Many AVFifoBuffer users operate on fixed-size elements (e.g. pointers),
but the current FIFO API deals exclusively in bytes, requiring extra
complexity in all these callers.
Add a new AVFifo API creating a FIFO with an element size
that may be larger than a byte. All operations on such a FIFO then
operate on complete elements.
This API does not reuse AVFifoBuffer and its API at all, but instead uses
an opaque struct called AVFifo. The AVFifoBuffer API will be deprecated
in a future commit once all of its users have been switched to the new
API.
Not reusing AVFifoBuffer also allowed to use the full range of size_t
from the beginning.
The API currently allows creating FIFOs up to
- UINT_MAX: av_fifo_alloc(), av_fifo_realloc(), av_fifo_grow()
- SIZE_MAX: av_fifo_alloc_array()
However the usable limit is determined by
- rndx/wndx being uint32_t
- av_fifo_[size,space] returning int
so no FIFO should be larger than the smallest of
- INT_MAX
- UINT32_MAX
- SIZE_MAX
(which should be INT_MAX an all commonly used platforms).
Return an error on trying to allocate FIFOs larger than this limit.
Avoids code duplication. It furthermore properly checks
for buf_size to be > 0 before doing anything.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This reduces sibilance distortion when sibilance and bass are
present at the same time. Bringing the protection of high
frequencies up to about the same level as for low frequencies
should also make the quality less dependent on the frequency
balance of the playback system.
Signed-off-by: Jason Jang <jcj83429@gmail.com>
Ignore more samples that are near the edge of the block. The reason
is that the filtering tends to cause these samples to go above the
window more than the samples near the middle. If these samples are
included in the unwindowed peak estimation, the peak can be
overestimated. Because the block is windowed again before
overlapping, overshoots near the edge of the block are not very
important.
0.1 is the value from the version originally contributed to calf.
Signed-off-by: Jason Jang <jcj83429@gmail.com>
With a complex FFT instead of real FFT, the negative frequencies
are not dropped from the spectrum output, so they need to be scaled
when the positive frequencies are scaled. The location of the top
bin is also different.
Signed-off-by: Jason Jang <jcj83429@gmail.com>
In previous state, a new frame was allocated on each timestamp step,
i.e. each frame/field transition. However, for interlace, a new frame
should be allocated on 1st field, completed with the 2nd and finally
freed.
This commit fixes the frame allocation and the detection of missing RTP
markers.
Signed-off-by: Patrick Keroulas <patrick.keroulas@radio-canada.ca>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
The fdk-aac decoder can return decoded audio data with a delay.
(Whether it does this or not depends on the options set; by default
it does add some delay.) Previously, this delay was handled by
adjusting the timestamps of the decoded frames, but the last delayed
samples weren't returned.
Set the AV_CODEC_CAP_DELAY flag to indicate that the caller should
flush remaining samples at the end. Also trim off the corresponding
amount of samples at the start instead of adjusting timestamps.
Signed-off-by: Martin Storsjö <martin@martin.st>
At present, side data printing forces display for all levels i.e.
stream, packets and frames. This can bloat output and also force
decode of all frames in selected streams.
Now, stream_side_data[=type], packet_side_data[=type] &
frame_side_data[=type] can be used with -show_entries to specify carrier
element.
Anton Khirnov
Andreas Rheinhardt
Zhao Zhili
Paul B Mahol
Zane van Iperen
Jason Jang
Pierre-Anthony Lemieux
James Almer
Marton Balint
Patrick Keroulas
Peter Ross
Martin Storsjö
Gyan Doshi
TianBo Zheng
Lynne