_sink_detect):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
(gst_auto_video_sink_clear_kid), (gst_auto_video_sink_reset),
(gst_auto_video_sink_detect):
Re-factor the gconfaudiosink into a "GstSwitchSink" base class
and a child that implements the GConf key monitoring. The end goal of
this is an audio sink that can be changed on the fly, but at the
moment it still only changes on the next READY transition.
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_sink_event), (gst_lame_chain),
(gst_lame_change_state):
* ext/lame/gstlame.h:
On receiving EOS, we try to push a last buffer with the remaining
samples. Don't do that if we got an unclean flow return on the last
gst_pad_push(), downstream might not handle this very gracefully
(see #403168).
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
Pass flow returns upstream (helps #403168).
Original commit message from CVS:
* ext/hal/hal.c: (gst_hal_get_string):
* ext/hal/hal.h:
Some small cleanups; deal with errors when parsing the HAL ALSA
capabilities a bit better.
Original commit message from CVS:
* ext/gconf/gconf.c: (gst_gconf_get_key_for_sink_profile),
(gst_gconf_render_bin_from_key),
(gst_gconf_get_default_audio_sink):
* ext/gconf/gconf.h:
* ext/gconf/gstgconfaudiosink.c: (get_gconf_key_for_profile),
(do_toggle_element), (gst_gconf_audio_sink_set_property),
(gst_gconf_audio_sink_get_property):
In gconfaudiosink, get the right key as the old key in do_toggle
(ie. one dependent on the profile selected). Log some more stuff so
we can see what's actually going on.
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query),
(gst_wavpack_parse_handle_seek_event),
(gst_wavpack_parse_create_src_pad):
Fix a off by one that leads to the duration reported as one
sample less than it is
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query):
Fix the SEEKING query. We can seek if we are in pull mode, not the
other way around. Also set the correct format in the seeking query and
handle the case where the headers are not read yet and we can't say
anything about our seeking capabilities.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init):
Fix spelling in 2 places: It's called Wavpack, not WavePack.
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo@circular-chaos.org>
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_mode_get_type),
(gst_wavpack_enc_correction_mode_get_type),
(gst_wavpack_enc_joint_stereo_mode_get_type):
Minor clean-up: use enum values instead of hardcoded constants (#395536).
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry fr>
* ext/jpeg/gstjpegdec.c:
* ext/jpeg/gstjpegenc.c:
* ext/jpeg/smokecodec.c:
These libjpeg callbacks should return a 'boolean' (unsigned char
apparently) and not a 'gboolean' (which maps to gint). Fixes
warnings when compiling with MingW (#393427).
* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
Use ioctlsocket on win32.
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Some printf format fixes for win32.
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_delay):
Don't return bogus values when esd_get_delay() fails for some
reason (#392189).
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_sink_setcaps), (gst_lame_chain):
warn when outgoing sample rate is different from incoming
Original commit message from CVS:
* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setcaps),
(gst_signal_processor_event):
Reset flow_state back to _OK after a flush stop so that we exit our
error state after the flush. Fixes#374213
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.h:
Use local copy of md5.h, as it disappeared in recent wavpack
installs.
Patch by: Sebastian Dröge <slomo at ubuntu dot com>
Fixes: #387076
Original commit message from CVS:
Based on patch by: Roland Kay <roland.kay at ox compsoc net>
* ext/lame/gstlame.c: (gst_lame_init), (gst_lame_chain),
(gst_lame_setup):
* ext/lame/gstlame.h:
Fix leak (by calling lame_init_params() before lame_close()); handle
NULL return from lame_init() more gracefully. Fixes#385311.
Original commit message from CVS:
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_init),
(gst_gconf_audio_src_dispose), (do_toggle_element):
* ext/gconf/gstgconfaudiosrc.h:
Remove gconf notify hook when the gconfaudiosrc element is
destroyed, otherwise the callback may be called on an
already-destroyed instance and bad things happen. Should fix
#378184.
Also ignore gconf key changes when the source is already running.
Original commit message from CVS:
* ext/taglib/gstid3v2mux.cc:
Don't attempt to write a NULL frame into the ID3 tag set when the
createFrame method returned NULL.
Fixes: #381857
Patch by: Jonathan Matthew <jonathan at 0kaolin wh9 net >
Original commit message from CVS:
* ext/libcaca/gstcacasink.c: (gst_cacasink_class_init):
Fix width and height properties.
* ext/libcaca/gstcacasink.h:
Fix compilation on newer libcaca that require us to include a new
header. Fixes#379918.
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo@circular-chaos.org>
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
(gst_wavpack_dec_init), (gst_wavpack_dec_change_state):
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_base_init),
(gst_wavpack_enc_class_init), (gst_wavpack_enc_reset),
(gst_wavpack_enc_init), (gst_wavpack_enc_set_wp_config),
(gst_wavpack_enc_change_state):
* ext/wavpack/gstwavpackparse.c:
Some small clean-ups: use enums instead of hard-coded numbers,
const-ify element details, re-factor some code into _reset()
functions (#352605).
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_sink_event), (gst_lame_chain),
(gst_lame_change_state):
* ext/lame/gstlame.h:
Make lame timestamp flushed eos buffer by some additional timestamp
accounting. Fixes#374760.
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo at circular-chaos.org>
* configure.ac:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_mode_get_type),
(gst_wavpack_enc_correction_mode_get_type),
(gst_wavpack_enc_joint_stereo_mode_get_type),
(gst_wavpack_enc_init), (gst_wavpack_enc_set_wp_config):
Fix enum nicks; only emit no-more-pads once; add support for very
fast encoding mode in upcoming 4.40.0 release (#369539).
Original commit message from CVS:
* ext/cdio/gstcdio.c: (gst_cdio_get_cdtext):
* ext/cdio/gstcdio.h:
* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open):
Move CD-TEXT utility function into common file so it can also be
used by a future cdioparanoiasrc.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
We require a -base more recent than 0.10.9, so it's safe to use
GST_TYPE_TAG_IMAGE_TYPE unconditionally now.
* ext/dv/gstdvdec.c: (gst_dvdec_sink_event):
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_sink_event):
Use _newsegment_full() now that we depend on a recent enough core.
* gst/wavparse/gstwavparse.c:
Remove cruft that we don't need any longer now that we depend on
a recent enough -base.
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (speex_dec_convert),
(speex_dec_sink_event), (speex_dec_chain_parse_header):
Some small cleanups, use _scale.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_set_wp_config):
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_create_src_pad):
* gst/nuvdemux/gstnuvdemux.c: (gst_nuv_demux_create_pads):
* tests/check/elements/wavpackparse.c: (wavpackparse_found_pad):
Activate pads before adding them to running element.
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_set_property):
Round up not allowed bitrates to the next higher allowed one
(Closes: #361140).
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_sink_setcaps),
(gst_lame_set_property), (gst_lame_get_property), (gst_lame_chain),
(plugin_init):
* po/POTFILES.in:
Add i18n magic to lame plugin. Throw decent error message when we
fail to setup the encoder (#361140, 361151); misc. minor clean-ups.
Original commit message from CVS:
* ext/speex/gstspeexenc.c: (gst_speexenc_finalize),
(gst_speexenc_set_last_msg), (gst_speexenc_setup),
(gst_speexenc_set_header_on_caps):
Fix some mem leaks.
Original commit message from CVS:
* ext/jpeg/gstjpegdec.c:
Set minimum height to 8 (from 16), our code should handle
that fine. Some of the buttons on the apple trailer site
are apparently only 15 pixels high (see #357470).
Original commit message from CVS:
Patch by: Alessandro Decina <alessandro at nnva dot org>
* ext/annodex/gstcmmldec.c: (gst_cmml_dec_chain):
Seeking back in a file containing a CMML stream errors out if the seek
goes back up to the CMML headers. This is because after the seek the xml
processing instruction <?xml ...?> is submitted to the xml parser again,
which results in an error. The attached patch fixes the problem.
Fixes#353908.
* ext/annodex/gstcmmlenc.h:
Fix authors name.
Original commit message from CVS:
* ext/lame/Makefile.am:
* ext/mpeg2dec/Makefile.am:
* gst/dvdlpcmdec/Makefile.am:
* gst/dvdsub/Makefile.am:
* gst/mpegaudioparse/Makefile.am:
Add missing GST_LIBS to the link flags
Original commit message from CVS:
2006-08-28 Andy Wingo <wingo@pobox.com>
* ext/raw1394/gstdv1394src.c (gst_dv1394src_from_raw1394handle):
New helper function to lessen the ifdefs.
(GST_INFO_OBJECT):
(gst_dv1394src_iso_receive): Use it.
(gst_dv1394src_create): Also use the control sockets in iec61883
mode.
(gst_dv1394src_start, gst_dv1394src_stop): Always use a separate
handle for AVC operations; fixes#348233.
Original commit message from CVS:
Patch by: Alessandro Decina <alessandro at nnva dot org>
* ext/annodex/gstannodex.c: (gst_annodex_granule_to_time):
Do some extra sanity checks.
Fixes#350340.
* ext/annodex/gstcmmlenc.c: (gst_cmml_enc_change_state),
(gst_cmml_enc_parse_tag_head), (gst_cmml_enc_parse_tag_clip),
(gst_cmml_enc_push_clip), (gst_cmml_enc_push):
Check if clip->start_time is valid before adding the clip to the
track list.
Reset enc->preamble going from PAUSED to READY.
Don't use GST_FLOW_UNEXPECTED for wrong usage of the element, it is
only used for EOS.
Only post an error message if we were the one that created the fatal
GstFlowReturn value.
* ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt),
(gst_cmml_clock_time_to_granule), (gst_cmml_track_list_has_clip):
Parse the seconds field of the npt-sec time format using %llu rather than
%d and check that the value scaled by GST_SECOND doesn't overflow.
Use guint64(s) to represent the keyindex and keyoffset fields of a granulepos.
Lookup a clip's track with clip->track rather than clip->id which
makes no sense.
Identify a clip by its track and start time and not its xml id.
do some more input checking and make sure we don't do undefined shifts.
* tests/check/elements/cmmldec.c: (setup_cmmldec),
(teardown_cmmldec), (check_output_buffer_is_equal), (push_data),
(cmml_tag_message_pop), (check_headers), (push_clip_full),
(push_clip), (push_empty_clip), (check_output_clip),
(GST_START_TEST), (cmmldec_suite):
* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
(teardown_cmmlenc), (check_output_buffer_is_equal), (push_data),
(check_headers), (push_clip), (check_clip_times), (check_clip),
(check_empty_clip), (GST_START_TEST), (cmmlenc_suite):
Added some more checks.
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo at circular-chaos.org>
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_post_tags),
(gst_wavpack_dec_chain):
Post audio codec and average bitrate tags on bus (#344472).
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init),
(gst_wavpack_parse_src_query):
Forward queries in other formats (BYTE format in particular)
upstream; add Sebastian to authors.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_push_block):
Fix mem leak, send newsegment event on correction pad
as well (#352476).
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init):
Restore original author (on Sebastian's request).
* tests/check/Makefile.am:
* tests/check/gst-plugins-bad.supp:
Add (so far empty) suppression file for -bad. Remove
wavpackenc test from VALGRIND_TO_FIX now that the leak
is fixed.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_dec_scan_got_frame),
(gst_flac_dec_write), (gst_flac_dec_loop),
(gst_flac_dec_sink_event), (gst_flac_dec_chain),
(gst_flac_dec_src_query):
* ext/flac/gstflacdec.h:
Make flac-in-ogg work (#352100).
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_comments):
Make metadata extraction actually work.
* ext/speex/gstspeexenc.c: (gst_speexenc_base_init),
(gst_speexenc_init), (gst_speexenc_create_metadata_buffer),
(gst_speexenc_chain):
Fix metadata writing: replace old code which wrote completely
broken tags with libgsttag-based code. Plus miscellaneous
code cleanups (use static pad templates etc.) and a bunch
of leak fixes.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_init),
(gst_wavpack_enc_finalize), (gst_wavpack_enc_sink_set_caps),
(gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_format_samples),
(gst_wavpack_enc_push_block), (gst_wavpack_enc_chain),
(gst_wavpack_enc_rewrite_first_block),
(gst_wavpack_enc_sink_event), (gst_wavpack_enc_change_state),
(gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property):
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
(gst_wavpack_parse_src_query), (gst_wavpack_parse_src_event),
(gst_wavpack_parse_init), (gst_wavpack_parse_get_upstream_length),
(gst_wavpack_parse_loop):
More clean-ups: use shorter variable names to make code easier to
read; prefix structures we define with 'Gst' to make it clearer
where they come from.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_init),
(gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_push_block),
(gst_wavpack_enc_chain), (gst_wavpack_enc_rewrite_first_block),
(gst_wavpack_enc_sink_event):
Fix caps set on buffers and template caps (output is framed)
and make them match (#351663); use GST_WARNING_OBJECT instead of
GST_ELEMENT_WARNING; simplify push_block(); do some small
clean-ups here and there; fix memleak (#351663).
Original commit message from CVS:
* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_get_unit_size):
* gst/videobox/gstvideobox.c: (gst_video_box_get_unit_size):
use g_assert in _get_unit_size
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/inspect/plugin-audiofxgood.xml:
cleanup -unused.txt to make it useful, add previously missing docs
* ext/Makefile.am:
* ext/esd/esdmon.c:
* ext/esd/esdsink.c:
* ext/esd/gstesd.c: (plugin_init):
reflow to get rid of two external symbols
* gst/audiofxgood/audiofx.c: (plugin_init):
re-add
Original commit message from CVS:
Based on patch by: Sebastian Dröge <slomo at circular-chaos.org>
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_sink_event),
(gst_wavpack_parse_get_upstream_length),
(gst_wavpack_parse_find_marker), (gst_wavpack_parse_resync_loop),
(gst_wavpack_parse_loop), (gst_wavpack_parse_resync_adapter):
Fix resyncing in push mode not stopping re-syncing at embedded
zeroes; skip garbage between frames in pull mode as well if
necessary; use gst_pad_query_peer_duration(); push EOS and
NEWSEGMENT event in right direction (#351659).
Original commit message from CVS:
* ext/dv/gstdvdemux.c: (gst_dvdemux_handle_pull_seek),
(gst_dvdemux_loop), (gst_dvdemux_change_state):
* ext/dv/gstdvdemux.h:
When handling seek requests, don't send the newsegment event from the
calling thread. Instead save it so it can be sent from the streaming
thread.
Original commit message from CVS:
* ext/ladspa/gstladspa.c: (gst_ladspa_base_init):
Convert ' ' into '_'. Try to keep as many characters in the padtemplate
names as possible.
Original commit message from CVS:
* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_flush),
(gst_signal_processor_do_pushes):
A push() gives away our refcount so we should not use the buffer on the
pen anymore.
Original commit message from CVS:
* configure.ac:
Require CVS of GStreamer core and -base (for
GST_TAG_EXTENDED_COMMENT and gst_tag_parse_extended_comment()).
* ext/taglib/gstid3v2mux.cc:
Write extended comment tags properly (#348762).
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_comment_frame):
Extract COMM frames into extended comments, which makes it
easier to properly retain the description bit of the tag
and maintain this information when re-tagging (#348762).
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo at circular-chaos.org>
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init):
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_resync_adapter), (gst_wavpack_parse_chain):
In push mode, re-sync to next wavpack header if sync is lost
(#351557). Also use hyphens instead of underscores in
GObject property names.
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_class_init),
(gst_wavpack_parse_reset), (gst_wavpack_parse_get_src_query_types),
(gst_wavpack_parse_src_query),
(gst_wavpack_parse_handle_seek_event),
(gst_wavpack_parse_sink_event), (gst_wavpack_parse_init),
(gst_wavpack_parse_create_src_pad),
(gst_wavpack_parse_push_buffer), (gst_wavpack_parse_loop),
(gst_wavpack_parse_chain), (gst_wavpack_parse_sink_activate),
(gst_wavpack_parse_sink_activate_pull):
* ext/wavpack/gstwavpackparse.h:
Patch by: Sebastian Dröge <slomo at circular-chaos.org>
Make wavpackparse also work in push-mode (not seekable yet though);
some small clean-ups along the way; add support for SEEKING query
and query types function. (#351495).
Original commit message from CVS:
* ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_setcaps),
(gst_smokeenc_resync), (gst_smokeenc_chain):
Refuse sink caps in the encoder if width or height is not a
multiple of 16, the encoder does not support that yet; along the
same lines, check the return value of the encoder setup function;
also remove some debug log clutter.
Original commit message from CVS:
2006-08-04 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstsignalprocessor.h: Add infrastructure for storing
whether a processor can work in place or not, and for keeping
track of its state. Change the FlowReturn instance variable from
"state" to "flow_state", all callers changed.
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setup)
(gst_signal_processor_start, gst_signal_processor_stop)
(gst_signal_processor_cleanup): New functions to manage the
processor's state.
(gst_signal_processor_setcaps): start() as well as setup() here.
(gst_signal_processor_prepare): Respect CAN_PROCESS_IN_PLACE.
(gst_signal_processor_change_state): Stop and cleanup the
processor as we go to NULL.
* ext/ladspa/gstladspa.c (gst_ladspa_base_init): Reuse buffers if
INPLACE_BROKEN is not set.
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_prepare):
Do the alloc_buffer in bytes, not frames.
Original commit message from CVS:
2006-08-04 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setcaps)
(gst_signal_processor_prepare)
(gst_signal_processor_update_inputs)
(gst_signal_processor_process, gst_signal_processor_pen_buffer)
(gst_signal_processor_flush)
(gst_signal_processor_sink_activate_push)
(gst_signal_processor_src_activate_pull)
(gst_signal_processor_change_state): Remove the last of the code
that assumes that we process whole buffers at a time. Fix some
debugging. Seems to work now in some cases.
Original commit message from CVS:
2006-08-01 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_process):
Fix nframes-choosing.
(gst_signal_processor_init): Init pending_in and pending_out.
Original commit message from CVS:
2006-08-01 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): No
more default sample rate, although we never check that the sample
rate actually gets set. Something for the future.
(gst_signal_processor_setcaps): Some refcount fixes, flow fixes.
(gst_signal_processor_event): Refcount fixen.
(gst_signal_processor_process): Pull the number of frames to
process from the sizes of the buffers in the input pens.
(gst_signal_processor_pen_buffer): Remove an incorrect FIXME :)
(gst_signal_processor_do_pulls): Add an nframes argument, and use
it instead of buffer_frames.
(gst_signal_processor_getrange): Refcount fixen, pass nframes on
to do_pulls.
(gst_signal_processor_chain)
(gst_signal_processor_sink_activate_push)
(gst_signal_processor_src_activate_pull): Refcount fixen.
* ext/ladspa/gstsignalprocessor.h: No more buffer_frames, yay.
Original commit message from CVS:
* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setcaps),
(gst_signal_processor_process):
don't query buffer-frames from caps, add lots of debug-log,
try fix for assert (#349189)
Original commit message from CVS:
2006-07-29 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps),
(gst_smokeenc_setcaps), (gst_smokeenc_chain):
Set caps on buffer correctly. Fixes bug #349155.
Original commit message from CVS:
2006-07-26 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* ext/lame/gstlame.c: (gst_lame_setup):
Fix lame putting lots of 0's at start of mp3. Fixes bug #348786.
Original commit message from CVS:
* ext/taglib/gstid3v2mux.cc:
Fix writing of comment frames (should be COMM not TCOM),
is still sub-optimal though, since we don't retain or
extract the comment descriptions properly (#334375,
also see #334375).
Original commit message from CVS:
* ext/taglib/gstid3v2mux.cc:
Handle multiple tags of the same type properly. Re-inject
unparsed ID3v2 frames that we get as binary blobs from
id3demux into the tag again so we don't lose information
when retagging (#334375).
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_open),
(gst_esdsink_factory_init):
Prevent libesd from auto-spawning a sound daemon if it
is not already running. Now that we don't do evil stuff
like that any longer we can give esdsink a rank so that
autoaudiosink will try it as well if all other audio
sinks fail (#343051).
Original commit message from CVS:
* ext/esd/README:
Remove, it contains nothing useful anyway.
* ext/esd/esdsink.c: (gst_esdsink_init), (gst_esdsink_prepare),
(gst_esdsink_delay):
Some small clean-ups; use GST_BOILERPLATE etc.
Original commit message from CVS:
Patch by: Alex Lancaster <alexl at users sourceforge net>
* ext/taglib/gstid3v2mux.cc:
Write GST_TAG_ENCODER and GST_TAG_ENCODER_VERSION as
ID3v2 TSSE frames (#347898).
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Fix caps after previous change to byte order endianness.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
(gst_wavpack_parse_sink_event), (gst_wavpack_parse_init),
(gst_wavpack_parse_loop):
* ext/wavpack/gstwavpackparse.h:
Queue incoming events if there's no source pad yet and
send them downstream later when the pad is there.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init),
(gst_wavpack_dec_format_samples),
(gst_wavpack_dec_clip_outgoing_buffer), (gst_wavpack_dec_chain),
(gst_wavpack_dec_change_state):
* ext/wavpack/gstwavpackdec.h:
Output audio in native byte order (which is also how we get
samples from wavpack); output samples with 21-24 bit depth
with 32 bit width (makes things easier for us).
