Commit Graph

128 Commits

Author SHA1 Message Date
81de14ffd4 Reduce implementations of CallStatsObserver by 2.
Both of these implementations are added as observers by
VideoReceiveStream, which is also an observer.

Instead, VideoReceiveStream now forward the necessary information
when the stats are updated as is already done for |frame_buffer_|.

Fewer observers registered, simplifies the code a little bit,
reduces synchronization when registering/unregistering observers
and reduces the set of existing implementations.

(I'm changing call_stats.* a little bit, so this is a by-the-way change)

Bug: none
Change-Id: I2d40b80a60e5ca632293c5118ac3b1a35d48ea40
Reviewed-on: https://webrtc-review.googlesource.com/64524
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22594}
2018-03-26 08:37:49 +00:00
0fa82a60e9 Moved FrameKey to api/video/encoded_frame.h and renamed it to VideoLayerFrameId.
Since we want the VideoStreamDecoder to callback with the last
continuous frame we need to move the FrameKey into the public API.

Bug: webrtc:8909
Change-Id: I39634145d848b8163778e31a1e0d04d91f9bbeb8
Reviewed-on: https://webrtc-review.googlesource.com/60864
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22495}
2018-03-19 15:13:11 +00:00
2e1d784956 Delete the VideoCodec::plName string.
It holds the same information as codecType, but in different format.

Bug: webrtc:8830
Change-Id: Ia83e2dff4fd9a5ddb489501b7a1fe80759fa4218
Reviewed-on: https://webrtc-review.googlesource.com/56100
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22307}
2018-03-06 11:17:41 +00:00
0a9f6de446 Removed VCMTiming from RtpVideoStreamReceiver.
Bug: webrtc:8909
Change-Id: Ib42e4cc4c2252f04ea5f7d440352546d36d2899f
Reviewed-on: https://webrtc-review.googlesource.com/58740
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22224}
2018-02-28 11:31:51 +00:00
132e28e6aa Add thread checks to ReceiveStatisticsProxy that reflect
design comments.

Change-Id: I4dd2b0be006440500cc7744de0e952263531ccd2
Bug: webrtc:8929
Reviewed-on: https://webrtc-review.googlesource.com/54940
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22179}
2018-02-26 09:21:00 +00:00
fbf3bce431 Reland "Reduce locking in VideoReceiver and check the threading model."
This is a reland of c75f1e45093a8d5cc62937c7708b87aa5c5bf0b0.

Original change's description:
> Reduce locking in VideoReceiver and check the threading model.
>
> Note: This is a subset of code that was previously reviewed here:
>   - https://codereview.webrtc.org/2764573002/
>
> * Added two notification methods, DecoderThreadStarting() and DecoderThreadStopped()
>   * Allows us to establish a period when the decoder thread is not running and it is
>     safe to modify variables such as callbacks, that are only read when the decoder
>     thread is running.
>   * Allows us to DCHECK thread guarantees/correctness.
>   * Allows synchronizing callbacks from the module process thread and have them only
>     active while the decoder thread is running.
>   * The above, allows us to establish two modes for the thread,
>     single-threaded-mutable and multi-threaded-const.
>   * Using that knowledge, we can remove |receive_crit_| as well as locking for a
>     number of member variables.
> * Removed |VCMFrameBuffer _frameFromFile| (unused).
> * Clean up several of my TODOs
>
> Bug: webrtc:7361, chromium:695438
> Change-Id: Id0048ee9624f76103c088d02825eb5c0d6c8913c
> Reviewed-on: https://webrtc-review.googlesource.com/55000
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22133}

Bug: webrtc:7361, chromium:695438
Change-Id: I32e1dc6c62cb30ad96e6366106f39fe415de49f1
Tbr: philipel@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/56803
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22163}
2018-02-22 18:03:45 +00:00
e7c891f953 Renamed FrameObject to EncodedFrame.
The plan is to:
 1. Move FrameObject to api/video.
 2. Rename FrameObject to EncodedFrame.
 3. Move EncodedFrame out of the video_coding namespace.

This is the 2nd CL.

Bug: webrtc:8909
Change-Id: I5e76a0a3b306156b8bc1de67834b4adf14bebef9
Reviewed-on: https://webrtc-review.googlesource.com/56182
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22158}
2018-02-22 16:12:48 +00:00
c4f9824cee Revert "Reduce locking in VideoReceiver and check the threading model."
This reverts commit c75f1e45093a8d5cc62937c7708b87aa5c5bf0b0.

Reason for revert: Breaking internal project

Original change's description:
> Reduce locking in VideoReceiver and check the threading model.
> 
> Note: This is a subset of code that was previously reviewed here:
>   - https://codereview.webrtc.org/2764573002/
> 
> * Added two notification methods, DecoderThreadStarting() and DecoderThreadStopped()
>   * Allows us to establish a period when the decoder thread is not running and it is
>     safe to modify variables such as callbacks, that are only read when the decoder
>     thread is running.
>   * Allows us to DCHECK thread guarantees/correctness.
>   * Allows synchronizing callbacks from the module process thread and have them only
>     active while the decoder thread is running.
>   * The above, allows us to establish two modes for the thread,
>     single-threaded-mutable and multi-threaded-const.
>   * Using that knowledge, we can remove |receive_crit_| as well as locking for a
>     number of member variables.
> * Removed |VCMFrameBuffer _frameFromFile| (unused).
> * Clean up several of my TODOs
> 
> Bug: webrtc:7361, chromium:695438
> Change-Id: Id0048ee9624f76103c088d02825eb5c0d6c8913c
> Reviewed-on: https://webrtc-review.googlesource.com/55000
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22133}

TBR=tommi@webrtc.org,philipel@webrtc.org

Change-Id: I4d78e8b2c05b36e1a3f64cb38d652579b3a23f22
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7361, chromium:695438
Reviewed-on: https://webrtc-review.googlesource.com/56280
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22141}
2018-02-21 19:37:21 +00:00
c75f1e4509 Reduce locking in VideoReceiver and check the threading model.
Note: This is a subset of code that was previously reviewed here:
  - https://codereview.webrtc.org/2764573002/

* Added two notification methods, DecoderThreadStarting() and DecoderThreadStopped()
  * Allows us to establish a period when the decoder thread is not running and it is
    safe to modify variables such as callbacks, that are only read when the decoder
    thread is running.
  * Allows us to DCHECK thread guarantees/correctness.
  * Allows synchronizing callbacks from the module process thread and have them only
    active while the decoder thread is running.
  * The above, allows us to establish two modes for the thread,
    single-threaded-mutable and multi-threaded-const.
  * Using that knowledge, we can remove |receive_crit_| as well as locking for a
    number of member variables.
* Removed |VCMFrameBuffer _frameFromFile| (unused).
* Clean up several of my TODOs

Bug: webrtc:7361, chromium:695438
Change-Id: Id0048ee9624f76103c088d02825eb5c0d6c8913c
Reviewed-on: https://webrtc-review.googlesource.com/55000
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22133}
2018-02-21 15:44:05 +00:00
d397a0d46e Add dropped frames metric on the receive side
Reported to UMA and logged for at the end of the call.

Bug: webrtc:8355
Change-Id: I4ef31bf9e55feaba9cf28be5cb4fcfae929c7179
Reviewed-on: https://webrtc-review.googlesource.com/53760
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22132}
2018-02-21 15:34:25 +00:00
d7ae3c34e5 Reland "Rename stereo video codec to multiplex"
This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.

Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}

TBR=niklas.enbom@webrtc.org

Bug: webrtc:7671
Change-Id: I6f38dc46126f279f334d52b56339b40acdc30511
Reviewed-on: https://webrtc-review.googlesource.com/45820
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21794}
2018-01-29 20:37:59 +00:00
1204448a68 Revert "Reland "Rename stereo video codec to multiplex""
This reverts commit 4954a77cf81e6793245f52d485834acd3e6eab1c.

Reason for revert: Breaks downstream build which was depending on the name "kVideoCodecStereo". Will need to do some sort of trickery to make this change without breaking the relevant code. Sorry. :(

Original change's description:
> Reland "Rename stereo video codec to multiplex"
> 
> This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
> This was reverted because of breaking internal build. I contacted sheriff
> and looked at logs but cannot find anything related to this CL. This was landed
> with #3850 build which caused exception, but 3847-3855 seem to all have failed.
> I am relanding to see if it will work this time or it will give some related
> error message that can guide me.
> 
> Original change's description:
> > Rename stereo video codec to multiplex
> >
> > This CL only does the rename from"stereo" to multiplex". With this we have a
> > better name that doesn't clash with audio's usage of stereo.
> >
> > Bug: webrtc:7671
> > Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> > Reviewed-on: https://webrtc-review.googlesource.com/43242
> > Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21769}
> 
> TBR=niklas.enbom@webrtc.org
> 
> Bug: webrtc:7671
> Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
> Reviewed-on: https://webrtc-review.googlesource.com/44520
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21780}

TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org

Change-Id: I0a71327c2ddfdd030b1e058cd6a41b1689836719
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44621
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21783}
2018-01-27 00:45:20 +00:00
4954a77cf8 Reland "Rename stereo video codec to multiplex"
This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
This was reverted because of breaking internal build. I contacted sheriff
and looked at logs but cannot find anything related to this CL. This was landed
with #3850 build which caused exception, but 3847-3855 seem to all have failed.
I am relanding to see if it will work this time or it will give some related
error message that can guide me.

Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}

TBR=niklas.enbom@webrtc.org

Bug: webrtc:7671
Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
Reviewed-on: https://webrtc-review.googlesource.com/44520
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21780}
2018-01-26 21:11:54 +00:00
6bc7bb659e Revert "Rename stereo video codec to multiplex"
This reverts commit bbdabe50db0cf09f6007dda12a6476dc4602b174.

Reason for revert: This breaks the internal build.

Original change's description:
> Rename stereo video codec to multiplex
> 
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
> 
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}

TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org

Change-Id: Icf019cb09e07de45821d31d7d8ea7707d01346ee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44360
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21774}
2018-01-26 12:44:54 +00:00
bbdabe50db Rename stereo video codec to multiplex
This CL only does the rename from"stereo" to multiplex". With this we have a
better name that doesn't clash with audio's usage of stereo.

Bug: webrtc:7671
Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
Reviewed-on: https://webrtc-review.googlesource.com/43242
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21769}
2018-01-25 23:16:04 +00:00
8e07c134ab Optional: Use nullopt and implicit construction in /video
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

Bug: None
Change-Id: Ie622c215e06956d8d5629733c76f531b7af45012
Reviewed-on: https://webrtc-review.googlesource.com/23568
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21535}
2018-01-09 15:14:10 +00:00
a9329dbae2 Bug Fix: Peers Cannot Communicate If One With Stereo Codec, One Not
When Chromium hooks up with the stereo codec, then it has difficulty
communicating with a google chrome without stereo codec. By design, we
do allow codec choice for the standalone codecs, but the problem is
that we do not handle the payload correctly, and thus the existence
of stereo codec will remove the payload registry of the standalone
version of its associated codec. (For example, stereo codec on top of
VP9 will remove the payload registry of standalone VP9 codec.)

This CL fixes the issue. When generating payload data, we should use
"stereo" as payload name, instead of its associated codecs.


Bug: webrtc:8657
Change-Id: I9e0b54de6bd41d370b9353f9553c998e4049789f
Reviewed-on: https://webrtc-review.googlesource.com/33122
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21523}
2018-01-08 17:53:39 +00:00
be214a26f8 Move videosinkinterface.h to common_video to solve a circular dep.
I updated some dependency enforcement rules to allow examples and pc
to depend on common_video. I reckoned depending on common_video is
not controversial when they already dependend on media/base, which
is a lower-level abstraction.

Bug: webrtc:6828
Change-Id: I77dbeb10187b4e70dda1d873a29994fa76070758
Reviewed-on: https://webrtc-review.googlesource.com/34187
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21495}
2018-01-04 13:19:49 +00:00
49b46e0085 Added WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME decoder return code.
Bug: None
Change-Id: I71e1d42c92273dc4ce3f5f5e7052615d68e65a38
Reviewed-on: https://webrtc-review.googlesource.com/31860
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21363}
2017-12-19 14:57:41 +00:00
1610f94ee3 Don't cast picture ids (of type int64_t) to int.
Also cleaned up a bit in RtpFrameReferenceFinder.

Bug: chromium:762556
Change-Id: Ib08d2e7ce4b146b359ce9ba823f3aa15776c71bc
Reviewed-on: https://webrtc-review.googlesource.com/32301
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21282}
2017-12-14 14:22:13 +00:00
0a37547033 Add optional stereo codec to SDP negotiation
- Defines stereo codec case, similar to RTX, that adds stereo codec to the SDP
negotiation. The underlying codec's payload type is similarly defined by "apt".
- If this negotiation is successful, codec name is included in sdp line via
"acn".
- Adds codec setting initializers for these specific stereo cases.
- Introduces new Stereo*Factory classes as optional convenience wrappers that
inserts stereo codec to the existing set of supported codecs on demand.

This CL is the step 5 for adding alpha channel support over the wire in webrtc.
Design Doc: https://goo.gl/sFeSUT

Bug: webrtc:7671
Change-Id: Ie12c56c8fcf7934e216135d73af33adec5248f76
Reviewed-on: https://webrtc-review.googlesource.com/22901
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21210}
2017-12-11 16:30:06 +00:00
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros.
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00
c3fa8e1ce7 New method RtpReceiver::GetLatestTimestamps.
The two timestamps, rtp time and corresponding system time, are always
used together, for audio/video sync. The new method reads both
timestamps, without releasing a lock in between. Ensures that the
caller gets values corresponding to the same packet.

Bug: webrtc:7135
Change-Id: I25bdcbe9ad620016bfad39841b339c266efade14
Reviewed-on: https://webrtc-review.googlesource.com/4062
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20120}
2017-10-03 16:14:29 +00:00
48462b6ef7 Continuously request keyframes if decoding does not recover.
This is a workaround until downstream projects have been fixed.

BUG=webrtc:8220

Review-Url: https://codereview.webrtc.org/3017613002
Cr-Commit-Position: refs/heads/master@{#19966}
2017-09-26 09:54:58 +00:00
e21be1db4c Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
Reason for revert:
Fixes has landed.

Original issue's description:
> Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
>
> Reason for revert:
> We are not certain this is the behavior we want.
>
> Original issue's description:
> > Fix the video buffer size should take rtt into consideration
> >
> > BUG=webrtc:8010
> >
> > Review-Url: https://codereview.webrtc.org/2980413002
> > Cr-Commit-Position: refs/heads/master@{#19285}
> > Committed: f1e08d0b58
>
> TBR=sprang@webrtc.org,gustavogb@gmail.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:8010
>
> Review-Url: https://codereview.webrtc.org/3002033002
> Cr-Commit-Position: refs/heads/master@{#19442}
> Committed: bdbc8895f3

TBR=sprang@webrtc.org,gustavogb@gmail.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8010

Review-Url: https://codereview.webrtc.org/3016633002
Cr-Commit-Position: refs/heads/master@{#19944}
2017-09-25 13:37:12 +00:00
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
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
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