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Author SHA1 Message Date
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