Patch set 1:
Run a script to replace occurrences of WEBRTC_TRACE logging with the new style,
in webrtc/modules/media_file/.
Patch set 2:
- Manually fix log lines not handled by the script
- Update the included headers
- Remove the now unused object ID variables
Bug: webrtc:5118
Change-Id: I1acbaec3fbbdf1deb7b934624a2f1fd38253c7e9
Reviewed-on: https://chromium-review.googlesource.com/602007
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19470}
Before this change we could crash in Debug when WebRTC audio was first
interrupted and then resumed again. The reason was that the new audio
stream stems from a new native I/O thread and that triggered thread
checkers. With this change, failing thread checkers are detached when
audio is interrupted to ensure that they don't fail when audio is restarted.
NOTRY=TRUE
Bug: webrtc:8126
Change-Id: Ib36ff6bc942477730aba60066f049ed0c43d3901
Reviewed-on: https://chromium-review.googlesource.com/628716
Commit-Queue: Henrik Andreasson <henrika@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19465}
no-unused-lambda-capture was suppressed, but it's been decided as desireable to stop suppressing it. This CL fixes places in the code that trigger it.
1. Some unnecessary captures removed.
2. s/constexpr/const when capturing a float by value - this is good enough to stop the error.
3. Complete removal of the constexpr/const-modifier for int-types as a workaround.
BUG=webrtc:7133
Review-Url: https://codereview.webrtc.org/3005433002
Cr-Commit-Position: refs/heads/master@{#19462}
On Windows a window may be covered by its own child window. So this change also
detects child windows by using EnumChildWindow().
The tooltip or context menu of the child window still cannot be detected after
this change. See bug for details.
Bug: webrtc:8062
Change-Id: I8455a9206d6a1d9da61013ac9debba4d3edae7d8
Reviewed-on: https://chromium-review.googlesource.com/619728
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19457}
This CL ensures that AEC3 recovers more quickly when capture data is
lost in such a manner that the echo path, as seen by AEC3, becomes
noncausal due to the AEC3 buffer misalignment caused by the data loss.
The CL adds the assumption of a minimum echo path delay of 5 blocks
and makes the hysteresis in the delay selection one-sided.
BUG=chromium:757796, webrtc:8131
Review-Url: https://codereview.webrtc.org/2998223002
Cr-Commit-Position: refs/heads/master@{#19454}
Moved the headers video_receive_stream.h and video_send_stream.h from
webrtc/ into webrtc/call/ as part of the Slim and Modular work.
The GN target webrtc:video_stream_api has moved to
webrtc/call:video_stream_api.
There are headers left in webrtc/ with the same name including the
moved headers in webrtc/call/ for not breaking external projects
depending on WebRTC.
At the same time, some minor cleanup is done: Non-pure-virtual functions declared in the two affected headers now have definitions in the same target. After making this change, our 'chromium-style' plugin detected some style violations that have now been fixed: non-inlined constructors and destructors have been added to a number of classes, both inside the GN target of the two affected headers, and in other targets.
BUG=webrtc:8107
Review-Url: https://codereview.webrtc.org/3000253002
Cr-Commit-Position: refs/heads/master@{#19448}
The whole point of all the audio codec stuff we've recently published
in api/ is to function as lego bricks so that building stuff like our
builtin audio codec factories will be easy.
BUG=webrtc:7821, webrtc:7822
Review-Url: https://codereview.webrtc.org/2997713002
Cr-Commit-Position: refs/heads/master@{#19446}
This test is and should be independent of RTP, so we don't need the
information provided in this struct.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2995403002
Cr-Commit-Position: refs/heads/master@{#19443}
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: f1e08d0b58TBR=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}
During a period of about one month we have only built with clang and not msvc, and during this period code that does not build with msvc have been submitted.
BUG=webrtc:8122
Review-Url: https://codereview.webrtc.org/2999343002
Cr-Commit-Position: refs/heads/master@{#19433}
* Guard members with a SequencedTaskChecker.
* Intercept encoder/decoder callbacks, and post onto task queue if needed.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2996253002
Cr-Commit-Position: refs/heads/master@{#19428}
The LockManagerOperation global function uses void** which does not work
well with the thread_annotations.h macros and produce compiler warnings
on clang (chromium default). Workarounds to this is hacky and unhelpful
so we disable the analysis for this function, which isn't helpful in
this case anyway.
webrtc_h264_config is no longer needed and is removed.
BUG=8090, 8119
Review-Url: https://codereview.webrtc.org/3000263002
Cr-Commit-Position: refs/heads/master@{#19425}
* Rename some members.
* Shorten visualization file names.
* Make some member functions static, in preparation for moving them
to be helper functions in an anonymous namespace.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/3001193002
Cr-Commit-Position: refs/heads/master@{#19424}
* Make ProcessFrame return void.
* Make |encode_callback_| and |decode_callback_| direct members.
* Remove ::EncodedFrameSize() and ::EncodedFrameType()
* Remove unused |timestamp| member from FrameInfo.
* Reorder printf output from PrintCodecSettings.
* Make some member functions const.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2998063002
Cr-Commit-Position: refs/heads/master@{#19421}
Clang is enabled by default and we currently lack coverage for building on Windows
with MSVC (see crbug.com/757293). This should unblock rolling WebRTC into
Chromium DEPS. We need to improve our trybot coverage for standalone WebRTC
to prevent things like this in the future though (crbug.com/756840).
BUG=webrtc:8119,chromium:756840,chromium:757293
TBR=hbos@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/3003473002
Cr-Commit-Position: refs/heads/master@{#19420}
1) Entering PROBE_RTT when necessary.
2) Congestion window gain of 0.65 instead of constant 4 packets.
3) {1.1, 0.9} pair instead of {1.25, 0.75}
4) Recovery mode.
5) No reaction to losses due to Recovery mode's implementation.
6) Supports encoder.
7) A new test compiling most of the simulation tests.
8) Bucket for high gain phase, disabled by default.
9) Pacer specific to BBR.
BUG=webrtc:7713
Review-Url: https://codereview.webrtc.org/2999073002
Cr-Commit-Position: refs/heads/master@{#19418}
Instead explicitly ignore only the flags we know should be ignored.
BUG=webrtc:7568
Review-Url: https://codereview.webrtc.org/2968003003
Cr-Commit-Position: refs/heads/master@{#19412}
Reason for revert:
iOS workaround.
Original issue's description:
> Revert of quest keyframes more frequently on stream start/decoding error. (patchset #2 id:170001 of https://codereview.webrtc.org/2996823002/ )
>
> Reason for revert:
> Causes iOS H264 calls received in the background to have increased delay before being able to decode stream from sender due to not having a keyframe.
>
> Original issue's description:
> > Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2994043002/ )
> >
> > Reason for revert:
> > Create fix CL.
> >
> > Original issue's description:
> > > Revert of Request keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.webrtc.org/2993793002/ )
> > >
> > > Reason for revert:
> > > Broke downstream test that was waiting for 5 keyframes to be received within 10 seconds. Maybe the issue is that "stats_callback_->OnCompleteFrame(frame->num_references == 0, ..." was changed to "frame->is_keyframe()"?
> > >
> > > Original issue's description:
> > > > Request keyframes more frequently on stream start/decoding error.
> > > >
> > > > In this CL:
> > > > - Added FrameObject::is_keyframe() convinience function.
> > > > - Moved logic to request keyframes on decoding error from VideoReceived to
> > > > VideoReceiveStream.
> > > > - Added keyframe_required as a parameter to FrameBuffer::NextFrame.
> > > >
> > > > BUG=webrtc:8074
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2993793002
> > > > Cr-Commit-Position: refs/heads/master@{#19280}
> > > > Committed: 26b4804358
> > >
> > > TBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,philipel@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:8074
> > >
> > > Review-Url: https://codereview.webrtc.org/2994043002
> > > Cr-Commit-Position: refs/heads/master@{#19295}
> > > Committed: 77a983185f
> >
> > TBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,deadbeef@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > BUG=webrtc:8074
> >
> > Review-Url: https://codereview.webrtc.org/2996823002
> > Cr-Commit-Position: refs/heads/master@{#19324}
> > Committed: 628ac5964e
>
> TBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,deadbeef@webrtc.org,philipel@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:8074
>
> Review-Url: https://codereview.webrtc.org/2995153002
> Cr-Commit-Position: refs/heads/master@{#19392}
> Committed: 53959fcc2bTBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,deadbeef@webrtc.org,tkchin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
BUG=webrtc:8074
Review-Url: https://codereview.webrtc.org/2996153003
Cr-Commit-Position: refs/heads/master@{#19410}
Reason for revert:
Create reland CL to add fix to.
Original issue's description:
> Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )
>
> Reason for revert:
> Speculative revet for breaking remoting_unittests in fyi bots.
> https://build.chromium.org/p/chromium.webrtc.fyi/waterfall?builder=Win7%20Tester
>
> Original issue's description:
> > Add a flags field to video timing extension.
> >
> > The rtp header extension for video timing shuold have an additional
> > field for signaling metadata, such as what triggered the extension for
> > this particular frame. This will allow separating frames select because
> > of outlier sizes from regular frames, for more accurate stats.
> >
> > This implementation is backwards compatible in that it can read video
> > timing extensions without the new flag field, but it always sends with
> > it included.
> >
> > BUG=webrtc:7594
> >
> > Review-Url: https://codereview.webrtc.org/3000753002
> > Cr-Commit-Position: refs/heads/master@{#19353}
> > Committed: cf5d485e14
>
> TBR=danilchap@webrtc.org,kthelgason@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7594
>
> Review-Url: https://codereview.webrtc.org/2995953002
> Cr-Commit-Position: refs/heads/master@{#19360}
> Committed: f0f7378b05TBR=danilchap@webrtc.org,kthelgason@webrtc.org,stefan@webrtc.org,emircan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2996153002
Cr-Commit-Position: refs/heads/master@{#19405}
No longer active with WebRTC, last commit 2014-10-10
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2999183002
Cr-Commit-Position: refs/heads/master@{#19403}
This change renames ScreenDrawerLockLinux into ScreenDrawerLockPosix and shares
it with Mac OSX.
Bug: webrtc:7950
Change-Id: Ib141781d2c35bfda0d6f9458fff235adbb643280
Reviewed-on: https://chromium-review.googlesource.com/607688
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19393}
Reason for revert:
Causes iOS H264 calls received in the background to have increased delay before being able to decode stream from sender due to not having a keyframe.
Original issue's description:
> Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2994043002/ )
>
> Reason for revert:
> Create fix CL.
>
> Original issue's description:
> > Revert of Request keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.webrtc.org/2993793002/ )
> >
> > Reason for revert:
> > Broke downstream test that was waiting for 5 keyframes to be received within 10 seconds. Maybe the issue is that "stats_callback_->OnCompleteFrame(frame->num_references == 0, ..." was changed to "frame->is_keyframe()"?
> >
> > Original issue's description:
> > > Request keyframes more frequently on stream start/decoding error.
> > >
> > > In this CL:
> > > - Added FrameObject::is_keyframe() convinience function.
> > > - Moved logic to request keyframes on decoding error from VideoReceived to
> > > VideoReceiveStream.
> > > - Added keyframe_required as a parameter to FrameBuffer::NextFrame.
> > >
> > > BUG=webrtc:8074
> > >
> > > Review-Url: https://codereview.webrtc.org/2993793002
> > > Cr-Commit-Position: refs/heads/master@{#19280}
> > > Committed: 26b4804358
> >
> > TBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,philipel@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:8074
> >
> > Review-Url: https://codereview.webrtc.org/2994043002
> > Cr-Commit-Position: refs/heads/master@{#19295}
> > Committed: 77a983185f
>
> TBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,deadbeef@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> BUG=webrtc:8074
>
> Review-Url: https://codereview.webrtc.org/2996823002
> Cr-Commit-Position: refs/heads/master@{#19324}
> Committed: 628ac5964eTBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,deadbeef@webrtc.org,philipel@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8074
Review-Url: https://codereview.webrtc.org/2995153002
Cr-Commit-Position: refs/heads/master@{#19392}
Make it possible for forced VP8 SW fallback encoder to set min_pixels_per_frame via GetScalingSettings().
Add a min required resolution (in addition to bitrate) before releasing forced SW fallback.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/3000693003
Cr-Commit-Position: refs/heads/master@{#19390}
This test is the only remaining one that does not use gtest and that's
blocking some infra cleanup tasks. Ideally this test would use
webrtc/rtc_base/flags.h but that's a lot of unnecessary work.
This also replaces some exit() status codes - the logic behind this is
if you get incorrectly specified command line arguments, exit(1) is
invoked for a failure, because it's not a test failure, and if flag
parsing was done properly, it would not be a gtest failure anyway.
BUG=webrtc:7568
Review-Url: https://codereview.webrtc.org/3000033002
Cr-Commit-Position: refs/heads/master@{#19388}
I want to publish an API for iSAC in webrtc/api/, and I want to use
the class names Audio{De,En}coderIsac{Fix,Float}.
BUG=webrtc:7835, webrtc:7841
Review-Url: https://codereview.webrtc.org/2996593002
Cr-Commit-Position: refs/heads/master@{#19381}
Reason for revert:
Reland
Original issue's description:
> Revert of Make the acceptable queue in the cwnd experiment configurable. (patchset #1 id:1 of https://codereview.webrtc.org/2998753002/ )
>
> Reason for revert:
> Speculative revert to see if this caused regressions in android perf tests.
>
> Original issue's description:
> > Make the acceptable queue in the cwnd experiment configurable.
> >
> > BUG=webrtc:7926
> >
> > Review-Url: https://codereview.webrtc.org/2998753002
> > Cr-Commit-Position: refs/heads/master@{#19320}
> > Committed: 7c83c56b6d
>
> TBR=philipel@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7926
>
> Review-Url: https://codereview.webrtc.org/2999893002
> Cr-Commit-Position: refs/heads/master@{#19337}
> Committed: c5d9e63c2bTBR=philipel@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7926
Review-Url: https://codereview.webrtc.org/2999083002
Cr-Commit-Position: refs/heads/master@{#19377}
WindowUnderPoint have different signatures on different platforms, which should
be abstract as an interface.
So this change adds a WindowFinder interface to replace WindowUnderPoint free
function. Meanwhile, this change also includes the implementation of
WindowFinderX11 for X11.
Bug: webrtc:7950
Change-Id: I897a50d4033e713b339b6b6f48b5dbbe601e8db0
Reviewed-on: https://chromium-review.googlesource.com/611745
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19375}
Reason for revert:
Reland
Original issue's description:
> Revert of Add functionality which limits the number of bytes on the network. (patchset #26 id:500001 of https://codereview.webrtc.org/2918323002/ )
>
> Reason for revert:
> Speculative revert to see if this caused regressions in android perf tests.
>
> Original issue's description:
> > Add functionality which limits the number of bytes on the network.
> >
> > The limit is based on the bandwidth delay product, but also adds some additional slack to compensate for the sawtooth-like BWE pattern and the slowness of the encoder rate control. The delay is estimated based on the time from sending a packet until an ack is received. Since acks are received in bursts (feedback is only sent periodically), a min filter is used to estimate the rtt.
> >
> > Whenever the in flight bytes reaches the congestion window, the pacer is paused, which in turn will result in send-side queues growing. Eventually the encoders will be paused as the pacer queue grows large (currently 2 seconds).
> >
> > BUG=webrtc:7926
> >
> > Review-Url: https://codereview.webrtc.org/2918323002
> > Cr-Commit-Position: refs/heads/master@{#19289}
> > Committed: 8497fdde43
>
> TBR=terelius@webrtc.org,philipel@webrtc.org,tschumim@webrtc.org,gnish@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7926
>
> Review-Url: https://codereview.webrtc.org/3001653002
> Cr-Commit-Position: refs/heads/master@{#19339}
> Committed: 64136af364TBR=terelius@webrtc.org,philipel@webrtc.org,tschumim@webrtc.org,gnish@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7926
Review-Url: https://codereview.webrtc.org/2994343002
Cr-Commit-Position: refs/heads/master@{#19373}
This replaces the WEBRTC_TRACE macros with LOG-macros.
Patchset 1: Run a formatting script, found in issue webrtc:5118.
Patchset 2: Apply manual fixes.
- Fix cases and formatting not handled by the script
- Replace a bit-shift / casting circus with utility function
in video_capture_linux.cc
Bug: webrtc:5118
Change-Id: Ib49c1c4d2502834b9d655dafa7c34bc47f1d73d9
Reviewed-on: https://chromium-review.googlesource.com/603709
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19371}
Example of new stop sequence:
PID TID
5155 5189 I WebRtcAudioTrack: stopPlayout
5155 5189 I WebRtcAudioTrack: underrun count: 0
5155 5189 I WebRtcAudioTrack: stopThread
5155 5189 I WebRtcAudioTrack: Stopping the AudioTrackThread...
5155 5236 I WebRtcAudioTrack: Stopping and flushing the audio track...
5155 5236 I WebRtcAudioTrack: The audio track has now been stopped.
5155 5189 I WebRtcAudioTrack: AudioTrackThread has now been stopped.
5155 5189 I WebRtcAudioTrack: releaseAudioResources
BUG=b/64692432
Review-Url: https://codereview.webrtc.org/3001703002
Cr-Commit-Position: refs/heads/master@{#19370}
This replaces the WEBRTC_TRACE macros with LOG-macros in the
following directories:
webrtc/modules/video_capture/objc/
webrtc/modules/video_capture/windows/
Patchset 1: Run a formatting script, found in issue webrtc:5118.
Patchset 2: Apply manual fixes.
- Fix cases and formatting not handled by the script
Bug: webrtc:5118
Change-Id: I0ac4b9f8f182d109844b57cfbba2574f47ab1e25
Reviewed-on: https://chromium-review.googlesource.com/605347
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19368}
The pacer has a mechanism to make sure all packets are sent within some
time limit. This is based on the average queue time of the packets in
the pacer queue.
If the pacer is paused while packets are still in the queue (for
instance if the underlying transport goes down temporarily), on resume
all those packets might be past the time limit and thus will all be
burst out onto the network in a tight loop.
This CL subtracts pause time from the queue time, effectively pausing
the clock for the queue while the pacer is paused, so that when we
resume the pacing bitrate will be the same as when we paused.
BUG=webrtc:7694
Review-Url: https://codereview.webrtc.org/2994323002
Cr-Commit-Position: refs/heads/master@{#19367}
