| Commit message (Collapse) | Author | Age | Files | Lines |
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Detected by LLVM's CLang static analyzer
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Don't crash when the timing info is not yet available.
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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.
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When the server is disconnected or when we are shut down, make our clock return
an invalid time instead of erroring out.
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Keep bps as gint instead of guint because we will be doing signed math with it
later on and we don't want weird results.
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Don't try to change the stream volume (and other things) when we don't have a
stream yet. Just store the values for later.
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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.
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Start filling up the buffer with empty samples when an underflow happens. We
need to do this to keep pulseaudio reporting the right time for us.
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Use the default basesink methods for implementing pull based scheduling, it
works fine for us.
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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.
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Derive from BaseAudioSink and implement our custom ringbuffer that maps to the
internal pulseaudio ringbuffer.
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Should work at least a tad better if the decoder can't keep up, and
should also spread dropped frames a bit more evenly over time.
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sending the samples
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And log decoder object.
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Don't decode frames that are going to be too late anyway.
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separately
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Make the state change function a bit more readable and only pause after the
parent had a change to pause first.
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Use an instance variable to track whether the stream is corked or not,
instead of using PA API that was only introduced in 0.9.11
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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.
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Don't send queries back to the element they just came from by sending
them to the peer of the wrong pad.
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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)
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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.
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Make some code more readable.
Fix a leak when pull range returns a shot buffer.
Push EOS after posting the error.
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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.
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ssh://thomasvs@git.freedesktop.org/git/gstreamer/gst-plugins-good
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PAL is TFF, NTSC is BFF. Some day I will learn to keep this
straight.
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Don't crash if we receive a buffer without caps. Fixes #572413.
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Remove some unused variables.
Avoid a useless _resync call.
Correctly use a gboolean.
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Fixes bug #571321.
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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.
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Cast (enum *) to (int *), not necessarily technically right,
but plugs #571153.
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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).
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If our read callback ran out of data, so had to abort reading, we return
GST_FLOW_ERROR instead of going into an infinite loop.
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