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Commit Graph

6017 Commits

Author SHA1 Message Date
Marton Balint
a4fc331118 avutil/channel_layout: add specific text versions for unknown and unused channels
Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-16 19:49:39 +01:00
Marton Balint
95d31db82c avutil/channel_layout: factorize parsing list of channel names
Also make use of the av_channel_from_string() function to determine the channel
id. This fixes some parse issues in av_channel_layout_from_string().

Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-16 19:49:39 +01:00
Marton Balint
0b3b8a1918 avutil/tests/channel_layout: add some av_channel_from_string and av_channel_layout_from_string tests
We lacked tests which supposed to fail, and there are some which should fail
but right now it does not. This will be fixed in a later commit.

Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-16 19:49:39 +01:00
Marton Balint
b2b22c2d1a avutil/tests/channel_layout: make printing results part of the tests
Deduplicates a lot of code.

Some minor differences (mostly white space and inconsistent use of quotes) are
expected in the fate tests, there was no point aiming for exactly the same
formatting.

Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-16 19:49:39 +01:00
Marton Balint
44b2769619 avformat/pcm: decrease target audio frame per sec to 10
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>
2024-03-16 19:19:42 +01:00
Marton Balint
05936403f9 avformat/wavdec: use ff_pcm_default_packet_size for the default packet size
Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-16 19:19:42 +01:00
Marton Balint
9c2c0c37f8 avformat/pcm: factorize and improve determining the default packet size
- 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>
2024-03-16 19:19:42 +01:00
James Almer
cfa694d811 fate/wmavoice: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
59f5cf5c71 fate/vqf: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
e48b221144 fate/voice: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
9906bef5c4 fate/vorbis: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
85da6e5c44 fate/real: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
2262c9ab0c fate/pcm: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
540d1b14d8 fate/mpc: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
2df103528c fate/mp3: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
6d569aa80c fate/mov: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
b80b3947dd fate/monkeysaudio: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
be9d9b7aba fate/lossless-audio: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
8b96aca432 fate/libswresample: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
47362785ae fate/iamf: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
92c7e27373 fate/hlsenc: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
6887a0292f fate/gapless: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
f3f2932f75 fate/ffprobe: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:44:59 -03:00
James Almer
536dfe92e0 fate/ffmpeg: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:37:50 -03:00
James Almer
4c8b8bc52b fate/filter-audio: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:37:42 -03:00
James Almer
20581cea3e fate/fate-container: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:05:38 -03:00
James Almer
7416d216aa fate/demux: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:05:35 -03:00
James Almer
04ab5cc584 fate/audio: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:05:27 -03:00
James Almer
fc17af7f8d fate/atrac: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:05:22 -03:00
James Almer
8755e7eb74 fate/amrwb: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:05:18 -03:00
James Almer
7a10d6521f fate/amrnb: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:05:14 -03:00
James Almer
b3ab87d320 fate/alac: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:04:31 -03:00
James Almer
234268b4ed fate/adpcm: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:03:36 -03:00
James Almer
73095bc19e fate/ac3: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:03:36 -03:00
James Almer
136f1cdf0f fate/aac: add missing aresample filter dependency
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 12:03:36 -03:00
Andreas Rheinhardt
c00cd007e8 configure: Remove av_restrict
All versions of MSVC that support C11 (namely >= v19.27)
also support the restrict keyword, therefore av_restrict
is no longer necessary since 75697836b1.

Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-15 12:51:15 +01:00
James Almer
a327434df7 fate/ffmpeg: add missing idct decoder option to fate-ffmpeg-loopback-decoding
Should fix failures on x86_32 targets.

Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-14 16:44:12 -03:00
James Almer
ad6347fc37 fate/ffmpeg: add a -threads input option to the loopback decoder
Honor the requested value passed when calling make fate.

Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-14 10:48:09 -03:00
James Almer
d925b2e139 fate/ffmpeg: add a test for loopback decoding
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-14 10:00:03 -03:00
Martin Storsjö
8ff4a4a4f4 checkasm: hevc_pel: Use checkasm_check for printing failing output
This simplifies the code for checking the output, and can print
the failing output (including a map of matching/mismatching
elements) if checkasm is run with the -v/--verbose option.

Signed-off-by: J. Dekker <jdek@itanimul.li>
2024-03-14 13:42:39 +01:00
Martin Storsjö
3ad3ada11f checkasm: hevc_pel: Split a couple excessively long lines
Signed-off-by: J. Dekker <jdek@itanimul.li>
2024-03-14 13:42:39 +01:00
Martin Storsjö
64a2cdca13 checkasm: hevc_pel: Check the full output in hevc_epel/hevc_qpel
Previously it only checked half the output in 8 bit per pixel mode,
as the output actually is 16 bit elements here.

Signed-off-by: J. Dekker <jdek@itanimul.li>
2024-03-14 13:42:39 +01:00
Marton Balint
2129d66a66 fate: use atrim filter instead of -frames:a 20 for fate-filter-tremolo
To make it independent of incoming wav demuxer packet size.

Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-14 01:37:31 +01:00
Marton Balint
6fc6cac4c6 fate: use a fixed wav demux packet size for amix tests
The dropout transition feature of the amix filter depends on the incoming
packet size.

Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-14 01:37:31 +01:00
Marton Balint
8c8ce4f233 fate: make filter-channelsplit test use a fixed frame size
Muxing multiple streams to raw files is allowed but the packets are
interleaved, so the output is dependant of packet size.

Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-14 01:37:31 +01:00
Marton Balint
7196b12b2b avformat/daudenc: force 2000 sample packet size with a bsf
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>
2024-03-14 01:37:31 +01:00
Kieran Kunhya
110d8549d5 avcodec/vvcdec: Mark as experimental
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-13 20:46:10 -03:00
James Almer
394abd8458 fftools/ffprobe: export IAMF Stream Group parameters
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-13 16:45:15 -03:00
James Almer
5cd8db3060 fftools/ffprobe: export Tile Grid Stream Group parameters
Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-13 16:06:10 -03:00
Anton Khirnov
daca5d1241 fftools/ffmpeg_filter: refactor setting input timebase
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.
2024-03-13 08:01:15 +01:00