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.
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.
