This makes the wav and pcm demuxer demux bigger packets, which is more
efficient.
As a side effect of the bigger packets, audio durations can become less exact
for command lines such as "ffmpeg -i $INPUT -c:a copy -t 1.0 $OUTPUT".
Signed-off-by: Marton Balint <cus@passwd.hu>
- Remove the 1024 cap on the number of samples, for high sample rate audio it
was suboptimal, calculate the low neighbour power of two for the number of
samples (audio blocks) instead.
- Make the function work correctly also for non-pcm codecs by using the stream
bitrate to estimate the target packet size. A previous version of this patch
used av_get_audio_frame_duration2() the estimate the desired packet size, but
for some codecs that returns the duration of a single audio frame regardless
of frame_bytes.
- Fallback to 4096/block_align*block_align if bitrate is not available.
Signed-off-by: Marton Balint <cus@passwd.hu>
The samples I found all have 2000 sample packets, and by forcing the packet
size with a bsf we could automagically make muxing work for packets containing
more than 3640 samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
Treat it analogously to stream parameters like format/dimensions/etc.
This is functionally different from previous code in 2 ways:
* for non-CFR video, the frame timebase (set by the decoder) is used
rather than the demuxer timebase
* for sub2video, AV_TIME_BASE_Q is used, which is hardcoded by the
subtitle decoding API
These changes should avoid unnecessary and potentially lossy timestamp
conversions from decoder timebase into the demuxer one.
Changes the timebases used in sub2video tests.
Some encoders, like flac, propagate updated extradata at the end of encoding
as packet side data. Use it to update the relevant codec_config.
Signed-off-by: James Almer <jamrial@gmail.com>
The wav demuxer by default tried to demux 4096-byte packets which caused
packets with very few number of samples for files with high channel count.
This caused a significant overhead especially since the latest ffmpeg.c
threading changes.
So let's use a similar approach for selecting audio frame size which is already
used in the PCM demuxer, which is to read 25 times per second but at most 1024
samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
Jpeg2000 decoder is decoding in native endian, so let's use the same workaround
as in fate-mxf-probe-applehdr10.
Fixes ticket #10868.
Signed-off-by: Marton Balint <cus@passwd.hu>
The old layout happened to be a native layout and therefore missed some
recently fixed layout parsing bugs.
Signed-off-by: Marton Balint <cus@passwd.hu>
If a custom layout is equivalent to a native one, check if it matches one of the
known layout names and print that instead.
Signed-off-by: James Almer <jamrial@gmail.com>
This together with adjusting the inclusion define allows for the
build to not fail with latest Vulkan-Headers that contain the
stabilized Vulkan AV1 decoding definitions.
Compilation fails currently as the AV1 header is getting included
via hwcontext_vulkan.h -> <vulkan/vulkan.h> -> vulkan_core.h, which
finally includes vk_video/vulkan_video_codec_av1std.h and the decode
header, leading to the bundled header to never defining anything
due to the inclusion define being the same.
This fix is imperfect, as it leads to additional re-definition
warnings for things such as
VK_STD_VULKAN_VIDEO_CODEC_AV1_DECODE_SPEC_VERSION. , but it is
not clear how to otherwise have the bundled version trump the
actually standardized one for a short-term compilation fix.
Currently, this only affects untagged RGB/XYZ/Gray, which get forced to
their corresponding metadata before entering the filter graph. The main
justification for this change, however, is the planned ability to add
automatic promotion of unspecified yuv to mpeg range yuv.
Notably, this change will never allow accidentally cross-promoting
unspecified to jpeg or to a specific YUV matrix, since that is still
bound by the constraints of YUV range negotiation as set up by
query_formats.
Previously, we produced output with either \r\n or mixed line endings.
This was undesirable unto itself, but also made working with patches affecting
FATE output particularly challenging, especially via the mailing list.
Everything that consumes the SSA/ASS format is line-ending-agnostic,
so \n is selected to simplify git/ML usage in FATE.
Extra \r characters at the end of a packet are dropped. These are always
ignored by the renderer anyway.
The previous assumption that DXV needs to be aligned to 16x16 was
erroneous. 4x4 works just as well, and FATE decoder tests pass for all
texture formats.
On the encoder side, we should reject input that isn't 4x4 aligned,
like the HAP encoder does, and stop aligning to 16x16. This both solves
the uninitialized reads causing current FATE tests to fail and produces
smaller encoded outputs.
With regard to correctness, I've checked the decoding path by encoding a
real-world sample with git master, and decoding it with
ffmpeg -i dxt1-master.mov -c:v rawvideo -f framecrc -
The results are exactly the same between master and this patch.
On the encoding side, I've encoded a real-world sample with both master
and this patch, and decoded both versions with
ffmpeg -i dxt1-{master,patch}.mov -c:v rawvideo -f framecrc -
Under this patch, results for both inputs are exactly the same.
In other words, the extra padding gained by 16x16 alignment over 4x4
alignment has no impact on decoded video.
Signed-off-by: Connor Worley <connorbworley@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes all ff_draw_* based filters aware of YUV colorspaces and
ranges. Needed for YUVJ removal. Also fixes a bug where e.g. vf_pad
would generate a limited range background even after conversion to
full-scale grayscale.
The FATE changes were a consequence of the aforementioned bugfix - the
gray scale files are output as full range (due to conversion by
libswscale, which hard-codes gray = full), and appropriately tagged as
such, but before this change the padded version incorrectly used
a limited range (16) black background for these formats.
Use 8 packets/frames by default rather than 1, which seems to provide
better throughput.
Allow -thread_queue_size to set the muxer queue size manually again.
Finishes fixing vp5/potter512-400-partial.avi
The fate-matroska-ms-mode test ref is updated to reflect that the Speex decoder
can now read the stream.
Signed-off-by: James Almer <jamrial@gmail.com>
Covers muxing from raw pcm audio input into FLAC, using several scalable layouts,
and demuxing the result.
Signed-off-by: James Almer <jamrial@gmail.com>
This is the 64bit version of Chris Doty-Humphreys SFC64
Compared to the LCGs these produce much better quality numbers.
Compared to LFGs this needs less state. (our LFG has 224 byte
state for its 32bit version) this has 32byte state
Also the initialization for our LFG is slower.
This is also much faster than KISS or PCG.
This commit replaces the broken LCG used before.
(broken as it had only a period ~200M due to being put in a double)
This changes the output from random() which is why libswresample.mak
is updated, update was done using the command in libswresample.mak
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Parse iprp and iinf boxes and its child boxes to get the actual codec used
(AV1 for avif, HEVC for heic), and properly export extradata and other
properties in a generic way.
The avif tests reference files are updated as the extradata is now exported.
Based on a patch by Swaraj Hota
Co-authored-by: Swaraj Hota <swarajhota353@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Marton Balint