Make the QtDemuxSample struct smaller by keeping the duration and the pts_offset
as their 32 bit values.
Make some macros to calculate PTS, DTS and duration of a sample.
Deref the sample index less often by keeping a ref to the sample we're dealing
with.
Move sample position checks into qtdemux_parse_samples where we can protect it
with a lock.
Refactor and make an qtdemux_ensure_index function.
Rename qtdemux_do_push_seek to qtdemux_seek_offset in order to avoid confusion
with gst_qtdemux_do_push_seek.
When nsamples_out is larger than nsamples_in, using unsigned
ints lead to a overflow and the resulting value is wrong and
way too large for allocating a buffer. Use signed integers
and returning immediatelly when that happens.
Use more efficient formula that uses less multiplies.
Reduce the amount of scalar code, use MMX to calculate the desired
alpha value.
Unroll and handle 2 pixels in one iteration for improved pairing.
Convert the alpha value to 0->255 when setting and to 0->256 when using as
a scaling factor. This makes sure we can reach the full opacity value of 0xff in
all cases.
Fix some comments, clarify some FIXMEs
Remove the on-ntp-stop signal check now that the jitterbuffer is in
-good and we know that it supports this signal.
Use GstRTPBaseAudioPayload as the base class. This saves a lot of code and fixes
a bunch of problems that were already solved in the base class.
Fixes#853367
Don't make copied in the getter and setter for SDES in the RTPSource. This
avoids a couple of copies of the SDES structure when generating RTCP
packets.
Add a new spspps-interval property to instruct the payloader to insert
SPS and PPS at periodic intervals in the stream.
Rework the SPS/PPS handling so that bytestream and AVC sample code both use the
same code paths to handle sprop-parameter-sets. This also allows to have the AVC
code to insert SPS/PPS like the bytestream code.
Fixes#604913
For some reason latest gcc/binutils accept movzxb here while
movzbl would be correct and is the only thing accepted by older
gcc/binutils.
Fixes bug #604679.
This provides another 7% speedup for the time domain convolution and 1.5%
speedup for the FFT convolution on Mono input.
This optimization assumes that the compiler simplifies calculations
and conditions on constant numbers and unrolls loops with a constant
number of repeats.
This will always use time-domain convolution, which lowers the latency.
With FFT convolution it's always a multiple of the kernel length,
with time domain convolution it's only the pre-latency of the filter kernel.
This provides a great speedup, especially the relationship between kernel
length and processing size is now logarithmic instead of linear. Below a
kernel size of 32 it's a bit slower, afterwards it's much faster:
17 0.788000 -> 0.950000
33 1.208000 -> 1.146000
65 2.166000 -> 1.146000
...
4097 107.444000 -> 1.508000
For sizes smaller 32 the normal time-domain convolution is chosen,
for larger sizes the FFT convolution is automatically used.
Fixes bug #594381.
Remove some redundant calculations, move comparisions out of
inner loops, etc.
This makes the convolution about 3 (!) times faster but
processing time is of course still proportional to the
filter size.
Quicktime uses ISO 639-2 for language codes, but GST_TAG_LANGUAGE
is supposed to hold a ISO 639-1 code, so convert as needed using
the new API from -base.
See #602126.
Matroska uses three-letter ISO 639-2B codes, but GST_TAG_LANGUAGE is
supposed to contain two-letter ISO 639-1 codes, so use new language
code mapping functions in -base to convert between those two as
needed.
Fixes#505823.
When an RTSP extension returns NULL or an empty transport string, just ignore it
and try to get the next possible transport. Fixes playback of RealMedia streams.
Set the current_entry to 0 (instead of -1) in push mode so that we correctly
calculate the current frame number and timestamp.
Add some more debug info and fic the duration debug.
Send pending tags only from the streaming thread, just after we've sent
the newsegment event, not with e.g. flush-start. This not only does the
right thing, but also makes sure we're not trampling over variables set
up in the streaming thread from the seeking thread in case someone tries
to issue a seek just as the demuxer is parsing the headers.
Fixes#601617. Spotted by Ognyan Tonchev.
We ref the buffer before pushing it downstream in order to get the CSRCs of it
after pushing. This causes performance problems when downstream elements want to
change the metadata because the buffer needs to be subbuffered.
Instead, read and store the CSRCs of the buffer in an array before pushing it
and process the array after pushing the buffer. This allows us to remove the
ref/unref pair.
Fixes#603376
We were only setting the suggested buf size for video,
we can set it for audio as well.
This and 195e14529d80ef318ce3a778c1995efb11f266cd
fix an issue that prevented seeking on large avi files
on WMP (non-recent versions).
