Commit graph

30 commits

Author SHA1 Message Date
Florent Castelli
8037fc6ffa Migrate absl::optional to std::optional
Bug: webrtc:342905193
No-Try: True
Change-Id: Icc968be43b8830038ea9a1f5f604307220457807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42911}
2024-09-02 12:16:47 +00:00
Sergio Garcia Murillo
469e69800f Remove kMaxNalusPerPacket hard limit for H264 frames
Bug: webrtc:346608838
Change-Id: I067401250994bc57897edff8e8a18c3088d96b08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354622
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42487}
2024-06-14 16:29:42 +00:00
Stephan Hartmann
fa4d7c92b7 libstdc++: replace deprecated std::is_pod<T>
std::is_pod is deprecated since C++20. Replace with std::trivial and
std::is_standard_layout. Avoids a lot of warnings.

Bug: chromium:957519
Change-Id: Idb4bde7401c14c0896a84c357ec668b9916f613e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325484
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41117}
2023-11-09 15:16:00 +00:00
qwu16
ae82df718c Add codec name H265 to support H265 in WebRTC
Bug: webrtc:13485
Change-Id: I352b15a65867f0d56fc8e9a9e03081bd3258108e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316283
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40773}
2023-09-20 09:25:32 +00:00
Tony Herre
be9b576188 Move video video receiver transformable frame to modules/rtc_rtcp/source
Step 1 of combining the sender and receiver types

Also moved the RtpFrameObject to rtp_rtcp/source, as it's heavily used
by the transformable receiver frame, I couldn't work out a better way
of managing the dependencies, and everything else seemed to work fine.

Bug: chromium:1412687
Change-Id: I55e816a0d7aa2962560ff9ebaf30ad63ab0b9810
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291710
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39255}
2023-02-03 12:59:19 +00:00
philipel
2182096e66 RtpFrameReferenceFinder return frames directly instead of via callback.
Bug: webrtc:12579
Change-Id: I41263f70a6f3dc60167e41f8b015a7d3b0dc3dd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219633
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@google.com>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34136}
2021-05-26 15:47:03 +00:00
philipel
ca18809ee5 Move RtpFrameObject and EncodedFrame out of video_coding namespace.
Bug: webrtc:12579
Change-Id: Ib7ecd624eb5c54abb77fe08440a014aa1e963865
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33542}
2021-03-23 14:22:47 +00:00
philipel
6a6715042a Move RtpFrameReferenceFinder out of video_coding namespace.
Namespace used because of copy-pasting an old pattern, should never have been used in the first place. Removing it now to make followup refactoring prettier.

Bug: webrtc:12579
Change-Id: I00a80958401cfa368769dc0a1d8bbdd76aaa4ef5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212603
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33536}
2021-03-23 08:48:37 +00:00
philipel
9465978a3b Remove framemarking RTP extension.
BUG=webrtc:11637

Change-Id: I47f8e22473429c9762956444e27cfbafb201b208
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176442
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31522}
2020-06-15 11:18:00 +00:00
Danil Chapovalov
4c3a7dbe14 Remove RtpVideoHeader::discardable flag.
Calculate it when used instead

Bug: webrtc:11358
Change-Id: Ib79a4ce5e48a1a5244925471c005f96c5ec5dfd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173702
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31109}
2020-04-20 10:25:43 +00:00
Danil Chapovalov
97b59f060c Reduce RtpFrameReferenceFinder fuzzer input to more reasonable value
frame_id is unwraped from a 16bit value.
Getting to int64_t boundaries would take more than 2^48 packets.
That scenario considered unrealistic and thus untested.

Bug: chromium:1053482
Change-Id: Ib3f52d4528b20915b2330773f616d9304f45cac6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168682
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30607}
2020-02-25 14:15:24 +00:00
Danil Chapovalov
e209fe6c68 Do not propagate generic descriptor on receiving frame
It was used only for the frame decryptor.
Decryptor needs only raw representation that it can recreate
in a way compatible with the new version of the descriptor.

This relands commit abf73de8ea.
with adjustments.

Change-Id: I935977179bef31d8e1023964b967658e9a7db92d
Bug: webrtc:10342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168489
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30532}
2020-02-17 14:52:03 +00:00
Danil Chapovalov
bc1750d52b Revert "Do not propagate generic descriptor on receiving frame"
This reverts commit abf73de8ea.

Reason for revert: breaks downstream tests

Original change's description:
> Do not propagate generic descriptor on receiving frame
> 
> It was used only for the frame decryptor.
> Decryptor needs only raw representation that it can recreate
> in a way compatible with the new version of the descriptor.
> 
> Bug: webrtc:10342
> Change-Id: Ie098235ebb87c6f5e2af42d0022d2365cd6bfa29
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166163
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30501}

TBR=danilchap@webrtc.org,sprang@webrtc.org,philipel@webrtc.org

Change-Id: I6634df06ee75aa8cdfda614994ab11f7a5845c70
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168488
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30502}
2020-02-11 16:54:07 +00:00
Danil Chapovalov
abf73de8ea Do not propagate generic descriptor on receiving frame
It was used only for the frame decryptor.
Decryptor needs only raw representation that it can recreate
in a way compatible with the new version of the descriptor.

Bug: webrtc:10342
Change-Id: Ie098235ebb87c6f5e2af42d0022d2365cd6bfa29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166163
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30501}
2020-02-11 16:12:16 +00:00
philipel
2f7d779471 Use new RtpFrameObject ctor for fuzzing.
Bug: webrtc:10979
Change-Id: Idd3f09955e8c93738a677c447dad958cc50f4f66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155161
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29353}
2019-10-01 08:34:37 +00:00
Danil Chapovalov
f7457e55fe Store PacketBuffer by value instead of as reference counted object
Bug: None
Change-Id: I5a594972e8a8dad731c927a1a374301e549f5d71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153887
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29267}
2019-09-23 13:28:09 +00:00
philipel
0cff4fce55 Removed unused frame_size param from RtpFrameObject ctor.
Bug: webrtc:10979
Change-Id: Idde493dc7f5165e3ca173d5a38861b444b5904a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153668
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29253}
2019-09-20 10:56:01 +00:00
philipel
b5e4785464 RtpFrameObject now takes an EncodedImageBuffer in its ctor.
Bug: webrtc:10979
Change-Id: Ibc8b4a524ca95b5faa8850a41df8f2f0136a2969
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153666
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29251}
2019-09-20 10:15:01 +00:00
philipel
f0be5b5380 Make GetBitstream non-virtual since it is no longer needed for testing.
Bug: webrtc:10979
Change-Id: Id313c7fddbec40b9f19dae95f736379b872e3082
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153663
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29242}
2019-09-19 14:04:09 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
Johannes Kron
a370556270 Refactor to free up PacketBuffer as soon as possible
The packets belonging to a frame were kept in PacketBuffer
until the frame was decoded. This CL clears the dependencies
of an existing RtpFrameObject to PacketBuffer so that we can
free up PacketBuffer as soon as the RtpFrameObject is created.

Bug: none
Change-Id: Ic939be91815519ae1d1c67ada82006417b2d26a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149818
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28977}
2019-08-28 08:07:32 +00:00
Johannes Kron
10da4a0025 Fix RtpFrameReferenceFinderFuzzer to not generate invalid input
Make sure that the packets in the packet buffer belonging to the
first and last sequence numbers are marked as first and last,
respectively.

Bug: chromium:989856
Change-Id: I57bdd7d62d585be2d2083a6b5ce67fce89ab4389
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147875
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28769}
2019-08-06 09:03:10 +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
Chen Xing
f00bf42d1c Add plumbing of RtpPacketInfos to each VideoFrame as input for SourceTracker.
This change adds the plumbing of RtpPacketInfo from RtpVideoStreamReceiver::OnRtpPacket() to VideoReceiveStream::OnFrame() for video. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.

Bug: webrtc:10668
Change-Id: Ib97d430530c5a8487d3b129936c7c51e118889bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139891
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28332}
2019-06-20 10:24:29 +00:00
Niels Möller
d5e02f0b92 Delete redundant members from VCMPacket.
The values are available as part of the RTPVideoHeader member.

Bug: None
Change-Id: I832fffc449929badec3796d7096c9cdc0d43d344
Reviewed-on: https://webrtc-review.googlesource.com/c/123234
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26773}
2019-02-20 14:39:10 +00:00
Ilya Nikolaevskiy
4348ce240a Calculate min and max receive timestamps for packets in a video frame
Bug: webrtc:10106
Change-Id: I1d3469abb1e7bb7c91a5912d7b781505526abaca
Reviewed-on: https://webrtc-review.googlesource.com/c/113507
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25935}
2018-12-07 16:22:34 +00:00
Sam Zackrisson
262047055d Update fuzzer max input length handling
The docs have been updated. max_len is libfuzzer specific, new way is
fuzzer agnostic.

Docs:
https://chromium.googlesource.com/chromium/src/+/master/testing/libfuzzer/getting_started.md#improving-your-fuzz-target

Bug: chromium:895082
Test: flexfec_sender_fuzzer input size still converges at <=200 after running locally for 5-10 minutes.
Change-Id: I7a5ce95cb4d8b8ca461f6e502b81b599daa855f9
Reviewed-on: https://webrtc-review.googlesource.com/c/107883
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25361}
2018-10-25 12:19:18 +00:00
Karl Wiberg
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
philipel
c1ee9d57d1 RtpFrameReferenceFinder fuzzer.
Bug: webrtc:7728
Change-Id: I641772837384a4d8070db2138b93f4157f997d03
Reviewed-on: https://webrtc-review.googlesource.com/74584
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23271}
2018-05-17 08:55:31 +00:00