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}
Instead just use MD5Context type directly. In C++ it is unnecessary to
alias the types using typedef, unline C (where if you don't you have to
spell out struct or enum infront of the user-type everytime you want to make a
variable).
So since WebRTC's base API is C++, it seems unnecessay to keep this
typedef around.
BUG=None
TEST=rtc_unittests --gtest_filter=Md5*
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46799004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8916}
Tested on Linux with the following command lines:
$ gn gen //out/Debug --args='is_debug=true target_cpu="x64" build_with_chromium=false'
$ ninja -C out/Debug frame_analyzer rgba_to_i420_converter
BUG=chromium:461019
TEST=see above
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42999004
Cr-Commit-Position: refs/heads/master@{#8914}
This should save a bunch of disk space but most important
of all it will not sync the Chromium DEPS-pinned copy of WebRTC,
which can be very confusing when using IDEs that indexes all the
source code recursively.
TESTED=
$ rm chromium/.last_sync_chromium
$ rm -rf chromium/src/third_party/webrtc/
$ gclient sync
Verified chromium/src/third_party/webrtc/ didn't come back.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51489004
Cr-Commit-Position: refs/heads/master@{#8910}
Pass content_browsertests in Chromium. Performance test result (lower is
better):
C version: 100%
old intrinsics Neon version (with bug): 16.5%
new intrinsics Neon version: 18.0%
asm Neon version: 23.3%
BUG=4002
R=andrew@webrtc.org, jridges@masque.com
Change-Id: Ia0a96ac237216b635fc528f67d39319cdf246281
Review URL: https://webrtc-codereview.appspot.com/46739004
Cr-Commit-Position: refs/heads/master@{#8907}
