This CL mainly updates the FEC code to use more C++11 features and
to be more in line with the style guide. These changes should
have no impact on the functionality provided by the FEC.
Summary of style fixes:
- Use range-based for loops, where applicable.
- Use auto type deduction for iterator type names.
- Use RTC_DCHECK instead of assert.
- Rename FEC to Fec, where applicable.
- Update test_fec.cc to use variable_names rather than variableNames.
- Avoid redefining the PacketList types outside ForwardErrorCorrection.
Another minor change is that storage for the packet masks, as these
are generated, now is provided by a member variable, rather than
being dynamically allocated on every call to GenerateFec.
BUG=webrtc:5654
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2080553003 .
Cr-Commit-Position: refs/heads/master@{#13403}
The function uses a deprecated function and is unused, so delete it.
BUG=6027
NOTRY=true
Review-Url: https://codereview.webrtc.org/2114013002
Cr-Commit-Position: refs/heads/master@{#13400}
Everything calls GetOSVersionName() instead, so no need to expose that.
Since it uses deprecated functions, this will make it easier to rewrite
it in a future change.
BUG=webrtc:6027
NOTRY=true
Review-Url: https://codereview.webrtc.org/2117823002
Cr-Commit-Position: refs/heads/master@{#13399}
There was a fast path in PreprocessToAddData that would just use the
input timestamps if the input format was equal to the required format of
the encoder. This works well as long as the codec never changes. If we
are first doing resampling (specifically upsampling) and then change to
a codec that does not require resampling, we'll need to stick to
whatever input timestamp we left off at, rather than silently accepting
whatever we're sent.
BUG=622435
Review-Url: https://codereview.webrtc.org/2119393002
Cr-Commit-Position: refs/heads/master@{#13398}
Landing these in WebRTC under a guard so they don't build in
Chromium. The guard can be removed once Chromium has migrated to use the
new GN targets.
BUG=webrtc:6081
NOTRY=true
Review-Url: https://codereview.webrtc.org/2117183005
Cr-Commit-Position: refs/heads/master@{#13397}
Add null checks for eglBase, and detach the egl context in
SurfaceViewRenderer.surfaceDestroyed().
BUG=webrtc:6032
R=sakal@webrtc.org
Review URL: https://codereview.webrtc.org/2124983002 .
Cr-Commit-Position: refs/heads/master@{#13396}
This experiment was used to test the NackModule but will soon (tm) be used to
test the completly new video jitter buffer.
BUG=webrtc:5514
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2123913002 .
Cr-Commit-Position: refs/heads/master@{#13395}
This is called on received network packets if dump_rtp_packets_ is on.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2126463002
Cr-Commit-Position: refs/heads/master@{#13394}
I did some archaelogy here and found that the binary was (incorrectly?)
added here: https://codereview.webrtc.org/1903663002/. I can't find any
trace that a rtc_sdk_peerconnection_objc_tests has ever existed, or at
least that it doesn't exist now.
Removing it seems the right thing to do. However, I also see
rtc_sdk_peerconnection_objc target is folded into rtc_unittests in
webrtc_tests.gyp, but that target isn't listed in common_tests.json.
Should it be?
BUG=623500
TBR=tkchin@webrtc.org
Review-Url: https://codereview.webrtc.org/2121963002
Cr-Commit-Position: refs/heads/master@{#13392}
Add an option to use Camera2 implemantion of CameraVideoCapturer in
AppRTC Android Demo. It is enabled by default.
BUG=webrtc:5519
Review-Url: https://codereview.webrtc.org/2077663003
Cr-Commit-Position: refs/heads/master@{#13391}
A recent refactoring (r13192) introduced a bug where the min transmit
config wasn't being respected. Specifically, if a VideoSendStream was
created without it and the reconfigured, the min transmit bitrate would
not take effect. Probably the other way around as well.
BUG=webrtc::5687
Review-Url: https://codereview.webrtc.org/2106183002
Cr-Commit-Position: refs/heads/master@{#13390}
When the target bitrate is zero, currently VideoSendStream.Stats.target_media_bitrate_bps show the last set rate before the target was set to zero.
BUG=webrtc::5687 b/29574845
Review-Url: https://codereview.webrtc.org/2122743003
Cr-Commit-Position: refs/heads/master@{#13386}
1. The tool now displays packet loss in %.
2. It can print header information to stdout like rtp_analyze.
3. It has a command-line switch that lets you override the sample rate
guessing. With the flag "--query_sample_rate" the tool asks you to
always provide a sample rate.
4. Less decimals are printed for the estimated sample rate.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2123773002
Cr-Commit-Position: refs/heads/master@{#13385}
This changes the corpus semantics, but libfuzzer should be smart enough to figure it out, and if not then we can add a seed_corpus to help.
BUG=webrtc:4771
NOTRY=true
Review-Url: https://codereview.webrtc.org/2072473002
Cr-Commit-Position: refs/heads/master@{#13384}
Chromium's src/build/config/android/internal_rules.gni says java lib
target names must end with _java or _javalib.
Review-Url: https://codereview.webrtc.org/2117283002
Cr-Commit-Position: refs/heads/master@{#13382}
A bug in the transpot feedback adapter causes new feedback message to
always start with a received packet. This makes it impossible for the
receiver to distinguish from actual dropped packets and dropped feedback
messages.
BUG=webrtc:6073
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2122863002 .
Cr-Commit-Position: refs/heads/master@{#13381}
Let the FrameObject class inherit from VCMEncodedFrame since the rest of the
decoding pipeline use VCMEncodedFrame.
BUG=webrtc:5514
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2110543005 .
Cr-Commit-Position: refs/heads/master@{#13380}
I have added build files and renamed the mixer so that it doesn't conflict with the old one. The header includes now point to this copy of the mixer. I have also fixed some of the more obvious cases of style guide non-conformance and run 'PRESUBMIT' on the old mixer.
This is a first step in the creation of a new mixing module that will replace AudioConferencMixer and OutputMixer.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2104363003
Cr-Commit-Position: refs/heads/master@{#13378}
we will periodically check if any network does not have any connection on it and if yes, attempt to re-gather on those networks.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2025573002 .
Cr-Commit-Position: refs/heads/master@{#13367}
If all subencoders support textures, the adapter will claim support.
Texture frames will be passed on directly to subencoders, without any
attempt at scaling, and subencoders will be expected to sample/scale
correctly from source textures.
BUG=
NOTRY=true
Review-Url: https://codereview.webrtc.org/2099483002
Cr-Commit-Position: refs/heads/master@{#13365}
Deadlock caused by two methods grabbing two locks in the opposite order:
renderFrame():
handlerLock
layoutLock
onMeasure():
layoutLock
handlerLock
This CL removs the nested locking to fix the deadlock and make it less
error prone for the future.
BUG=webrtc:6003
R=sakal@webrtc.org
Review URL: https://codereview.webrtc.org/2111933002 .
Cr-Commit-Position: refs/heads/master@{#13364}
Reason for revert:
Issues fixed
Original issue's description:
> Revert of Combine webrtc/api/java/android and webrtc/api/java/src. (patchset #1 id:1 of https://codereview.webrtc.org/2111823002/ )
>
> Reason for revert:
> Breaks downstream dependencies
>
> Original issue's description:
> > Combine webrtc/api/java/android and webrtc/api/java/src.
> >
> > It used to be that there was a Java api for devices not running Android
> > but that is no longer the case. I combined the directories and made
> > the folder structure chromium style.
> >
> > BUG=webrtc:6067
> > R=magjed@webrtc.org, tommi@webrtc.org
> >
> > Committed: ceefe20dd6
>
> TBR=magjed@webrtc.org,tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6067
>
> Committed: 9b0dc622d4TBR=magjed@webrtc.org,tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6067
Review-Url: https://codereview.webrtc.org/2111923003
Cr-Commit-Position: refs/heads/master@{#13363}
This CL changes the auto-pause logic to suspend a stream based on the
encoder target bitrate instead of the allocated bitrate for a stream,
to account for possible protection, e.g. FEC and NACK.
This CL also adds periodic logging of the current BWE and possibility
to run with suspension in video loopback test.
BUG=webrtc:5868
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2117493002 .
Cr-Commit-Position: refs/heads/master@{#13360}
Permits CHECKing/DCHECKing that methods are being accessed in a
thread-safe manner, even if they are not used by one single thread
(thread pools such as VideoToolbox OK).
BUG=
R=danilchap@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2097403002 .
Cr-Commit-Position: refs/heads/master@{#13358}
Reason for revert:
Breaks downstream dependencies
Original issue's description:
> Combine webrtc/api/java/android and webrtc/api/java/src.
>
> It used to be that there was a Java api for devices not running Android
> but that is no longer the case. I combined the directories and made
> the folder structure chromium style.
>
> BUG=webrtc:6067
> R=magjed@webrtc.org, tommi@webrtc.org
>
> Committed: ceefe20dd6TBR=magjed@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6067
Review URL: https://codereview.webrtc.org/2106333005 .
Cr-Commit-Position: refs/heads/master@{#13357}
It used to be that there was a Java api for devices not running Android
but that is no longer the case. I combined the directories and made
the folder structure chromium style.
BUG=webrtc:6067
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2111823002 .
Cr-Commit-Position: refs/heads/master@{#13356}
I'm also removing media_optimization_unittest.cc, since it only tested the
suspension logic and nothing else.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/2119503002 .
Cr-Commit-Position: refs/heads/master@{#13355}
When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).
This effectively reduces the number of TURN candidates and connections created by TURN ports.
BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2093623004 .
Committed: https://crrev.com/17aac053f585e892114974d2eb248e05ad37f973
Cr-Original-Commit-Position: refs/heads/master@{#13335}
Cr-Commit-Position: refs/heads/master@{#13354}
