Commit Graph

42 Commits

Author SHA1 Message Date
e1c707c40f Remove unused incomplete_frame argument from JitterEstimator.
Bug: webrtc:14151
Change-Id: I6764315f0c10b304f50e4639a3e49e4ed013c41e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267842
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37443}
2022-07-05 12:53:13 +00:00
10944e61ac Move JitterEstimator into timing sub-folder
Bug: webrtc:14111
Change-Id: Ic5c66f35e3cedac7a328fbb6613e5cdf13a8005e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263582
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37000}
2022-05-25 09:15:29 +00:00
2377226851 Start moving timing helper classes into timing/ sub-folder.
Putting these classes in a sub folder increases
structure and clarifies that they are used as
helper classes. Affected classes in this change:
  * CodecTimer
  * InterFrameDelay
  * RttFilter
VCMTiming will be moved in a separate CL.

Additional changes:
  * Remove VCM prefix from class names.
  * Introduce granular BUILD.gn targets.
  * Update some includes.

Bug: webrtc:14111
Change-Id: Ia75128aa955a819033b97d4784cb61904de7230b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262960
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36975}
2022-05-23 13:43:40 +00:00
e62c2f2c77 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
rename WebRtcKeyValueConfig to FieldTrialsView

Bug: webrtc:10335
Change-Id: If725bd498c4c3daf144bee638230fa089fdde833
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 10:14:00 +00:00
e02f9eedb3 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 10/inf
This patch takes a stab at modules/video_coding,
but reaches only about half.

Bug: webrtc:10335
Change-Id: I0d47d0468b818145470c51ae4e8e75ff58d499ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256112
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36335}
2022-03-25 12:35:36 +00:00
e9126c18bf Migrate VCMInterFrameDelay to use Time units
Additionally,
* Moved to its own GN target.
* Added unittests.
* Removed unused variable `_zeroWallClock`.
* Renamed variables to match style guide.
* Moved fields _dTS and _wrapArounds to variables.

Change-Id: I7aa8b8dec55abab49ceabe838dabf2a7e13d685d
Bug: webrtc:13756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253580
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36147}
2022-03-08 09:05:12 +00:00
13e42a88df Use TimeDelta and Timestamp in VCMJitterEstimator
* Uses DataSize to represent incoming and outgoing bytes.
* Puts units into doubles as they enter the Kalman filter
* Moved to its own GN target.

Change-Id: I1e7d5486a00a7158d418f553a6c77f9dd56bf3c2
Bug: webrtc:13756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253121
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36143}
2022-03-07 14:58:22 +00:00
d3251968d1 Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.

Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
2021-11-15 21:44:59 +00:00
dcd7fc7ea8 Use backticks not vertical bars to denote variables in comments for /modules/video_coding
Bug: webrtc:12338
Change-Id: Ia8a9adea291d594e4f59a6a1203a7bfb0758adac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227165
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34684}
2021-08-09 15:26:22 +00:00
190244bb59 Remove all #include <assert.h>/<cassert> and usage in Obj-C code.
This CL completes the removal of assert() and relative headers from
the codebase (excluded
//examples/objc/AppRTCMobile/third_party/SocketRocket which is in a
third_party sub-directory).

Bug: webrtc:6779
Change-Id: I93ed57168d2c0e011626873d66529488c5f484f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225546
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34528}
2021-07-22 14:00:26 +00:00
25ab3228f3 Replace assert() with RTC_DCHECK().
CL partially auto-generated with:

git grep -l "\bassert(" | grep "\.[c|h]" | \
  xargs sed -i 's/\bassert(/RTC_DCHECK(/g'

And with:

git grep -l "RTC_DCHECK(false)" |  \
  xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'

With some manual changes to include "rtc_base/checks.h" where
needed.

A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.

The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.

This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).

Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34442}
2021-07-09 07:49:43 +00:00
1b0d5437c9 Removed _completeFrame since we never allow incomplete frames.
In the old jitter buffer the two VCMVideoProtection modes |kProtectionNone| and |kProtectionFEC| could be set on the jitter buffer for it to not wait for NACK and instead generate incomplete frames. This has not been possible for a long time.

Bug: webrtc:9378, webrtc:7408
Change-Id: I0a2d3ec34d721126c1128306d5fad88314f8d59f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32513}
2020-10-28 16:00:27 +00:00
6deec38ede Migrate modules/video_coding to webrtc::Mutex.
Bug: webrtc:11567
Change-Id: I8023fbe7595f7ba8ae7c7db3583fc2e560ec3df2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178803
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31644}
2020-07-07 11:41:21 +00:00
2eecfc1f9a Trim dependencies in modules/video_coding/
And move jitter_buffer_common.h to the legacy build target.

Bug: None
Change-Id: I986649f2f0773cadfa7dd9c8b533af7ecf01f3a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152382
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29150}
2019-09-11 09:31:10 +00:00
fe407b7a1d Move code related to VideoCodingModule to its own build target
The new target, modules/video_coding:video_coding_legacy, is not
depended upon by any webrtc non-test code.

Bug: webrtc:7408
Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29133}
2019-09-10 12:34:38 +00:00
e0f370471a Add cap to video jitter buffer size/latency in experiment branches only.
Bug: webrtc:10664
Change-Id: I03762c8b318f26f2689e89545aa8cc8e5b4a4329
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138081
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28155}
2019-06-04 15:50:27 +00:00
4ed7e511f6 Revert "Add ability to cap the video jitter estimate to a max value."
This reverts commit a8ae407a480a2a9982eecf9e3a9b10da5373cd9a.

Reason for revert: This CL incorrectly affects non-experiment branch.  A new CL affecting only the experiment will be uploaded.

Original change's description:
> Add ability to cap the video jitter estimate to a max value.
>
> Bug: webrtc:10572
> Change-Id: I21112824dc02afa71db61bb8c2f02723e8b325b6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133963
> Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27744}

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

Bug: webrtc:10572
Change-Id: I4af334168ca70ecfae7fd18fc7c852819a98d866
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138063
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28025}
2019-05-22 15:07:33 +00:00
a8ae407a48 Add ability to cap the video jitter estimate to a max value.
Bug: webrtc:10572
Change-Id: I21112824dc02afa71db61bb8c2f02723e8b325b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133963
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27744}
2019-04-24 16:03:15 +00:00
abbc50e9b2 Move frame_type member from RtpDepacketizer::ParsedPayload to RTPVideoHeader
The latter is also a member of the former. This cleanup is also
a preparation for dropping WebRtcRTPHeader::frameType (or deleting
WebRtcRTPHeader right away), now that it's a video-specific member.


Tbr: kwiberg@webrtc.org # Comment change in modules/include/
Bug: None
Change-Id: I5c1f3f981f0d750713fc9b9b145278150fe32b5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133024
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27740}
2019-04-24 13:13:04 +00:00
50150a18f4 Delete VCMJitterBuffer::UpdateRtt and related unused code
Bug: webrtc:7408
Change-Id: Ic4bd745ed36dd7f7d5eec6531e05204035a545fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133188
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27648}
2019-04-16 15:13:13 +00:00
7eb4248b02 Delete unused class Vp9SsMap
Appears unused since https://codereview.webrtc.org/1426813002

Bug: webrtc:7408
Change-Id: I13b8c35548314705eaf31ff2366636f358fef718
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133203
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27645}
2019-04-16 14:36:55 +00:00
691f62cfaa Make VCMJitterBuffer::SetNackMode(kNack, -1, -1) the only mode
Bug: webrtc:7408
Change-Id: I9d9e4f97c7705b42c9575167710a3e79781b83e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130220
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27568}
2019-04-11 14:29:42 +00:00
10aac06886 Delete VCMJitterBuffer::IncomingRateStatistics
Bug: webrtc:7408
Change-Id: I2da20bca053bb757957aec56e1fc7f7f915a0b55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130102
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27378}
2019-04-01 07:43:28 +00:00
db64d99130 Delete unused constructor arguments from VCMJitterBuffer
The NackSender* and KeyFrameRequestSender* arguments were unused.

Bug: webrtc:7408
Change-Id: Iab3b443373953e88c48867d031cb114b4891ab78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130503
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27377}
2019-04-01 07:42:23 +00:00
20393ee9b7 Delete all stats-related logic from VCMJitterBuffer.
Bug: webrtc:7408
Change-Id: I0347746f8c6cd2d8fb4b2daa61d4e3ef8f550b77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129930
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27338}
2019-03-28 12:55:33 +00:00
b8fffa1ad8 Delete method VCMJitterBuffer::RegisterStatsCallback
Unused in the legacy VideoCodingModule api since
https://webrtc-review.googlesource.com/c/src/+/62101/
and unused by the VideoReceiveStream code path since
https://webrtc-review.googlesource.com/c/src/+/128870

Bug: webrtc:7408
Change-Id: I800dba08a6e0e8f5de6169241d217bd5e8e5d0de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129961
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27319}
2019-03-27 16:12:38 +00:00
8f7ce222e7 Make VideoFrameType an enum class, and move to separate file and target
Bug: webrtc:5876, webrtc:6883
Change-Id: I1435cfa9e8e54c4ba2978261048ff3fbb993ce0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126225
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27239}
2019-03-22 12:44:51 +00:00
375b346b30 Delete all logic related to VCMDecodeErrorMode
Bug: webrtc:8064
Change-Id: I345f342a314d88390fff8b305b121076b45a51e8
Reviewed-on: https://webrtc-review.googlesource.com/c/116692
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26283}
2019-01-16 15:41:07 +00:00
ebad1770ab Include event_wrapper.h only where used.
It's currently used only by the VCMJitterBuffer and VCMReceiver
classes. Injection is needed by the VCMReceiverTimingTest test, which
defines a subclass(!) of EventWrapper.

Bug: webrtc:3380
Change-Id: I765be0ceac58e941928319cc426ba49f1cbdc5fa
Reviewed-on: https://webrtc-review.googlesource.com/c/113002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25893}
2018-12-04 14:50:18 +00:00
f0eee0087f Move size() method to EncodedImage base class
Deleted from subclass video_coding::EncodedFrame. Also delete Length
and SetLength methods on the intermediate class
video_coding::VCMEncodedFrame.

Bug: webrtc:9378
Change-Id: I3c90b14735f622f50b2f403f79072e22fc025d11
Reviewed-on: https://webrtc-review.googlesource.com/c/112131
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25838}
2018-11-29 13:44:47 +00:00
2377588c82 Add accessor methods for RTP timestamp of EncodedImage.
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.

Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
2018-08-21 09:15:51 +00:00
29d8846df9 Remove RTPVideoHeader::vp9() accessors.
TBR=stefan@webrtc.org

Bug: none
Change-Id: Ia2f728ea3377754a16a0b081e25c4479fe211b3e
Reviewed-on: https://webrtc-review.googlesource.com/93024
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24243}
2018-08-09 10:53:28 +00:00
dbdb3a0079 Refactoring PayloadRouter.
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
  VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
  of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
  renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.

Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
2018-07-17 14:46:15 +00:00
5ab67a5d71 Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
This CL is in preparation to change the RTPVideoTypeHeader into an absl::variant.

Bug: none
Change-Id: I1672d866df0395f3417d8e278cc67f017ab0ff98
Reviewed-on: https://webrtc-review.googlesource.com/87261
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23856}
2018-07-05 14:29:07 +00:00
52f53d5419 Revert "Add Timestamp accessor methods to the EncodedImage class."
This reverts commit f34d467b03da4f20a1d036a20966fcad43d2433f.

Reason for revert: Seems to break downstream project.

Original change's description:
> Add Timestamp accessor methods to the EncodedImage class.
> 
> Bug: webrtc:9378
> Change-Id: I59bf14f631f92f0f4e05f60d4af25641a23a53f9
> Reviewed-on: https://webrtc-review.googlesource.com/82100
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23734}

TBR=brandtr@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,philipel@webrtc.org

Change-Id: I3aa0c0119426886bc583c918aae862eb7f4b6b63
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9378
Reviewed-on: https://webrtc-review.googlesource.com/85600
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23739}
2018-06-26 11:52:45 +00:00
f34d467b03 Add Timestamp accessor methods to the EncodedImage class.
Bug: webrtc:9378
Change-Id: I59bf14f631f92f0f4e05f60d4af25641a23a53f9
Reviewed-on: https://webrtc-review.googlesource.com/82100
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23734}
2018-06-26 09:40:18 +00:00
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
9633cff81a Remove "webrtc_rtp" traces.
They have been disabled by default for years, and should have been made redundant by the event logs.

Bug: webrtc:8982
Change-Id: I491923cbc93378d28f5166d24756b335619d9c12
Reviewed-on: https://webrtc-review.googlesource.com/82800
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23598}
2018-06-13 14:46:24 +00:00
80ba333fc5 Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix
Bug: chromium:805946
Change-Id: Ibb5dce9af27d0e48c9aee6b0a860b6f62b3c76a0
Reviewed-on: https://webrtc-review.googlesource.com/46145
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21889}
2018-02-05 11:24:59 +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
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