"error: values of type 'NSInteger' should not be used as format arguments; add an explicit cast to 'long' instead"
Casting to long is already a common practice in the code base.
This has been blocking the Chromium roll which contains an update to clang 6.0.0
BUG=None
Review-Url: https://codereview.webrtc.org/2987693002
Cr-Commit-Position: refs/heads/master@{#19127}
This broke WebRTC's presubmit
e79ddeaabf%5E%21/
GClientKeywords has been removed and replaced with a more direct substitution.
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2989603002
Cr-Commit-Position: refs/heads/master@{#19126}
There is an inconsistency in behavior of PeerConnection.
When I remove track from PeerConnection observer->OnRenegotiationNeeded is called, however if I remove track from MediaStream then there is no notification to renegotiate.
This patch adds missing OnRenegotiationNeeded calls.
BUG=webrtc:7966
Review-Url: https://codereview.webrtc.org/2977493002
Cr-Commit-Position: refs/heads/master@{#19125}
These traces will be traced instead when getStats()
is called by JavaScript.
BUG=chromium:653087
Review-Url: https://codereview.webrtc.org/2972393002
Cr-Commit-Position: refs/heads/master@{#19124}
when creating RtpRtcp module for video send stream.
BUG=webrtc:8016
Review-Url: https://codereview.webrtc.org/2979363002
Cr-Commit-Position: refs/heads/master@{#19122}
We currently don't close the peerconnection before deallocing. That
could potentially cause race conditions if it's still being processed on
other threads.
BUG=webrtc:7976
Review-Url: https://codereview.webrtc.org/2976983002
Cr-Commit-Position: refs/heads/master@{#19121}
A multiplication result doesn't fit in an int32_t type. This change
rewrites the code to avoid the overflowing multiplication.
Here y[0], y[1] are int16 numbers containing the (truncated) topmost
18 and (scaled Q2 to use the full int16) the least significant 13
bits of a 32-bit value. The change makes y[1] to be calculated
directly instead of using y[0] as an intermediate value.
TESTED=this change passes the bit exactness tests, and has also been
running on the audio_processing fuzzer with a CHECK comparing the
old and new value.
Bug: chromium:747202
Change-Id: Iafc69eb7391d494afdadf65f5b7f399a57bbe9a8
Reviewed-on: https://chromium-review.googlesource.com/580907
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19120}
These tests will be reenabled and updated after Opus has been updated in
Chromium and rolled into WebRTC.
BUG=737323, webrtc:8024
Review-Url: https://codereview.webrtc.org/2963673002
Cr-Commit-Position: refs/heads/master@{#19118}
Current implementation requires MouseCursorMonitor to understand the SourceId of
a DesktopCapturer implementation. But SourceId has different meanings across
various DesktopCapturer implementations. So this change decouples the
MouseCursorMonitor from DesktopCapturer, i.e. it does not need to know
DesktopCapturer anymore, instead it always returns the absolute position of the
mouse cursor. In DesktopAndCursorComposer, it can use the newly added
DesktopFrame::top_left() to decide the relative position of mouse cursor and the
DesktopFrame.
Bug: webrtc:7950
Change-Id: Idfbde5cb0f79ff0acf4ad1e9a0ac5126f1bb2e98
Reviewed-on: https://chromium-review.googlesource.com/575315
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19115}
Reason for revert:
Breaks IpcNetworkManagerTest.TestMergeNetworkList, because it has built-in assumptions about network ordering that it shouldn't have. Will reland after fixing that test.
Original issue's description:
> Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting.
>
> This CL moves the responsibility for restricting the number of IPv6
> interfaces used for ICE to BasicPortAllocator. This is the right place
> to do it in the first place; it's where all the rest of the filtering
> occurs. And NetworkManager shouldn't need to know about ICE limitations;
> only the ICE classes should.
>
> Part of the reason I'm doing this is that I want to add a
> "max_ipv6_networks" API to RTCConfiguration, so that applications can
> override the default easily (see linked bug). But that means that
> PeerConnection would need to be able to call "set_max_ipv6_networks" on
> the underlying object that does the filtering, and that method isn't
> available on the "NetworkManager" base class. So rather than adding
> another method to a place it doesn't belong, I'm moving it to the place
> it does belong.
>
> In the process, I noticed that "CompareNetworks" is inconsistent with
> "SortNetworks"; the former orders interfaces alphabetically, and the
> latter reverse-alphabetically. I believe this was unintentional, and
> results in undesirable behavior (like "eth1" being preferred over
> "eth0"), so I'm fixing it and adding a test.
>
> BUG=webrtc:7703
>
> Review-Url: https://codereview.webrtc.org/2983213002
> Cr-Commit-Position: refs/heads/master@{#19112}
> Committed: ad9561404cTBR=zhihuang@webrtc.org,pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7703
Review-Url: https://codereview.webrtc.org/2984853002
Cr-Commit-Position: refs/heads/master@{#19114}
Allows using sizeof() on the class constants and reduces space usage by
a pointer.
Bug: None
Change-Id: Ie919b13094903d50bdadc92b23a5aa5b6cc100ec
Reviewed-on: https://chromium-review.googlesource.com/581878
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19113}
This CL moves the responsibility for restricting the number of IPv6
interfaces used for ICE to BasicPortAllocator. This is the right place
to do it in the first place; it's where all the rest of the filtering
occurs. And NetworkManager shouldn't need to know about ICE limitations;
only the ICE classes should.
Part of the reason I'm doing this is that I want to add a
"max_ipv6_networks" API to RTCConfiguration, so that applications can
override the default easily (see linked bug). But that means that
PeerConnection would need to be able to call "set_max_ipv6_networks" on
the underlying object that does the filtering, and that method isn't
available on the "NetworkManager" base class. So rather than adding
another method to a place it doesn't belong, I'm moving it to the place
it does belong.
In the process, I noticed that "CompareNetworks" is inconsistent with
"SortNetworks"; the former orders interfaces alphabetically, and the
latter reverse-alphabetically. I believe this was unintentional, and
results in undesirable behavior (like "eth1" being preferred over
"eth0"), so I'm fixing it and adding a test.
BUG=webrtc:7703
Review-Url: https://codereview.webrtc.org/2983213002
Cr-Commit-Position: refs/heads/master@{#19112}
This is the first in a series of CLs to add support for media
identification as part of unified plan SDP.
Bug: webrtc:4050
Change-Id: I0eb5639d240a9a1412c2b047a33d5112e4901f26
Reviewed-on: https://chromium-review.googlesource.com/576374
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19111}
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}
