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Author SHA1 Message Date
Erik Språng
c12f625938 Adds VideoDecoder::GetDecoderInfo() 
This adds a new way to poll decoder metadata.
A default implementation still delegates to the old methods.
Root call site is updates to not use the olds methods.

Follow-ups will dismantle usage of the olds methods in wrappers.

Bug: webrtc:12271
Change-Id: Id0fa6863c96ff9e3b849da452d6540e7c5da4512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196520
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32976}
 2021-01-14 13:33:22 +00:00
Niels Möller
d381eede92 Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h 
We can then finally delete the top-level common_types.h, and the
corresponding build target webrtc_common.

Bug: webrtc:7660
Change-Id: I1c1096541477586d90774c7a3405b9d36edec14a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182800
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32044}
 2020-09-07 08:37:14 +00:00
Philip Eliasson
2b068ce1b8 Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." 
This reverts commit a4f23ad0ce4382e3a11bc6a8c1f9f6183e722fd8.

Reason for revert: Downstream fix landed.

TBR=mflodman@webrtc.org

Original change's description:
> Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
>
> This reverts commit acb9d8365a5f9eb1e2a9e9902690d62dab1e5759.
>
> Reason for revert: Break downstream stuff.
>
> Original change's description:
> > Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory.
> >
> > Bug: webrtc:9106
> > Change-Id: I85712f3ab6a734d3fad7819491d3b8e3388b47e7
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180342
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31834}
>
> TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org
>
> Change-Id: I6cfdb85a154a78135839f84edf5f69673d5ab715
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9106
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180807
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31835}

TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org

# Not skipping CQ checks because this is a reland.

Bug: webrtc:9106
Change-Id: I03b3e68532107bec37bcc6e47a5489c84fe91ef9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180808
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31866}
 2020-08-06 11:50:08 +00:00
Philip Eliasson
a4f23ad0ce Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." 
This reverts commit acb9d8365a5f9eb1e2a9e9902690d62dab1e5759.

Reason for revert: Break downstream stuff.

Original change's description:
> Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory.
> 
> Bug: webrtc:9106
> Change-Id: I85712f3ab6a734d3fad7819491d3b8e3388b47e7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180342
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31834}

TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org

Change-Id: I6cfdb85a154a78135839f84edf5f69673d5ab715
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9106
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180807
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31835}
 2020-08-03 15:45:41 +00:00
philipel
acb9d8365a Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory. 
Bug: webrtc:9106
Change-Id: I85712f3ab6a734d3fad7819491d3b8e3388b47e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180342
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31834}
 2020-08-03 14:10:37 +00:00
philipel
668b0530e1 Removed unused MockVideoDecoder from VideoReceiveStream tests. 
Bug: webrtc:9106
Change-Id: I2ccb3287dd87b1dce74b73727e1264a75c87a010
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180121
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31786}
 2020-07-23 11:51:53 +00:00
Markus Handell
3d2210876e Remove unused critical section includes. 
Bug: webrtc:11567
Change-Id: Ic5e43c51ce06c0619adc265d12ad4bef73a9df76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179521
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31745}
 2020-07-16 13:52:28 +00:00
Danil Chapovalov
91fdc607d8 In video/ replace mock macros with unified MOCK_METHOD macro 
Bug: webrtc:11564
Change-Id: Iec9dded7a3f045e048d0546c268ae206a7d3a7c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175128
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31264}
 2020-05-14 18:12:40 +00:00
Danil Chapovalov
0c626afcf3 Use newer version of TimeDelta and TimeStamp factories in webrtc 
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format

Bug: None
Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30491}
 2020-02-10 12:21:17 +00:00
Sebastian Jansson
340af975e9 Always enter yield policy scope using simulated TimeControllers. 
This makes the class easier to use at a minor cost of making it slightly
more magic.

Bug: webrtc:9883
Change-Id: If807cfbf046615333c3bcd3b58a001813102a9f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161231
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30008}
 2019-12-04 17:16:32 +00:00
Markus Handell
486cc55a02 TimeController: Rename Sleep to AdvanceTime. 
This change renames TimeController's Sleep method to AdvanceTime, unifying
the same name with the same semantic as for downstream projects.

Bug: webrtc:11154
Change-Id: Id79bcf0eafcd0b47a76407ba220479d84df5a736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161092
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29989}
 2019-12-03 16:08:54 +00:00
Markus Handell
269ac81a86 VideoReceiveStream: Enable encoded frame sink. 
This change ultimately enables wiring up VideoRtpReceiver::OnGenerateKeyFrame and
OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded frames
can flow to sinks installed in VideoTrackSourceInterface.

Bug: chromium:1013590
Change-Id: I0779932c251a2159880a39b2d42d5ce439cc88e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161090
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29988}
 2019-12-03 15:55:04 +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
3b69817e62 Revert "Reland "Preserve min and max playout delay from RTP header extension"" 
This reverts commit 87bed4793ff8f463202f442381339626d0b27f0d.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Reland "Preserve min and max playout delay from RTP header extension"
> 
> This reverts commit f31cc08ba01ed403e89255b5f3f38d5dbdde855e.
> 
> Reason for revert: Reland with fixes
> 
> Original change's description:
> > Revert "Preserve min and max playout delay from RTP header extension"
> > 
> > This reverts commit 85ba9972c42544c4771e394c9aa1d20bf5d09a91.
> > 
> > Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream.
> > 
> > Original change's description:
> > > Preserve min and max playout delay from RTP header extension
> > > 
> > > Audio and video synchronization can sometimes override the minimum
> > > and maximum playout delay that is set through the RTP header
> > > extension. This CL makes sure that the playout delay always is
> > > within the limits set by the RTP header extension.
> > > 
> > > Bug: webrtc:10886
> > > Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#28980}
> > 
> > TBR=stefan@webrtc.org,kron@webrtc.org
> > 
> > Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:10886
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28984}
> 
> TBR=stefan@webrtc.org,kron@webrtc.org
> 
> Change-Id: I5a3908a8c45f7faedab6f009b22df81d674e13a0
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10886
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150653
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28985}

TBR=stefan@webrtc.org,kron@webrtc.org

Change-Id: Id2e5d1ff804881e956a07fa4ae0f8301895dcc95
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10886
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150654
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28986}
 2019-08-28 12:41:56 +00:00
Johannes Kron
87bed4793f Reland "Preserve min and max playout delay from RTP header extension" 
This reverts commit f31cc08ba01ed403e89255b5f3f38d5dbdde855e.

Reason for revert: Reland with fixes

Original change's description:
> Revert "Preserve min and max playout delay from RTP header extension"
> 
> This reverts commit 85ba9972c42544c4771e394c9aa1d20bf5d09a91.
> 
> Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream.
> 
> Original change's description:
> > Preserve min and max playout delay from RTP header extension
> > 
> > Audio and video synchronization can sometimes override the minimum
> > and maximum playout delay that is set through the RTP header
> > extension. This CL makes sure that the playout delay always is
> > within the limits set by the RTP header extension.
> > 
> > Bug: webrtc:10886
> > Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28980}
> 
> TBR=stefan@webrtc.org,kron@webrtc.org
> 
> Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10886
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28984}

TBR=stefan@webrtc.org,kron@webrtc.org

Change-Id: I5a3908a8c45f7faedab6f009b22df81d674e13a0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10886
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150653
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28985}
 2019-08-28 12:40:53 +00:00
Johannes Kron
f31cc08ba0 Revert "Preserve min and max playout delay from RTP header extension" 
This reverts commit 85ba9972c42544c4771e394c9aa1d20bf5d09a91.

Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream.

Original change's description:
> Preserve min and max playout delay from RTP header extension
> 
> Audio and video synchronization can sometimes override the minimum
> and maximum playout delay that is set through the RTP header
> extension. This CL makes sure that the playout delay always is
> within the limits set by the RTP header extension.
> 
> Bug: webrtc:10886
> Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28980}

TBR=stefan@webrtc.org,kron@webrtc.org

Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10886
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28984}
 2019-08-28 12:38:43 +00:00
Johannes Kron
85ba9972c4 Preserve min and max playout delay from RTP header extension 
Audio and video synchronization can sometimes override the minimum
and maximum playout delay that is set through the RTP header
extension. This CL makes sure that the playout delay always is
within the limits set by the RTP header extension.

Bug: webrtc:10886
Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28980}
 2019-08-28 11:00:02 +00:00
Chen Xing
17f9ee5358 Enable VideoReceiveStreamTestWithFakeDecoder.RenderedFrameUpdatesGetSources for iOS. 
This change re-enables a previously flaky unit tests for iOS. It seems to have the same root cause as webrtc:10827 and which was fixed by: https://webrtc-review.googlesource.com/c/src/+/149171

Bug: webrtc:10872, webrtc:10827
Change-Id: I71b2581cf8c75e0dd6a39b77e6cf34c121ff22f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149802
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28893}
 2019-08-19 10:44:42 +00:00
Sami Kalliomäki
c14b2335d9 Disable the most flaky tests on iOS. 
Disables:
 - RtpVideoSenderTest.DoesNotRetrasmitAckedPackets
 - VideoReceiveStreamTestWithFakeDecoder.RenderedFrameUpdatesGetSources

Bug: webrtc:10870, webrtc:10872, webrtc:10873
Change-Id: I410f781c30e45387c35055bf97424a68658174d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148984
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28858}
 2019-08-14 15:42:11 +00:00
Ilya Nikolaevskiy
a0e2609a08 Partially revert of ColorSpace information copying around decoders 
This partially reverts these 2 CLs:
1) Reland "Copy video frames metadata between encoded and plain frames in one place"
https://webrtc.googlesource.com/src/+/2ebf5239782bf6b46d4aa812f34fa9f9e5a02be9

2) Don't copy video frame metadata in each encoder/decoder
https://webrtc.googlesource.com/src/+/ab62b2ee51e622be6d0aade15e87e927fa60e6f2

The problem with them were that ColorSpace was made to always be copied from the
EncodedImage in the GenericDecoder, which overwrote ColorSpace information from
the decoder.

If decoder applied color space transition or bitstream color space information
was different from the WebRTC signaled one, the incorrect color space data were
passed to the renderer.

This CL removes introduced change regarding color space data: GenericDecoder
doesn't copy or store it and software decoders are restored to copy it.
Relevant tests are also removed.

Bug: chromium:982486
Change-Id: I989e01476ff7f7df376c05578ab8f540b95a1dd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145323
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28556}
 2019-07-12 11:27:07 +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
90f3b89550 Replace the implementation of GetContributingSources() on the video side. 
This change replaces the `ContributingSources`-implementation of `GetContributingSources()` and `GetSynchronizationSources()` on the video side with the spec-compliant `SourceTracker`-implementation.

The most noticeable impact is that the per-frame dictionaries are now updated when frames are delivered to the RTCRtpReceiver's MediaStreamTrack rather than when RTP packets are received on the network.

Bug: webrtc:10545
Change-Id: I895b5790280ac94c1501801d226c643633c67349
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143177
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28386}
 2019-06-26 11:57:33 +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
Ilya Nikolaevskiy
2ebf523978 Reland "Copy video frames metadata between encoded and plain frames in one place" 
Reland with fixes.

Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.

Also, added some missing tests.

Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346

Bug: webrtc:10460
Change-Id: I98629589fa55ca1d74056033cf86faccfdf848cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136582
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27930}
 2019-05-13 14:51:11 +00:00
Ilya Nikolaevskiy
de20b9683c Revert "Reland "Copy video frames metadata between encoded and plain frames in one place"" 
This reverts commit 4fb12b0caec9faa57cfbceb0f86b0e10c32a0cc2.

Reason for revert: Breaks some asan chromium bots

Original change's description:
> Reland "Copy video frames metadata between encoded and plain frames in one place"
> 
> Reland with fixes.
> 
> Currently some video frames metadata like rotation or ntp timestamps are
> copied in every encoder and decoder separately. This CL makes copying to
> happen at a single place for send or receive side. This will make it
> easier to add new metadata in the future.
> 
> Also, added some missing tests.
> 
> Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
> 
> Bug: webrtc:10460
> Change-Id: Ia71198685de7fbd990704b575231cdce94dc0645
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134961
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27828}

TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,kron@webrtc.org,artit@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10460
Change-Id: I9c87a43a716622b389974cb8377f973573fc29a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135747
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27895}
 2019-05-09 17:47:51 +00:00
Ilya Nikolaevskiy
4fb12b0cae Reland "Copy video frames metadata between encoded and plain frames in one place" 
Reland with fixes.

Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.

Also, added some missing tests.

Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346

Bug: webrtc:10460
Change-Id: Ia71198685de7fbd990704b575231cdce94dc0645
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134961
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27828}
 2019-05-02 13:29:14 +00:00
Artem Titarenko
4b1afbe60a Revert "Reland "Copy video frames metadata between encoded and plain frames in one place"" 
This reverts commit c9a2c5e93aa51606916e6728454bcff26bb75f79.

Reason for revert: Breaks downstream test

Original change's description:
> Reland "Copy video frames metadata between encoded and plain frames in one place"
> 
> Reland with fixes: Do not remove extra metadata copies in software decoders as some downstream projects assumes these fields are copied by the encoders.
> 
> Currently some video frames metadata like rotation or ntp timestamps are
> copied in every encoder and decoder separately. This CL makes copying to
> happen at a single place for send or receive side. This will make it
> easier to add new metadata in the future.
> 
> Also, added some missing tests.
> 
> Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
> 
> Bug: webrtc:10460
> Change-Id: I8e49589bf75f406e2b5ddee34882de0faedbd09a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134102
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27756}

TBR=ilnik@webrtc.org,sprang@webrtc.org,kron@webrtc.org,artit@webrtc.org

Change-Id: I34cc563ec6383735c2a76a6f45a72a7726b74421
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134204
Reviewed-by: Artem Titarenko <artit@google.com>
Commit-Queue: Artem Titarenko <artit@google.com>
Cr-Commit-Position: refs/heads/master@{#27765}
 2019-04-25 11:39:31 +00:00
Ilya Nikolaevskiy
c9a2c5e93a Reland "Copy video frames metadata between encoded and plain frames in one place" 
Reland with fixes: Do not remove extra metadata copies in software decoders as some downstream projects assumes these fields are copied by the encoders.

Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.

Also, added some missing tests.

Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346

Bug: webrtc:10460
Change-Id: I8e49589bf75f406e2b5ddee34882de0faedbd09a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134102
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27756}
 2019-04-25 09:13:15 +00:00
Artem Titarenko
84ae2b6efd Revert "Copy video frames metadata between encoded and plain frames in one place" 
This reverts commit 00d0a0a1a9520fb4323d7e3a1c02133b7b942978.

Reason for revert: Breaks downstream tests

Original change's description:
> Copy video frames metadata between encoded and plain frames in one place
> 
> Currently some video frames metadata like rotation or ntp timestamps are
> copied in every encoder and decoder separately. This CL makes copying to
> happen at a single place for send or receive side. This will make it
> easier to add new metadata in the future.
> 
> Also, added some missing tests.
> 
> Bug: webrtc:10460
> Change-Id: Ia49072c3041e75433f125a61050d2982b2bec1da
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27719}

TBR=ilnik@webrtc.org,sprang@webrtc.org,kron@webrtc.org

Change-Id: I8960a6cc15e552925129ba0037f197ff3fd93c25
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134100
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27737}
 2019-04-24 12:56:52 +00:00
Ilya Nikolaevskiy
00d0a0a1a9 Copy video frames metadata between encoded and plain frames in one place 
Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.

Also, added some missing tests.

Bug: webrtc:10460
Change-Id: Ia49072c3041e75433f125a61050d2982b2bec1da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27719}
 2019-04-23 14:31:03 +00:00
Danil Chapovalov
d3ba236686 Stop using GlobalTaskQueueFactory in video unittests 
instead use DefaultTaskQueueFactory directly

Bug: webrtc:10284
Change-Id: I58ae120cf185553d0145d7feb365deca90a93bc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132401
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27610}
 2019-04-15 09:24:18 +00:00
Niels Möller
7aacdd9515 Reland "Delete CodecSpecificInfo argument from VideoDecoder::Decode" 
This is a reland of 39d3a7de02d63894d12e7332322e1d80cd7c0d40

Original change's description:
> Delete CodecSpecificInfo argument from VideoDecoder::Decode
>
> Bug: webrtc:10379
> Change-Id: I079b419604bf4e9c1994fe203d7db131a0ccddb6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125920
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27022}

Tbr: kwiberg@webrtc.org
Bug: webrtc:10379
Change-Id: I8197bebd2ae7dc460644a98795b8257b033c27c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126480
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27565}
 2019-04-11 13:03:52 +00:00
Mirko Bonadei
6a489f22c7 Fully qualify googletest symbols. 
Semi-automatically created with:

git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format

After this, two .cc files failed to compile and I have fixed them
manually.

Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
 2019-04-09 17:18:20 +00:00
Jeroen de Borst
2c7b9825bc Revert "Delete CodecSpecificInfo argument from VideoDecoder::Decode" 
This reverts commit 39d3a7de02d63894d12e7332322e1d80cd7c0d40.

Reason for revert: This change broke an internal project

Original change's description:
> Delete CodecSpecificInfo argument from VideoDecoder::Decode
> 
> Bug: webrtc:10379
> Change-Id: I079b419604bf4e9c1994fe203d7db131a0ccddb6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125920
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27022}

TBR=brandtr@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org

Change-Id: I2c730cc1834a3b23203fae3d7881f0890802c37b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10379
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126320
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27026}
 2019-03-07 19:40:17 +00:00
Niels Möller
39d3a7de02 Delete CodecSpecificInfo argument from VideoDecoder::Decode 
Bug: webrtc:10379
Change-Id: I079b419604bf4e9c1994fe203d7db131a0ccddb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125920
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27022}
 2019-03-07 16:18:49 +00:00
Sebastian Jansson
74682c1191 Inject TaskQueueFactory to video streams. 
Bug: webrtc:10365
Change-Id: Ib655d8eac4467926bcb86cf2cb3728eabf5342d8
Reviewed-on: https://webrtc-review.googlesource.com/c/125089
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26921}
 2019-03-01 11:35:39 +00:00
Ruslan Burakov
493a650b1e Propagate base minimum delay from video jitter buffer to webrtc/api. 
On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface,
GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes
the same concept.


Bug: webrtc:10287
Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957
Reviewed-on: https://webrtc-review.googlesource.com/c/123482
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26877}
 2019-02-27 15:08:34 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
 2019-01-11 17:11:39 +00:00
Niels Möller
c572ff3c71 Add default constructor for rtc::Event 
Bug: webrtc:9962
Change-Id: Icaa91e657e6881fcb1553f354c07866109a0ea68
Reviewed-on: https://webrtc-review.googlesource.com/c/109500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25535}
 2018-11-07 08:57:50 +00:00
Niels Möller
cbcbc22568 Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." 
This is a reland of 529d0d9795b81dbed5e4231f15d3752a5fc0df32

Original change's description:
> Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.
> 
> Preparation for deleting EnableFrameRecordning, and also a step
> towards landing of the new VideoStreamDecoder.
> 
> Bug: webrtc:9106
> Change-Id: I50964ee458b08a702ec69b82a62e4995c57cee82
> Reviewed-on: https://webrtc-review.googlesource.com/97660
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24861}

Bug: webrtc:9106
Change-Id: I2eb894773b3f33ff6a980e8008e8248607e32668
Reviewed-on: https://webrtc-review.googlesource.com/102480
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24882}
 2018-09-28 08:48:02 +00:00
Sebastian Jansson
377b26ec65 Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." 
This reverts commit efb94d57eb88638c323d93dddc281390dada5021.

Reason for revert: Investigate and fix build errors.

Original change's description:
> Revert "Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.""
>
> This reverts commit 7961dc2dbdb3391a003d63630d5107e258ff3e78.
>
> Reason for revert: WebRTC does not build
>
> Original change's description:
> > Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."
> >
> > This reverts commit 529d0d9795b81dbed5e4231f15d3752a5fc0df32.
> >
> > Reason for revert: Seems to break perf tests, likely some breakage in video_quality_tests decoder configuration.
> >
> > Original change's description:
> > > Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.
> > >
> > > Preparation for deleting EnableFrameRecordning, and also a step
> > > towards landing of the new VideoStreamDecoder.
> > >
> > > Bug: webrtc:9106
> > > Change-Id: I50964ee458b08a702ec69b82a62e4995c57cee82
> > > Reviewed-on: https://webrtc-review.googlesource.com/97660
> > > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#24861}
> >
> > TBR=brandtr@webrtc.org,nisse@webrtc.org,andersc@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,srte@webrtc.org
> >
> > Change-Id: Id34e4a3452a7dbc06167a4df5bb4c2825ebd7bd0
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:9106
> > Reviewed-on: https://webrtc-review.googlesource.com/102421
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24866}
>
> TBR=brandtr@webrtc.org,nisse@webrtc.org,andersc@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,srte@webrtc.org
>
> Change-Id: I23a439e1ceef79109b1f966b80b2663203968269
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9106
> Reviewed-on: https://webrtc-review.googlesource.com/102422
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24867}

TBR=brandtr@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,andersc@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,srte@webrtc.org

Change-Id: I9dafbc070e7f39dcb0ddbd61cb620164258fe894
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9106
Reviewed-on: https://webrtc-review.googlesource.com/102460
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24872}
 2018-09-27 16:04:50 +00:00
Oleh Prypin
efb94d57eb Revert "Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."" 
This reverts commit 7961dc2dbdb3391a003d63630d5107e258ff3e78.

Reason for revert: WebRTC does not build

Original change's description:
> Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."
> 
> This reverts commit 529d0d9795b81dbed5e4231f15d3752a5fc0df32.
> 
> Reason for revert: Seems to break perf tests, likely some breakage in video_quality_tests decoder configuration.
> 
> Original change's description:
> > Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.
> > 
> > Preparation for deleting EnableFrameRecordning, and also a step
> > towards landing of the new VideoStreamDecoder.
> > 
> > Bug: webrtc:9106
> > Change-Id: I50964ee458b08a702ec69b82a62e4995c57cee82
> > Reviewed-on: https://webrtc-review.googlesource.com/97660
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24861}
> 
> TBR=brandtr@webrtc.org,nisse@webrtc.org,andersc@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,srte@webrtc.org
> 
> Change-Id: Id34e4a3452a7dbc06167a4df5bb4c2825ebd7bd0
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9106
> Reviewed-on: https://webrtc-review.googlesource.com/102421
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24866}

TBR=brandtr@webrtc.org,nisse@webrtc.org,andersc@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,srte@webrtc.org

Change-Id: I23a439e1ceef79109b1f966b80b2663203968269
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9106
Reviewed-on: https://webrtc-review.googlesource.com/102422
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24867}
 2018-09-27 13:55:44 +00:00
Niels Moller
7961dc2dbd Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." 
This reverts commit 529d0d9795b81dbed5e4231f15d3752a5fc0df32.

Reason for revert: Seems to break perf tests, likely some breakage in video_quality_tests decoder configuration.

Original change's description:
> Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.
> 
> Preparation for deleting EnableFrameRecordning, and also a step
> towards landing of the new VideoStreamDecoder.
> 
> Bug: webrtc:9106
> Change-Id: I50964ee458b08a702ec69b82a62e4995c57cee82
> Reviewed-on: https://webrtc-review.googlesource.com/97660
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24861}

TBR=brandtr@webrtc.org,nisse@webrtc.org,andersc@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,srte@webrtc.org

Change-Id: Id34e4a3452a7dbc06167a4df5bb4c2825ebd7bd0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9106
Reviewed-on: https://webrtc-review.googlesource.com/102421
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24866}
 2018-09-27 13:24:13 +00:00
Niels Möller
529d0d9795 Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config. 
Preparation for deleting EnableFrameRecordning, and also a step
towards landing of the new VideoStreamDecoder.

Bug: webrtc:9106
Change-Id: I50964ee458b08a702ec69b82a62e4995c57cee82
Reviewed-on: https://webrtc-review.googlesource.com/97660
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24861}
 2018-09-27 11:25:21 +00:00
Niels Möller
cb7e1d2edb Use SdpVideoFormat in VideoReceiveStream::Decoder 
Replaces payload_name and codec_params.

Tbr: srte@webrtc.org
Bug: webrtc:9106
Change-Id: Ib45c501c6eb41e92fbb24ab00ada18bf10be42ed
Reviewed-on: https://webrtc-review.googlesource.com/98161
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24691}
 2018-09-11 15:03:04 +00:00
philipel
6ad6e1f04c Removed old and unused WebRTC-NewVideoJitterBuffer field trial from VideoReceiveStreamTest. 
Bug: none
Change-Id: Ide15295feb8ebba71a11ac083f8ca84902c4d24c
Reviewed-on: https://webrtc-review.googlesource.com/98560
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24619}
 2018-09-07 09:49:59 +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
Niels Möller
8df3a388a3 Deprecate RTPFragmentationHeader argument to VideoDecoder::Decode 
Intend to delete in a later cl.

Bug: webrtc:6471
Change-Id: Icf0fcd40e0d3287dc59b684fae6552b40b47204a
Reviewed-on: https://webrtc-review.googlesource.com/39511
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23162}
 2018-05-08 08:09:35 +00:00
Tommi
38c5d9345d Reduce locking for CallStats (preparation for TaskQueue). 
Reduce synchronization in the class significantly and not hold a lock
while calling out to external implementations.

* Rewrite tests to use a real ProcessThread.
* Update some code to use C++ 11 constructs & library features.

Bug: webrtc:9064
Change-Id: I240a819efb6ef8197da3f2edf7acf068d2a27e8b
Reviewed-on: https://webrtc-review.googlesource.com/64521
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22649}
 2018-03-28 13:24:07 +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