39 Commits

Author SHA1 Message Date
Åsa Persson
e644a03195 Add field trial for rampup in quality based on available bandwidth.
Bug: none
Change-Id: I32e1ea6fb2f2e20fc631e09b02c8f3a11b6c9fac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158888
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29751}
2019-11-11 10:13:28 +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
Elad Alon
8f01c4e1b6 Define FecControllerOverride and plumb it down to VideoEncoder
The purpose of this interface is to allow VideoEncoder to override
the bandwidth allocation set by FecController in RtpVideoSender.

This CL defines the interface and sends it down to VideoSender.
Two upcoming CLs will:
1. Make LibvpxVp8Encoder pass it on to the (injectable)
   FrameBufferController, where it might be put to good use.
2. Modify RtpVideoSender to respond to the message sent to it
   via this API.

TBR=kwiberg@webrtc.org

Bug: webrtc:10769
Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28416}
2019-06-28 15:57:22 +00:00
Elad Alon
370f93a34a Reland "Inform VideoEncoder of negotiated capabilities"
This is a reland of 11dfff0878c949f2e19d95a0ddc209cdad94b3b4

Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.

Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}

TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org

Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
2019-06-11 14:49:37 +00:00
Philip Eliasson
49d661a7d3 Revert "Inform VideoEncoder of negotiated capabilities"
This reverts commit 11dfff0878c949f2e19d95a0ddc209cdad94b3b4.

Reason for revert: Downstream import failure.

Original change's description:
> Inform VideoEncoder of negotiated capabilities
> 
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
> 
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}

TBR=sakal@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org

Change-Id: I7f833055c67f1f879b01dd8c156ba7b8840e8747
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141411
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28225}
2019-06-11 11:56:04 +00:00
Elad Alon
11dfff0878 Inform VideoEncoder of negotiated capabilities
After this CL lands, an announcement will be made to
discuss-webrtc about the deprecation of one version
of InitEncode().

Bug: webrtc:10720
Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28224}
2019-06-11 11:32:13 +00:00
Erik Språng
16cb8f5d74 Reland "Replace usage of old SetRates/SetRateAllocation methods"
This is a reland of 7ac0d5f348f0b956089c4ed65c46e65bac125508

Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}

TBR=brandtr@webrtc.org,sakal@webrtc.org,perkj@webrtc.org

Bug: webrtc:10481
Change-Id: I2978d5c527a18e885b7845c4e53a2424e8ad5b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132551
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27593}
2019-04-12 13:37:32 +00:00
Minyue Li
7ddef1af88 Revert "Replace usage of old SetRates/SetRateAllocation methods"
This reverts commit 7ac0d5f348f0b956089c4ed65c46e65bac125508.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
> 
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
> 
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}

TBR=brandtr@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org

Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}
2019-04-11 10:50:29 +00:00
Erik Språng
7ac0d5f348 Replace usage of old SetRates/SetRateAllocation methods
This rather large CL replaces all relevant usage of the old
VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
API is unchanged to allow downstream projects to update without
breakage.

Bug: webrtc:10481
Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27554}
2019-04-11 07:46:09 +00:00
Sebastian Jansson
b55015e4e1 Replacing SequencedTaskChecker with SequenceChecker.
Bug: webrtc:9883
Change-Id: I5e3189da2a46e6f4ed1a3c5a5dfd2f7d75a16b5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130961
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27518}
2019-04-09 12:28:04 +00:00
Niels Möller
9d8eaac4ee Delete unneeded direct includes of common_types.h
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:

api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/

There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.

Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
2019-04-01 07:18:13 +00:00
Danil Chapovalov
22ed366fec Avoid using global task queue factory in fake encoder
by not using convenient rtc::TaskQueue constructor

Bug: webrtc:10284
Change-Id: I3c6703fd8c86b2a230f62cac734eb616039e4abe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128603
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27207}
2019-03-20 12:33:05 +00:00
Niels Möller
b859b326ba Update more VideoEncoder implementations to drop CodecSpecificInfo input
Followup to https://webrtc-review.googlesource.com/c/src/+/125900.

Bug: webrtc:10379
Change-Id: If81c50c862bbcfd65a3cf7000c8327ebafe519c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126002
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27016}
2019-03-07 12:26:57 +00:00
Niels Möller
87e2d785a0 Prepare for splitting FrameType into AudioFrameType and VideoFrameType
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.

After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.

Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
2019-03-07 10:12:57 +00:00
Niels Möller
d738071e63 Refactor FakeEncoder to avoid writing to a const EncodedImage
Subclasses of FakeEncoder need to fill out the CodecSpecificInfo and
RTPFragmentationHeader, and they also write to the encoded data of the
EncodedImage. This used to be done by subclasses chaining onto the
parent's OnEncodedImage callback, but that's not so nice, since the
EncodedImage argument is passed as a const ref there.

This change introduces a protected method EncodeHook for this purpose.
FakeEncoder calls this prior to calling OnEncodedImage, with non-const
pointers.

In addition, change FakeEncoder to use EncodedImage::Allocate, rather
than explicit new and delete.

Bug: webrtc:9378
Change-Id: Ie8182d1d5224aa3b7f15905612f6dbcebef0a555
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125880
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26988}
2019-03-06 11:11:48 +00:00
Per Kjellander
65cc52ebca Fix heap use overrun in FakeEncoder
By removing unnecessary fixed size buffer.

BUG=webrtc:10276

Change-Id: I303303d8c4aa356372875abe6db5711cd10bcc71
Reviewed-on: https://webrtc-review.googlesource.com/c/120811
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26509}
2019-02-01 11:26:57 +00:00
Emircan Uysaler
7c03bdc1d3 Reland "Add a high bitrate full stack test with fake codec"
In this reland, I disabled high bitrate webrtc perf test on Android32.

This is a reland of 15df2774f4e85cf8900768c1793edcf17d651dcd

Original change's description:
> This CL adds a fake codec factory  in WebRTC that can be used in tests to
> produce target bitrate output.

> We also add a high bitrate test that makes use of fake codec. This test assumes
> ideal network conditions with target bandwidth being available and exercises
> WebRTC calls with a high target bitrate(100 Mbps) end-to-end.

TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,emircan@webrtc.org,kron@webrtc.org

Bug: chromium:879723
Change-Id: I31a4b48d986bef9ca003ae71afeb567ae3e562c9
Reviewed-on: https://webrtc-review.googlesource.com/c/117980
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26285}
2019-01-16 21:03:22 +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
Oskar Sundbom
8984cd61ca Revert "Add a high bitrate full stack test with fake codec"
This reverts commit 15df2774f4e85cf8900768c1793edcf17d651dcd.

Reason for revert: It's causing the Android perf bots to fail. E.g.: https://ci.chromium.org/buildbot/client.webrtc.perf/Android32%20Tests%20%28L%20Nexus4%29/6666

Original change's description:
> Add a high bitrate full stack test with fake codec
> 
> This CL adds a fake codec factory  in WebRTC that can be used in tests to
> produce target bitrate output.
> 
> We also add a high bitrate test that makes use of fake codec. This test assumes
> ideal network conditions with target bandwidth being available and exercises
> WebRTC calls with a high target bitrate(100 Mbps) end-to-end.
> 
> Bug: chromium:879723
> Change-Id: I981124e2087054ed72c5447e239f28aae0878e29
> Reviewed-on: https://webrtc-review.googlesource.com/c/97185
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26182}

TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,emircan@webrtc.org,kron@webrtc.org

Change-Id: I33cd01ce345d81d66543f9be99750fa100760b5c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:879723
Reviewed-on: https://webrtc-review.googlesource.com/c/116785
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26192}
2019-01-10 11:49:05 +00:00
Emircan Uysaler
15df2774f4 Add a high bitrate full stack test with fake codec
This CL adds a fake codec factory  in WebRTC that can be used in tests to
produce target bitrate output.

We also add a high bitrate test that makes use of fake codec. This test assumes
ideal network conditions with target bandwidth being available and exercises
WebRTC calls with a high target bitrate(100 Mbps) end-to-end.

Bug: chromium:879723
Change-Id: I981124e2087054ed72c5447e239f28aae0878e29
Reviewed-on: https://webrtc-review.googlesource.com/c/97185
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26182}
2019-01-09 23:49:03 +00:00
Mirta Dvornicic
1ec2a16121 Revert "Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo"
This reverts commit cdc5eb0de179dcc866ef770ea303879c64466879.

Reason for revert: Causes wrong CPU adaptation to be used for some HW codecs since GetEncoderInfo() is polled before InitEncode().

Original change's description:
> Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo
> 
> Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
> until it is removed downstream and remove all implementations of it.
> 
> Bug: webrtc:10065
> Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
> Reviewed-on: https://webrtc-review.googlesource.com/c/113065
> Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25924}

TBR=brandtr@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,mirtad@webrtc.org

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

Bug: webrtc:10065
Change-Id: Idaa452e1d8c1c58cdb4ec69b88fce9042589cc3c
Reviewed-on: https://webrtc-review.googlesource.com/c/113800
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25943}
2018-12-10 10:36:00 +00:00
Mirta Dvornicic
cdc5eb0de1 Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo
Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
until it is removed downstream and remove all implementations of it.

Bug: webrtc:10065
Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
Reviewed-on: https://webrtc-review.googlesource.com/c/113065
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25924}
2018-12-06 15:24:45 +00:00
Yves Gerey
3e70781361 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.

Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 18:25:07 +00:00
Niels Möller
2222a80e79 Delete unneeded includes of common_types.h and gn deps on webrtc_common.
Bug: webrtc:5876
Change-Id: Iae14e5f1679067a5a5e0584ca830aee0870c8807
Reviewed-on: https://webrtc-review.googlesource.com/c/111463
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25715}
2018-11-20 16:28:39 +00:00
Per Kjellander
17fc7e2320 Add counter to the end of FakeEncoder frames in order to make them unique.
Bug: none
Change-Id: Ia5dccbe1868cc3fda5693d7e13eac21f75a08b83
Reviewed-on: https://webrtc-review.googlesource.com/c/109582
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25536}
2018-11-07 09:08:35 +00:00
Erik Språng
df351f4b54 Update FakeEncoder to use EncoderInfo
Bug: webrtc:9890
Change-Id: Ia6f268b35b5cc62fb6318f59bd66bbf65e97a23d
Reviewed-on: https://webrtc-review.googlesource.com/c/109565
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25515}
2018-11-06 12:01:47 +00:00
philipel
ee49f7087f Remove VideoEncoder::SetChannelParameters.
The SetChannelParameters function was used when WebRTC supported decoding
with errors, which we no longer do.

This cleanup CL is related to the work tracked by 9946.

Bug: webrtc:9946
Change-Id: Id2d5ed23031388f890c42651bfbe5f79eda701e5
Reviewed-on: https://webrtc-review.googlesource.com/c/108861
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25505}
2018-11-05 17:37:07 +00:00
Yves Gerey
0414040724 Fix race condition for SupportsFlexfecWithMultithreadedH264/0 test.
Guard FakeEncode.last_frame_info_ against concurrent access.

Bug: webrtc:9833
Change-Id: Idf36cee15307a64cd79d85f0f65914b516fc6590
Reviewed-on: https://webrtc-review.googlesource.com/c/104500
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25057}
2018-10-09 07:29:08 +00:00
Ilya Nikolaevskiy
b0588e6368 Reland "Move FakeCodec to separate target and behave like real encoder."
Reland after fixes for ramp-up-tests

original reviewed on: https://webrtc-review.googlesource.com/95182

TBR=mbonadei@webrtc.org,ilnik@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,titovartem@webrtc.org

Bug: none
Change-Id: I4f9af0c98a0341dd4fadd5184bb85d7511efbdc0
Reviewed-on: https://webrtc-review.googlesource.com/96120
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24450}
2018-08-27 13:09:37 +00:00
Ilya Nikolaevskiy
8d92e8d323 Revert "Reland "Move FakeCodec to separate target and behave like real encoder.""
This reverts commit f2a8287cc5bbe982cc008d0550df83533623b780,
original reviewed on: https://webrtc-review.googlesource.com/95182

Reason for revert: Breaks ramp-up tests

TBR=mbonadei@webrtc.org,ilnik@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,titovartem@webrtc.org

Bug: none
Change-Id: I11ddf8619c33cf93825088fd293bcdf11e8cedab
Reviewed-on: https://webrtc-review.googlesource.com/96083
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24438}
2018-08-27 09:19:33 +00:00
Ilya Nikolaevskiy
f2a8287cc5 Reland "Move FakeCodec to separate target and behave like real encoder."
Add FakeVp8Encoder, change FakeEncoder to use BitrateAllocator for simulcast.

Change call_test to use VP8 payload name for simulcast tests.
This is reland after fixes for broken perf tests.

 
Original Reviewed-on: https://webrtc-review.googlesource.com/91861

Bug: none
Change-Id: I6999a499408787be43a74a26a16b7826a0814a7b
Reviewed-on: https://webrtc-review.googlesource.com/95182
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24383}
2018-08-22 09:48:32 +00:00
Ilya Nikolaevskiy
7d13a6e5b9 Revert "Move FakeCodec to separate target and behave like real encoder."
This reverts commit 223eba5f72b5228847eeebaaef1c4305a29e8b3d.

Reason for revert: Breaks perf tests and downstream projects.

Original change's description:
> Move FakeCodec to separate target and behave like real encoder.
> 
> Add FakeVp8Encoder, change FakeEncoder to use BitrateAllocator for simulcast.
> Change call_test to use VP8 payload name for simulcast tests.
> 
> Bug: none
> Change-Id: I5a34c52e66bbd6c05859729ed14ae87ac26b5969
> Reviewed-on: https://webrtc-review.googlesource.com/91861
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24359}

TBR=mbonadei@webrtc.org,ilnik@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org

Change-Id: I602acecb3f340cc8d737ca074bf52496593419c8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/95181
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24365}
2018-08-21 15:20:32 +00:00
Ilya Nikolaevskiy
223eba5f72 Move FakeCodec to separate target and behave like real encoder.
Add FakeVp8Encoder, change FakeEncoder to use BitrateAllocator for simulcast.
Change call_test to use VP8 payload name for simulcast tests.

Bug: none
Change-Id: I5a34c52e66bbd6c05859729ed14ae87ac26b5969
Reviewed-on: https://webrtc-review.googlesource.com/91861
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24359}
2018-08-21 13:44:32 +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
Erik Språng
566124a6df Move BitrateAllocation to api/ and rename it VideoBitrateAllocation
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.

Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.

Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
2018-04-23 15:31:27 +00:00
Niels Möller
1e06289cdb Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.
Both macros do the same thing, as wrappers for
__attribute__((guarded_by)), and more names for the same thing doesn't
add to clarity.

Bug: none
Change-Id: Iaaf7b21dbf3345ee90fee22c39b636823d195eb0
Reviewed-on: https://webrtc-review.googlesource.com/48361
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21929}
2018-02-07 10:07:28 +00:00
Mirko Bonadei
7120742701 Adding NOLINT for typedefs.h and common_types.h
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.

The cpplint complaint is:
Include the directory when naming .h files  [build/include] [4]

This CL disables the error but we have to remove these two headers
from the root directory.

NOPRESUBMIT=true

Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
2017-09-15 13:03:51 +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