There is no point in returning an error when Free() fails. In fact it can only happen if we have a null pointer as object. There is further no place where the return value is used.
Affected components are
- aec
- aecm
- agc
- ns
BUG=441
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50579004
Cr-Commit-Position: refs/heads/master@{#8966}
Now that android_webview_build is no longer supported, remove build
conditionals referencing it and also remove the extra level of
indirection used to reference the cpufeatures target.
BUG=chromium:440793
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44119005
Patch from Richard Coles <torne@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8963}
We should not continue the program if the user asked for help.
Tested on Linux with the following command line:
$ out/Debug/frame_analyzer --help
BUG=None
TEST=see above
R=kjellander@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44069004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8961}
We used to symlink the .gn file from Chromium but it's now
replaced by our own copy (needed for recent GN changes in
https://codereview.chromium.org/988563002).
Relevant changes:
* src/third_party/boringssl/src: e2e1326..40acdae
* src/third_party/icu: 46be516..10834e8
* src/third_party/nss: bb4e75a..d1edb68
* src/tools/gyp: d174d75..2889664
The entries for
* src/third_party/jsoncpp/source/include
* src/third_party/jsoncpp/source/src/lib_json
are removed and replaced by:
* src/third_party/jsoncpp/source @ab1e40f
(which doesn't matter for us since we symlink third_party/jsoncpp
Details: 5333e14..d8f8dc8/DEPS
Clang version was not updated in this roll.
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48919004
Cr-Commit-Position: refs/heads/master@{#8959}
CopyCodecSpecific nulls out the rtpheader pointer hence causing the crash downstream.
More details about the codec type enums:
There are 2 enums defined. webrtc::VideoCodecType webrtc::RtpCodecTypes and they don't match. Inside CopyCodecSpecific in generic_encoder.cc, it was converted from the first to the 2nd type. At that point, it'll be kRtpVideoNone (as the effect of memset to 0). kRtpVideoNone is a bad value as it could cause assert. Later, it'll be reset to kRtpVideoGeneric in RTPSender::SendOutgoingData so it's not a concern.
BUG=4511
R=pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
Committed: https://crrev.com/29b1a1c0c7c6f4b1ae4d63844b1dfaa7a72530a0
Cr-Commit-Position: refs/heads/master@{#8951}
Review URL: https://webrtc-codereview.appspot.com/47999004
Cr-Commit-Position: refs/heads/master@{#8955}
CopyCodecSpecific nulls out the rtpheader pointer hence causing the crash downstream.
More details about the codec type enums:
There are 2 enums defined. webrtc::VideoCodecType webrtc::RtpCodecTypes and they don't match. Inside CopyCodecSpecific in generic_encoder.cc, it was converted from the first to the 2nd type. At that point, it'll be kRtpVideoNone (as the effect of memset to 0). kRtpVideoNone is a bad value as it could cause assert. Later, it'll be reset to kRtpVideoGeneric in RTPSender::SendOutgoingData so it's not a concern.
BUG=4511
R=pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47999004
Cr-Commit-Position: refs/heads/master@{#8951}
Since the approach in https://webrtc-codereview.appspot.com/48869004/
was not possible, this is the only way to set
the target_subarch GYP_DEFINES for iOS.
WebRTC doesn't use or support the target_subarch
GYP variable in the same way as Chromium does, and we
currently don't build fat binaries at our buildbots.
This is needed to unblock the chromium_revision roll in
https://webrtc-codereview.appspot.com/50569004/ and it
also makes it a bit easier to build for iOS since it
is no longer needed to set target_subarch=arm64 when
target_arch=arm64 (i.e. when you build for ARM64).
BUG=4503
TESTED=Ran successful project generations using:
GYP_DEFINES="OS=ios target_arch=arm" webrtc/build/gyp_webrtc
and verified the -arch compiler flag was set to 'armv7'
GYP_DEFINES="OS=ios target_arch=arm64" webrtc/build/gyp_webrtc
and verified the -arch compiler flag was set to 'arm64'
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48909004
Cr-Commit-Position: refs/heads/master@{#8949}
Since the GYP variables use_openssl and build_ssl are both
enabled by default on Windows today, I believe we should be able
to clean up the GYP paths that build NSS for Windows.
BUG=4497
TESTED=Passing tryjobs with --clobber specified.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50559004
Cr-Commit-Position: refs/heads/master@{#8945}
- Check texture properties update flag using the same lock under which
the flag value is set.
- Adjust texture properties inside frame queue lock.
- Plus adding extra logging to track video renderer properties updates.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45929004
Cr-Commit-Position: refs/heads/master@{#8941}
WebRTC doesn't use or support the target_subarch
GYP variable in the same way as Chromium does, and we
currently don't build fat binaries at our buildbots.
This is needed to unblock the chromium_revision roll in
https://webrtc-codereview.appspot.com/50569004/ and it
also makes it a bit easier to build for iOS since it
is no longer needed to set target_subarch=arm64 when
target_arch=arm64 (i.e. when you build for ARM64).
TBR=tkchin@webrtc.org
BUG=4503
Review URL: https://webrtc-codereview.appspot.com/48869004
Cr-Commit-Position: refs/heads/master@{#8937}
This CL makes ViEFrameCallback::DeliverFrame const and removes the potential frame copy in ViEFrameProviderBase by moving it to ViEEncoder::DeliverFrame instead, for clients that use the FrameCallback functionality to modify the frame content.
BUG=1128
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43949004
Cr-Commit-Position: refs/heads/master@{#8934}
- Make PeerConnectionClient a singleton.
- Fix crash in CpuMonitor.
- Remove reading constraints from room response.
- Catch and report camera errors.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43059004
Cr-Commit-Position: refs/heads/master@{#8930}
UDP case should not be changed.
Active TCPConnection will initiate Reconnect after OnClose and when Send or Ping fails.
Passive TCPConnection will prune itself as usual as the active side will create a new connection.
The Reconnect could make P2PCT choose a different best_connection in the case where connectivities exist b/w more than 1 Network.
Also, to avoid upper layer triggers ice restart, the WRITE_TIMEOUT caused by the socket disconnection is delayed to give the reconnect mechanism chance to kick in. The timeout event is only fired if the reconnect can't work in 5 sec. If the reconnect, there should be no ICE disconnected state trigger either in active or passive side.
BUG=1926
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31359004
Cr-Commit-Position: refs/heads/master@{#8929}
No need to include this header since the API is just taking a pointer to
it.
BUG=1092
TEST=./webrtc/build/gyp_webrtc && ninja -C out/Debug
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44059004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8928}
This cl uses the YV12 buffers from Java without a copy if no rotation is needed. Buffers are returned to the camera when the encoder and renderers no longer needs them.
This add a new frame type WrappedI420Buffer based in that allows for wrapping existing memory buffers and getting a notification when it is no longer used.
AndroidVideoCapturer::FrameFactory::CreateAliasedFrame wraps frame received from Java. For each wrapped frame a new reference to AndroidVideoCapturerDelegate is held to ensure that the delegate can not be destroyed until all frames have been returned.
Some overlap exist in webrtcvideoframe.cc and webrtcvideengine.cc with https://webrtc-codereview.appspot.com/47399004/ that is expected to be landed before this cl.
BUG=1128
R=glaznev@webrtc.org, magjed@webrtc.orgTBR=mflodman@webrtc.org // For changes in webrtc/common_video/video_frame_buffer
Review URL: https://webrtc-codereview.appspot.com/49459004
Cr-Commit-Position: refs/heads/master@{#8923}
This should fix the TODO in that header.
BUG=None
TEST=ninja -C out/Debug still compiles everything.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47919004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8921}
And remove pass-through stringdigest.h include.
This should fix the TODO in stringdigest.h that were that saying to update the callers to the new location.
BUG=None
TEST=ninja -C out/Debug still works fine
R=henrika@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48779004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8920}
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
hence trivial.
The macro name may in fact mislead the user to assume a cast/truncation to int32_t is done.
- Removing usage of it.
- Some style changes.
BUG=3348, 3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46749005
Cr-Commit-Position: refs/heads/master@{#8918}
