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Author SHA1 Message Date
Artem Titov
0e61fdd27c Use backticks not vertical bars to denote variables in comments for /api 
Bug: webrtc:12338
Change-Id: Ib97b2c3d64dbd895f261ffa76a2e885bd934a87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34554}
 2021-07-26 18:27:34 +00:00
Ilya Nikolaevskiy
c98aebbbef Change how alignment requirements are processed 
Software fallback wrapper now reports least common multiple of requirements
for two encoders.

SimulcastEncoderAdapter queries actual encoder before InitEncode call
and requests alignment for all layers if simulcast is not supported by
any of the encoders.

Bug: chromium:1084702
Change-Id: Iaed8190737125d447036b6c664b863be72556a5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225881
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34466}
 2021-07-13 16:49:13 +00:00
Åsa Persson
aab91c7b3a Remove temporal layer restriction for forced resolution based fallback. 
Bug: none
Change-Id: Id8d30b6759bc6d5a517d1363395d4495251b32fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205860
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33172}
 2021-02-05 07:31:54 +00:00
Jakob Ivarsson
a9961b3839 Allow temporal based switch if temporal layers are undefined. 
Bug: webrtc:11324
Change-Id: Iee4717f453bb9883683d752832fbc7bf999a96c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193704
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32630}
 2020-11-18 10:59:22 +00:00
Karl Wiberg
c95b939667 Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED() 
And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.

Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}
 2020-11-09 10:47:55 +00:00
Fabien Vallée
f8b5bfeaf2 Fix "control reaches end of non-void function" warnings 
"warning: control reaches end of non-void function [-Wreturn-type]"
Reported by gcc (8.3)

In all the reported cases, the end of function is never actually
reached. Add RTC_CHECK(false) to ensure the compiler is aware that
this path is a dead-end.

Bug: webrtc:12008
Change-Id: I7f816fde3d1897ed2774057c7e05da66e1895e60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189784
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fabien VALLÉE <fabien.vallee@netgem.com>
Cr-Commit-Position: refs/heads/master@{#32503}
 2020-10-27 10:22:23 +00:00
Evan Shrubsole
5089a8ea14 Use VideoFrameBuffer::Scale in encoder wrappers 
This sincludes the SimulcastEncoderAdapter and the
VideoEncoderSoftwareFallbackWrapper. This avoids converting
the frame when that is not needed.

Bug: webrtc:11976
Change-Id: I686725ecfb79c3b8d87d587a907da1602483bfe8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187343
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32389}
 2020-10-13 12:33:42 +00:00
Mirko Bonadei
06d3559b79 Replace std::string::find() == 0 with absl::StartsWith (part 2). 
This CL has been generated using clang-tidy [1] except for changes to
BUILD.gn files.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/abseil-string-find-startswith.html

Bug: None
Change-Id: Ibf75601065a53bde28623b8eef57bec067235640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172586
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30984}
 2020-04-02 14:38:30 +00:00
Evan Shrubsole
546a9e4350 Scale native frames when doing a SW codec fallback 
If the incoming frame is a native frame but the native encoder fails,
we should ensure the fallback encoder can handle the native frame. If
not then the native frame should be scaled and converted.

Bug: webrtc:11346
Change-Id: I692350dc69b5ce2db7ba5ee98d28f94cb12054cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168345
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30504}
 2020-02-12 08:55:51 +00:00
Erik Språng
261f792f83 Allow software fallback on lowest simulcast stream for temporal support 
Bug: webrtc:11324
Change-Id: Ie505be0cda74c0444065d86c3727671c62bd4842
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167527
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30437}
 2020-01-31 16:44:47 +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
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
65764e4ed7 Add missing overrides in VideoEncoder proxies/adapters 
Add:
1. OnPacketLossRateUpdate
2. OnRttUpdate
3. OnLossNotification

Add them to:
1. VideoEncoderSoftwareFallbackWrapper
2. SimulcastEncoderAdapter
3. MultiplexEncoderAdapter

Bug: None
Change-Id: I4b0799f7d8c19211741f48da87106daccd39af95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144030
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28423}
 2019-06-28 22:45:53 +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 11dfff0878

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 11dfff0878.

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 7ac0d5f348

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 7ac0d5f348.

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
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
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
c8d2e73ed0 Delete CodecSpecificInfo argument from VideoEncoder::Encode 
Bug: webrtc:10379
Change-Id: If9f92eb1e5891df284881082c53f0b1db1c26a38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125900
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26992}
 2019-03-06 14:01:31 +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
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
Erik Språng
e2fd86a79c Move encoder metadata into EncoderInfo struct. 
This deprecates the following methods in VideoEncoder:
  virtual ScalingSettings GetScalingSettings() const;
  virtual bool SupportsNativeHandle() const;
  virtual const char* ImplementationName() const;

Though they are not marked RTC_DEPRECATED since we still want to call
them from within the default GetEncoderInfo() until downstream
projects have been updated.

Furthmore, implementation name is changed from const char* to
std:string, which prevents some lifetime issues with dynamic encoder
names, and CodecSpecificInfo.codec_name is removed in favor of getting
the implementation name via GetEncoderInfo().

This CL removes calls to these deprecated methods, follow-ups will also
remove implementations of the methods and replace them with new
GetEncoderInfo() substitutions.

Bug: webrtc:9890
Change-Id: I6fd6e531480c0b952f53dbd5105e0b0adc3e3b0c
Reviewed-on: https://webrtc-review.googlesource.com/c/106905
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25351}
 2018-10-25 08:51:53 +00:00
Oleh Prypin
a1d9ca47f9 Revert "Add ability to specify if rate controller of video encoder is trusted." 
This reverts commit 3e335d1423.

Reason for revert: breaks downstream project

Original change's description:
> Add ability to specify if rate controller of video encoder is trusted.
>
> If rate controller is trusted, we disable the frame dropper in the
> media optimization module.
>
> Bug: webrtc:9722
> Change-Id: I821f21fd74a400ee9d5aa3f6b42d4e569033acbe
> Reviewed-on: https://webrtc-review.googlesource.com/c/105020
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25107}

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

Change-Id: Ifdb0aae684894854a184ec1e7423a7c62e7ba237
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9722
Reviewed-on: https://webrtc-review.googlesource.com/c/105360
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25117}
 2018-10-11 15:37:40 +00:00
Erik Språng
3e335d1423 Add ability to specify if rate controller of video encoder is trusted. 
If rate controller is trusted, we disable the frame dropper in the
media optimization module.

Bug: webrtc:9722
Change-Id: I821f21fd74a400ee9d5aa3f6b42d4e569033acbe
Reviewed-on: https://webrtc-review.googlesource.com/c/105020
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25107}
 2018-10-11 09:07:34 +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
Anders Carlsson
dd3e0ab2bf Make rtc_software_fallback_wrappers target visible. 
Need to depend on them from Chromium.

Bug: webrtc:7925
Change-Id: Iea1bb3b937c602920bfd87f885c87c790ac7bc17
Reviewed-on: https://webrtc-review.googlesource.com/82061
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23580}
 2018-06-12 12:51:34 +00:00
Renamed from media/engine/videoencodersoftwarefallbackwrapper.cc (Browse further)