Print general usage information for event_log_analyzer (in addition to listing the command line flags) when called with '--help'.
BUG=None
Review-Url: https://codereview.webrtc.org/2986573002
Cr-Commit-Position: refs/heads/master@{#19104}
Removes keying on pattern_idx inside TemporalLayers implementations for
several properties that are different between screencast temporal layers
and normal/default temporal layers.
This is a step towards sharing PopulateCodecSpecific between the layer
patterns and code deduplication to longer term be able to separate the
packetizer step from encoder settings, so that temporal patterns can be
used for asynchronous hardware encoders where there may be outstanding
frames.
BUG=chromium:702017, webrtc:7349
R=brandtr@webrtc.org
Review-Url: https://codereview.webrtc.org/2924993002
Cr-Commit-Position: refs/heads/master@{#19097}
SrtpTransport currently just delegates everything to RtpTransport.
Also makes BaseChannel::rtp_transport_ an RtpTransportInternal and constructs an SrtpTransport if srtp is required.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/2981013002
Cr-Commit-Position: refs/heads/master@{#19095}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2976293002
Cr-Commit-Position: refs/heads/master@{#19094}
Patch set 1:
Run a script to replace occurrences of WEBRTC_TRACE logging with the new style, on webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc.
Patch set 2:
- Manually fix log lines not handled by the script
- Adjust local macros that use WEBRTC_TRACE
- Adjust some lines to conform with code style
- Update the included headers
- Remove the now unused object ID variables
BUG=webrtc:5118
Review-Url: https://codereview.webrtc.org/2985443002
Cr-Commit-Position: refs/heads/master@{#19088}
through streams related to a call object.
The Call object does not know directly when packets pass through it, only which
AudioSendStreams are used. Each AudioSendStream has a pointer to the Transport
object through which its packets are send.
This CL:
By registering an internal wrapper class, TimedTransport, the AudioSendStream
can stay up-to-date on when packets have passed through its Transport. This
lifetime (as an interval) is then queried by the Call when the AudioSendStream
is destroyed. When Call is destroyed, all streams are guaranteed to have been
destroyed and hence Call is up-to-date on packet activity.
The class TimeInterval keeps the code in Call and AudioSendStream smaller, with
fewer get methods in their APIs and less code for updating values.
Also modifies the unit test for AudioSendStream: it previously enforced that
the stream registers (with its channel proxy) the same transport that it was
constructed with.
BUG=webrtc:7882
Review-Url: https://codereview.webrtc.org/2979833002
Cr-Commit-Position: refs/heads/master@{#19087}
HardwareVideoEncoderFactory can now take an EglBase.Context on creation.
When it does, it creates video encoders in texture mode. It uses the
COLOR_FormatSurface colorFormat. It passes the EglBase.Context to the
HardwareVideoEncoder.
The HardwareVideoEncoder sets up an input surface for its codec and handles
incoming frames by drawing them onto the input surface.
BUG=webrtc:7760
R=pthatcher@webrtc.org, sakal@webrtc.org
Review-Url: https://codereview.webrtc.org/2977153003 .
Cr-Commit-Position: refs/heads/master@{#19083}
HardwareVideoDecoder is now a listener for SurfaceTextureHelper. It takes a
SurfaceTextureHelper on construction. If it is non-null, it operates in texture
mode instead of byte-buffer mode.
When in texture mode, the HardwareVideoDecoder renders output frames to a Surface,
listens for the texture frame to become available, wraps it in a VideoFrame, and
pushes it to the decoder callback.
As in MediaCodecVideoDecoder, it may queue up to three buffers while waiting for
the surface to become available for rendering. If more buffers are queued, it will
drop the oldest.
This change also implements the VideoFrame.TextureBuffer and reorganizes code
for wrapping an existing ByteBuffer into an I420Buffer. This makes it easier
to implement the texture buffer's ToI420() method.
BUG=webrtc:7760
R=pthatcher@webrtc.org, sakal@webrtc.org
Review-Url: https://codereview.webrtc.org/2977643002 .
Cr-Commit-Position: refs/heads/master@{#19081}
This change ensures DirectX capturer to return the same ScreenId as GDI capturer
for each monitor. So MouseCursoeMonitor can work correctly with the DirectX
capturer.
This is a temporary fix of webrtc:7950.
Bug: webrtc:7950
Change-Id: Icd3f40556701811c21c773a39260a74db43979f3
Reviewed-on: https://chromium-review.googlesource.com/571101
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19079}
The variable 'tmp_int32' in LowCutFilter::BiqueadFilter::Process can
be negative. This replaces a left shift with multiplication.
Bug: chromium:735593, chromium:743330
Change-Id: Idec7fbcc17495f7241eb4bea44920585740e3695
Reviewed-on: https://chromium-review.googlesource.com/575136
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19074}
The surface view renderer size was set to match parent so it couldn't
adjust based on the frame size. The size is now set to wrap_content
which allows the renderer to adjust. The root element of the call
activity is changed to FrameLayout to allow the renderer to center.
requestLayout is added to SurfaceView setScalingType so onMeasure gets
called again.
BUG=webrtc:7901
Review-Url: https://codereview.webrtc.org/2978173002
Cr-Commit-Position: refs/heads/master@{#19073}
Run a script to replace occurrences of WEBRTC_TRACE logging with the new
style, on webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc.
Patch set 2:
- Manually fix log lines not handled by the script
- Adjust some lines, to conform with code style
- Update the included headers
- Remove the now unused object ID variables
- - This explains why there's so many files edited
BUG=webrtc:5118
TBR=henrika@webrtc.org
Review-Url: https://codereview.webrtc.org/2978083002
Cr-Commit-Position: refs/heads/master@{#19071}
WebRtcSpl_AnalysisQMF takes raw (user) audio input represented by
int16_t samples. The samples are converted to Q10 with the
WEBRTC_SPL_LSHIFT_W32 macro. The macro is implemeted as a left
shift. This CL replaces the shift with a multiplication, similar
to https://codereview.webrtc.org/2253943002TBR=kwiberg@webrtc.org
Bug: chromium:735773
Change-Id: Ic4e63269390e82b86f304e5aa1b5e2dc22122bcb
Reviewed-on: https://chromium-review.googlesource.com/552124
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19068}
This was included to avoid breaking chromium, which now includes its own implementation (725cb26dab).
BUG=webrtc:7395
Review-Url: https://codereview.webrtc.org/2924243003
Cr-Commit-Position: refs/heads/master@{#19063}
Bug: webrtc:7969
Change-Id: I3fbb723d35fa6cc4c7b92aa1e155b974e9fb0b55
Reviewed-on: https://chromium-review.googlesource.com/567698
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@chromium.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19062}
IsCurrentSessionSupported() is useful to decide whether Windows version should
be used to evaluate the capability of DirectX capturer on the system.
Bug: 741926
Change-Id: Iaaf6011a9e464d7cf5e7dda097007778c73953e0
Reviewed-on: https://chromium-review.googlesource.com/571378
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19060}
Reason for revert:
Still having problems with no video. Reverting.
Once no video is visible, no video is available from then on even if the callee app is in the foreground.
Original issue's description:
> Reland of Injectable Obj-C video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2979973002/ )
>
> Reason for revert:
> Fix the broken build file
>
> Original issue's description:
> > Revert of Injectable Obj-C video codecs (patchset #3 id:400001 of https://codereview.webrtc.org/2981583002/ )
> >
> > Reason for revert:
> > Breaks bots. Build file incorrect.
> >
> > Original issue's description:
> > > Reland of Injectable Obj-C video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2975963002/ )
> > >
> > > Reason for revert:
> > > New CL for fixing the issues
> > >
> > > Original issue's description:
> > > > Revert of Injectable Obj-C video codecs (patchset #8 id:140001 of https://codereview.webrtc.org/2966023002/ )
> > > >
> > > > Reason for revert:
> > > > Causes no video in certain scenarios. Please come up with a test plan or unit test to prevent such problems in the future.
> > > >
> > > > Original issue's description:
> > > > > Injectable Obj-C video codecs
> > > > >
> > > > > Initial CL for this effort, with a working RTCVideoEncoder/Decoder for H264
> > > > > (wrapping the VideoToolbox codec).
> > > > >
> > > > > Some notes / things left to do:
> > > > > - There are some hard-coded references to codec types that are supported by
> > > > > webrtc::VideoCodec, cricket::VideoCodec, webrtc::CodecSpecificInfo etc
> > > > > since we need to convert to/from these types in ObjCVideoEncoder/Decoder.
> > > > > These types would need to be more codec agnostic to avoid this.
> > > > > - Most interfaces are borrowed from the design document for injectable
> > > > > codecs in Android. Some data in the corresponding C++ classes is discarded
> > > > > when converting to the Obj-C version, since it has fewer fields. I have not
> > > > > verified whether all data that we do keep is needed, or whether we might be
> > > > > losing anything useful in these conversions.
> > > > > - Implement the VideoToolbox codec code directly in the RTCVideoEncoderH264
> > > > > classes, instead of wrapping webrtc::H264VideoToolboxEncoder / decoder.
> > > > > Eliminates converting between ObjC/C++ types outside the ObjCVideoEncoder/
> > > > > Decoder wrapper classes.
> > > > > - List the injected codec factory's supported codecs in the list of codecs in
> > > > > AppRTCMobile.
> > > > >
> > > > > BUG=webrtc:7924
> > > > > R=magjed@webrtc.org
> > > > >
> > > > > Review-Url: https://codereview.webrtc.org/2966023002 .
> > > > > Cr-Commit-Position: refs/heads/master@{#18928}
> > > > > Committed: a0349c138d
> > > >
> > > > TBR=magjed@webrtc.org,andersc@webrtc.org
> > > > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > > > BUG=webrtc:7924
> > > > NOTRY=true
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2975963002
> > > > Cr-Commit-Position: refs/heads/master@{#18979}
> > > > Committed: 1095ada7ad
> > >
> > > R=magjed@webrtc.org
> > > TBR=tkchin@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:7924
> > >
> > > Review-Url: https://codereview.webrtc.org/2981583002 .
> > > Cr-Commit-Position: refs/heads/master@{#19002}
> > > Committed: a5f1de1e65
> >
> > TBR=magjed@webrtc.org,tkchin@webrtc.org,jtteh@webrtc.org,andersc@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7924
> >
> > Review-Url: https://codereview.webrtc.org/2979973002
> > Cr-Commit-Position: refs/heads/master@{#19004}
> > Committed: 81d40ee149
>
> TBR=magjed@webrtc.org,tkchin@webrtc.org,jtteh@webrtc.org,sprang@webrtc.org
> BUG=webrtc:7924
>
> Review-Url: https://codereview.webrtc.org/2979983002
> Cr-Commit-Position: refs/heads/master@{#19005}
> Committed: 732a3437daTBR=magjed@webrtc.org,tkchin@webrtc.org,sprang@webrtc.org,haysc@webrtc.org,andersc@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7924
Review-Url: https://codereview.webrtc.org/2980173002
Cr-Commit-Position: refs/heads/master@{#19036}
This reverts commit 93adc3209b5ff10adaba54d5eab6b53bc2780685.
Reverted originally because it depended on a CL which was reverted.
That CL has been reinstated in:
https: //chromium-review.googlesource.com/#/c/572070/
Bug: webrtc:7969
Change-Id: I608bbeaaba02e84908433c8260cf236df0307a97
Reviewed-on: https://chromium-review.googlesource.com/572405
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19035}
This reverts commit 26d5e2e2809558148dc1e977ec1bc8318a2047bc.
Reverted originally because it dependend on a CL which was reverted. That CL has been reinstated in: https://chromium-review.googlesource.com/#/c/572070/
Bug: webrtc:7969
Change-Id: I404c3a42ad447312d981646dca0aa4cf0ec3134e
Reviewed-on: https://chromium-review.googlesource.com/572403
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19034}
Use rtc::SystemTimeNanos() instead of std::random_device() for PRNG seed
to avoid crashing when /dev/urandom is unavailable.
This reverts commit 3beb20720db349f651c2c04970c45b1b171c025c.
Bug: webrtc:7969
Change-Id: I5ed58a789939ee4caa99ac3abf9cab18e3e19c69
Reviewed-on: https://chromium-review.googlesource.com/572070
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19033}
This is a partial re-land. The change doesn't make the default Thread ctor protected anymore but it does mark it as deprecated and updates all use of it in WebRTC.
Original issue's description:
Make the default ctor of rtc::Thread, protected.
The goal is to force use of Thread::Create or Thread::CreateWithSocketServer.
The default constructor constructs a 'default' socket server, which is usually a 'physical' socket server, but not always. Not every instance of Thread actually needs to have network support, so it's better to have this be explicit instead of unknowingly instantiate one.
BUG=none
Review-Url: https://codereview.webrtc.org/2977953002
Cr-Commit-Position: refs/heads/master@{#19031}
Currently, when we generate the list of supported video codecs that will
be signaled in SDP, we start with the internal video codecs and then
append the external video codecs. When we create a video encoder for a
given codec, we prefer an external encoder over an internal encoder.
This CL lists the external video codecs first in SDP instead, so that we
consistently prefer external video codecs over internal.
The reason for doing this is that we will otherwise prefer an internal
SW H264 encoder over an external HW H264 encoder if the H264 profiles
differs.
BUG=chromium:688541
Review-Url: https://codereview.webrtc.org/2974383002
Cr-Commit-Position: refs/heads/master@{#19026}
This reverts commit aa41f0cfa64ece911ae2ecee83fc3190d4a42935.
Reason for revert:
Apparently, use of std::random_device() causes chromium on Linux to fail with this error:
terminating with uncaught exception of type std::__1::system_error: random_device failed to open /dev/urandom: Operation not permitted
Link to bot with failure:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Tester/builds/37563
Original change's description:
> API for periodically regathering ICE candidates
>
> Adds to the RTCConfiguration `ice_regather_interval_range` which, when
> set, specifies the randomized delay between automatic runs of ICE
> regathering. The regathering will occur on all networks and re-use the
> existing ICE ufrag/password. New connections are established once the
> candidates come back and WebRTC will automatically switch to the new
> connection that corresponds to the currently selected connection.
>
> Bug: webrtc:7969
> Change-Id: I6bbf5439a48e285f704aed9f408631cba038c82b
> Reviewed-on: https://chromium-review.googlesource.com/562505
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18978}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,steveanton@webrtc.org
No-Try: true
Bug: webrtc:7969
Change-Id: I86ef99e9f1070d3ac265398831317b68f562c614
Reviewed-on: https://chromium-review.googlesource.com/571008
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19024}
Run a script to replace occurrences of WEBRTC_TRACE logging with the new
style, on webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc.
Patch set 2:
Manually fix log lines not handled by the script, remove unused header
and variable.
I would like to do this will the following files, too:
webrtc/modules/audio_device/..
.../linux/audio_device_alsa_linux.cc
.../linux/audio_device_pulse_linux.cc
.../linux/audio_mixer_manager_alsa_linux.cc
.../linux/audio_mixer_manager_pulse_linux.cc
.../linux/latebindingsymboltable_linux.cc
.../mac/audio_device_mac.cc
.../mac/audio_mixer_manager_mac.cc
.../win/audio_device_core_win.cc
BUG=webrtc:5118
Review-Url: https://codereview.webrtc.org/2978953003
Cr-Commit-Position: refs/heads/master@{#19019}
If a SurfaceViewRenderer is reinitialized, the onFirstFrameRendered
callback is not fired.
Ensure that we reset the flag when the SurfaceViewRenderer is
initialized.
BUG=webrtc:7985
Review-Url: https://codereview.webrtc.org/2981793002
Cr-Commit-Position: refs/heads/master@{#19016}
Both DXGI_OUTPUT_DESC and DISPLAY_DEVICE contain the DeviceName, which may be
able to map a DirectX screen id with the GDI screen id.
So this change exports the field from both DirectX and GDI implementations.
BUG=webrtc:7950
Review-Url: https://codereview.webrtc.org/2971393002
Cr-Commit-Position: refs/heads/master@{#19010}
Check that ice_regather_interval_range is set only when continual
regathering is also set.
Bug: webrtc:7969
Change-Id: Ifcfeee744d817cf00914418d7e682f11528faf05
Reviewed-on: https://chromium-review.googlesource.com/569358
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19009}
GetSystemInfo will not return more than 32 for dwNumberOfProcessors when
called from a 32-bit process. This means that Chrome lies to me whenever
I enable logging. Calling GetNativeSystemInfo allows Chrome to return up
to 64 as the processor count. GetNativeSystemInfo even runs on WindowsXP
if that matters.
With the fix applied in a Chromium repo the logging at startup now says:
[320:196:712/335.515:INFO:cpu_info.cc(50)] Available number of cores: 48
BUG=webrtc:7981
Review-Url: https://codereview.webrtc.org/2978863002
Cr-Commit-Position: refs/heads/master@{#19008}
