Fixes stuttering audio when iOS AU is resampling. To make AU resample,
one has to request a rate that differs from AVAudioSession's
sampleRate. The resampling itself is not the culprit, but rather our
API misuse.
AudioUnitRender modifies the mDataByteSize members with the
actual read bytes count. Therefore, they must be reinitialized
before each AudioUnitRender. (The buffers themselves can be
preallocated.)
The "stutter" was caused by one AudioUnitRender making the buffer
too small for other AudioUnitRender invocations, making them fail
with -50 (paramErr). By way of luck, when AU didn't resample, all
AudioUnitRender invocations read the same number of bytes.
(This patch addresses some non-interleaved audio concerns, but
at this moment the elements do not support non-interleaved audio
and non-interleaved is untested.)
https://bugzilla.gnome.org/show_bug.cgi?id=744922
When there is no allocation parameters in the query, enable copy
threshold. When this threshold is reached, the buffer pool will start
copying when the pool reaches a critical level. If the driver supports
CREATE_BUFS, this will be used instead.
When we hit emulated formats, we disable CREATE_BUFS since libv4l2
cope very badly with it. Also clear the allocator flags so we will
never try to allocate more buffers. This fixes failure when the copy
threshold is reached as we where calling CREATE_BUFS, which lead to
libv4l2 instability.
In the fraction 1 / 2. 1 is the numerator and 2 is the denominator.
The arguments of fraction gst_value_set_fractions() are value,
numerator and denominator.
Also, gst_value_set_fraction() fails if denominator is 0 for obvious
reasons.
When importing buffers from a downstream pool, we need to deactivate
that pool to ensure it will be usable again later. Relying on the
refcount to reach zero does not work, since elements like xvimagesink
keeps a reference on their proposed pool.
We need different symbol names, because these symbols are also present
in the fragmented plugin ... which will cause conflicts when doing
static linking
The test had a function to print the error, but was not parsing it.
This was causing warning about dbg_info being used uninitialized. If
the test was testing any errors, this would have crashed.
The number of FFTs is calculated with the following formula:
guint nfft = 2 * bands - 2;
nfft is passed to gst_fft_f32_new() as the len argument and is of type
unsigned integer. This method required that len is at leas 1, then
maximum G_MAXINT, as other values would be negative. If we extrapolate
from the formula above it means we need "bands" to be between 2 and
((guint)G_MAXINT + 2) / 2).
https://bugzilla.gnome.org/show_bug.cgi?id=744213
When memory (that has been shared using gst_memory_share()) are freed,
the memory (or the DMABUF FD) should not bee freed. These memories have
a parent. This also removes the extra _v4l2mem_free function and avoid
calling close twice on the DMABUF FD.
https://bugzilla.gnome.org/show_bug.cgi?id=744573
Using the sparse streams can make the push-based seeking return
too far in the stream. It also can lead to issues as the
sparse streams will be ignored when restarting playback and,
if the sparse stream is the one that has the earliest sample,
it will confuse qtdemux's offsets as one stream will have
an earlier offset than the demuxer's one which might lead to
early EOS.
https://bugzilla.gnome.org/show_bug.cgi?id=742661
In pulsesink_query function, we use a switch for the query
type. In the CAPS case, there is no 'break', instead we
return right away. Use a break and return at the end of
the function instead for better code readability.
https://bugzilla.gnome.org/show_bug.cgi?id=744461
Parse the 'sidx' atom and update the total duration according to the
parser result. The isoff parser code is imported from
gst-plugins-bad's dashdemux and a gst_isoff_sidx_parser_add_data()
function was factored out of the gst_isoff_sidx_parser_add_buffer()
function.
https://bugzilla.gnome.org/show_bug.cgi?id=743578
According to RFC 4585 section 3.5.3 step 1 we are not allowed to send
an early RTCP packet for the very first one. It must be a regular one.
Also make sure to not use last_rtcp_send_time in any calculations until
we actually sent an RTCP packet already. In specific this means that we
must not use it for forward reconsideration of the current RTCP send time.
Instead we don't do any forward reconsideration for the first RTCP packet.
Assignment is done to variable segment.stop when the intention was to assign to
local variable stop. Instead of overwriting it, the value is now clamped and
segment.stop is set to it soon after.
CID #1265773
Handle the case where a short file reaches EOS while we're still
waiting for no-more-pads, and make sure we continue to the internal
READY state for real playback to work properly later.
Implement 2 new elements - splitmuxsink and splitmuxsrc.
splitmuxsink is a bin which wraps a muxer and takes 1 video stream,
plus audio/subtitle streams, and starts a new file
whenever necessary to avoid overrunning a threshold of either bytes
or time. New files are started at a keyframe, and corresponding audio
and subtitle streams are split at packet boundaries to match
video GOP timestamps.
splitmuxsrc is a corresponding source element which handles
the splitmux:// URL and plays back all component files,
reconstructing the original elementary streams as it goes.
Our ones were expired. The new ones were copied from libsoup's
tests files.
Also sets the property to use our own cert to validate the
server, otherwise the default system certs would be used
and it would fail.
We detect a container correctly now so we need to revert the weird
check there was before.
Use gst_rtspsrc_stream_push_event() to push the caps event on the
right pad.
See https://bugzilla.gnome.org/show_bug.cgi?id=739391
Keep global and stream tags separately and parse the udta node
that can be found under the trak atom. The udta will contain
stream specific tags and will be pushed as such
https://bugzilla.gnome.org/show_bug.cgi?id=692473
