Commit Graph

25 Commits

Author SHA1 Message Date
a376518817 Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex.
Also migrates test/ partly.

Bug: webrtc:11567
Change-Id: If5b2eae65c5f297f364b6e3c67f94946a09b4a96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178862
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31672}
2020-07-08 12:21:08 +00:00
a827a30bb7 Revert "Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex."
This reverts commit 0eba415fb40cc4e3958546a8ee53c698940df0a1.

Reason for revert: previously unknown lock recursion occurring downstream.

Original change's description:
> Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex.
> 
> Also migrates test/ partly.
> 
> Bug: webrtc:11567
> Change-Id: I4203919615c087e5faca3b2fa1d54cba9f171e07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178813
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31653}

TBR=sprang@webrtc.org,handellm@webrtc.org

Change-Id: I13f337e0de5b8f0eb19deb57cb5623444460ec4d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178842
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31656}
2020-07-07 20:46:48 +00:00
0eba415fb4 Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex.
Also migrates test/ partly.

Bug: webrtc:11567
Change-Id: I4203919615c087e5faca3b2fa1d54cba9f171e07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178813
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31653}
2020-07-07 18:01:44 +00:00
ee3d995091 New class VideoReceiver2, a trimmed-down vcm::VideoReceiver
The vcm::VideoReceiver class is used by both VideoReceiveStream and
the legacy api VideoCodingModule. They have different requirements,
since the latter uses the old jitterbuffer and runs the code on a
ProcessThread.

By making a copy and trimming it down to what's actually used by
VideoReceiveStream, we can drop the dependency on the old
jitterbuffer, without breaking the legacy api. This should also make
it easier to do follow-up refactorings to trim down the class further,
and ultimately remove it.

Bug: webrtc:7408
Change-Id: Iec8a167fe5d0425114b0b67a5b4c2fd5fc4fa150
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151910
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29108}
2019-09-09 11:23:54 +00:00
0c141c591a Fix frames dropped statistics
The |frames_dropped| statistics contain not only frames that are dropped
but also frames that are in internal queues. This CL changes that so
that |frames_dropped| only contains frames that are dropped.

Bug: chromium:990317
Change-Id: If222568501b277a75bc514661c4f8f861b56aaed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150111
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28968}
2019-08-27 07:43:01 +00:00
bfd343b9be Add totalDecodeTime to RTCInboundRTPStreamStats
Pull request to WebRTC stats specification:
https://github.com/w3c/webrtc-stats/pull/450

Bug: webrtc:10775
Change-Id: Id032cb324724329fee284ebc84595b9c39208ab8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144035
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28440}
2019-07-02 08:28:06 +00:00
2317c5ee0a Delete method DecodedImageCallback::ReceivedDecodedReferenceFrame
The code invoking it was deleted in
https://codereview.webrtc.org/2753783002

Tbr: kwiberg@webrtc.org # Change to mock class in api/test
Bug: webrtc:7408
Change-Id: I576d7aacd7dc60e42a05d2ea837fddf16594e685
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133348
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27680}
2019-04-18 08:14:40 +00:00
c29fa1bf67 Drop usage of legacy VCMPacketRequestCallback
It's called from VideoReceiver::Process, which we no longer use.

Bug: webrtc:7408
Change-Id: I2ce5b7a07437e110d20d04aa159dddf245504abe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133189
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27657}
2019-04-17 10:14:06 +00:00
4168437782 Remove unused callbacks from VideoStreamDecoder
VCMReceiveStatisticsCallback originates in the old jitter buffer, and
is no longer used.

VCMFrameTypeCallback originates in VideoReceiver::RequestKeyFrame,
which is called from OncomingPacket, Process, Decode(uint16_t
maxWaitTimeMs), all of which are unused by VideoReceiveStream.

So delete the code to wire them up via VideoStreamDecoder.

Bug: webrtc:7408
Change-Id: I173bc94eb32f2641f943c125083db038c3bcaeb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128870
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@{#27277}
2019-03-26 09:25:43 +00:00
9b0b1e0063 Delete unused method VCMReceiveStatisticsCallback::OnReceiveRatesUpdated
Only interesting call deleted in cl
https://codereview.webrtc.org/2704183002.

Move call to QualitySample (used for bad call detection) to
OnRenderedFrame

Bug: webrtc:7408
Change-Id: I0e9ae2ed62fe19a282377cb840e38bd2aae8f3e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128768
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27243}
2019-03-22 14:56:18 +00:00
f24908e611 Delete unused enum StreamType and kViEStreamType* values.
Unused since https://codereview.webrtc.org/1693553002
Also drop an unneeded include of video_stream_decoder.h.

Bug: webrtc:7408
Change-Id: I249ecfe41b55b59abbd2e880ef144d64f130b0b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128767
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27242}
2019-03-22 14:55:14 +00:00
133b307c7d Delete method VideoStreamDecoder::UpdateRtt
It's routed to the old jitter buffer, which is instantiated but
unused.

Bug: webrtc:7408
Change-Id: I8fe2d8a2bfa6c2119945cc294d17982af4f830cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127541
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27100}
2019-03-13 14:57:20 +00:00
d970807e0c Remove rtc_base/scoped_ref_ptr.h.
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.

More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.

Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:29:58 +00:00
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
8fdcac3f06 Remove clang:find_bad_constructs suppression from call:call.
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.

Bug: webrtc:9251, webrtc:163
Change-Id: I74cb86c29cebb69dd22083718f1446f18f705cd4
Reviewed-on: https://webrtc-review.googlesource.com/95883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24483}
2018-08-29 11:57:00 +00:00
a12c42a6b2 Delete root header file typedef.h.
Usage replaced with stdint.h, rtc_base/system/arch.h and
rtc_base/system/unused.h, as appropriate.

Bug: webrtc:6854
Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18
Reviewed-on: https://webrtc-review.googlesource.com/90249
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 14:59:26 +00:00
b9b146c9fe Replace rtc::Optional with absl::optional in audio, call and video
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
2018-06-15 12:09:49 +00:00
4c8811b255 Delete some obsolete forward declarations
Bug: None
Change-Id: I3a9b59bf3dd63c206854ab949cf2d606046182c9
Reviewed-on: https://webrtc-review.googlesource.com/77427
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23292}
2018-05-18 07:29:25 +00:00
c6ce9c5938 New file api/video/BUILD.gn
Build targets involving files under api/video/ are moved into this
file, from api/BUILD.gn. In addition, drop "_api" part of target
names, and move the header file api/videosinkinterface.h to
api/video/video_sink_interface.h.

Bug: webrtc:9253
Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155
Reviewed-on: https://webrtc-review.googlesource.com/75500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-14 06:57:38 +00:00
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
6b28e20257 Remove unused 'last_rtt_ms_' variable from vcm::VideoStreamDecoder.
The effect of this is one less lock being grabbed when updating rtt stats.

(Using NoTry=True since the mac_asan and mac_rel bots are acting up)

Bug: None
Change-Id: I7dac4031573aa54550068a7039e5c2d3054cb50b
NoTry: true
Reviewed-on: https://webrtc-review.googlesource.com/64522
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22593}
2018-03-25 20:08:58 +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
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