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9 Commits

Author SHA1 Message Date
Philipp Hancke
c1dc8abeeb Remove non-span NAL unit splitter and SPS parser 
after cleaning up the Chromium dependency

BUG=webrtc:42225170

Change-Id: Icd3934ca51f829c55e061fc1943500435c845a8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362569
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43053}
 2024-09-19 17:49:37 +00:00
Sergio Garcia Murillo
45e5e385f3 Use ArrayView on H264 bitstream parsing 
No-Try: true
Bug: webrtc:42225170
Change-Id: I4682f400054fee5c86ea24bebf6d703fb90074da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354722
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42650}
 2024-07-19 08:16:11 +00:00
Danil Chapovalov
cd8b36bb9f Fix integer underflow in H264 bitstream parser 
num_ref are represented using golomb and may be very large.
BitstreamReader is generally resilent to many consenquites fail reads, but not when number of reads comparable to int limit.
This change address the issue in two ways, either one is enough, but both are helpful in their own way:

H264 parser now fails faster when number of references is too large.
BitstreamReader now is resilent to unlimited number of fail reads.

Bug: chromium:352402499
Change-Id: I19646bc3f53cd2970393d00bc143400b1fdf5473
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357100
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42628}
 2024-07-12 15:01:55 +00:00
Qiu Jianlin
7121680f38 Expose WebRTC H.264 SPS parser APIs. 
Blink RTC video encoder will need this simplified parser to replace
current full H.264 SPS parser, to reduce parsing cost for extracting
frame size from compressed H.264 bitstream.

Bug: b:309132190
Change-Id: I6863f10bd139766d47fe4bff89143c1a91a09287
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332468
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#41466}
 2024-01-03 10:48:51 +00:00
Qiu Jianlin
67468c5b92 Ignore prefix NAL unit for slice parsing. 
For temporal scalability, prefix NALU contains layer information, but
should not be parsed as a slice.

Bug: webrtc:12991
Change-Id: Ic1e7d41f568310390a743d4ace016aa7d57a4864
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226501
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34523}
 2021-07-22 06:25:53 +00:00
Jonas Olsson
a4d873786f Format almost everything. 
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
 2019-07-08 13:45:15 +00:00
Yves Gerey
988cc0870b [Cleanup] Add missing #include. Remove useless ones. 
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
 2018-10-23 11:32:56 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/ 
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
 2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/. 
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
 2017-09-15 04:25:06 +00:00