For simulcast VideoSendStreams there are more than one entry in the SSRC
table causing RTCP to currently be delivered more than once per stream.
This messes up unique NACK stats as all NACK requests sent to such a
sender will be delivered multiple times and not look unique.
BUG=4544
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50709004
Cr-Commit-Position: refs/heads/master@{#8998}
https://webrtc-codereview.appspot.com/48909004/ only fixed
the target_subarch problem for ARM. This fixes it for the
x86 targets that are used for the iOS simulator (ia32/x64).
BUG=4532
TESTED=Built locally using:
export GYP_DEFINES="build_with_libjingle=1 build_with_chromium=0 libjingle_objc=1 OS=ios target_arch=ia32"
export GYP_GENERATOR_FLAGS="$GYP_GENERATOR_FLAGS output_dir=out_sim"
export GYP_CROSSCOMPILE=1
gclient runhooks
ninja -C out_sim/Release-iphonesimulator iossim AppRTCDemo
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44159004
Cr-Commit-Position: refs/heads/master@{#8991}
When Chromium decides to move conflicting dependencies
to the _bad_scm directory it is possible to end up with
problems when that directory gets added to the local index.
Ignoring it should avoid such problems.
R=hbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45969004
Cr-Commit-Position: refs/heads/master@{#8988}
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) \
((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h)
BUG=3348,3353
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43129004
Cr-Commit-Position: refs/heads/master@{#8986}
Acquiring stream_lock_ in WebRtcVideoChannel2 in a callback from Call
forms a lock-order inversion between process-thread locks and libjingle
locks, manifesting as CPU adaptation requests blocking on stream
creation that is blocked on the CPU adaptation request finishing.
R=asapersson@webrtc.org, mflodman@webrtc.org
BUG=4535,chromium:475065
Review URL: https://webrtc-codereview.appspot.com/50679004
Cr-Commit-Position: refs/heads/master@{#8985}
Including check.h causes build failure in Chrome due to LOG macros redefinition.
Review URL: https://webrtc-codereview.appspot.com/51629004
Cr-Commit-Position: refs/heads/master@{#8984}
Changes:
1. Removed _voiceEngineObserver boolean flag, because its value is equal to (_voiceEngineObserverPtr != NULL).
2. Removed WEBRTC_TRACE macro usage wherever it was unnecessary to log. Replaced its usage with LOG_F (new and preferred way to log messages) wherever it is useful to log.
3. Replaced asserts with CHECKs.
Discussion:
To make it easier to review the changes, I didn't reformat the code to make it compliant to the new coding standards. It is up for debate how much reformatting to do: the whole file/class or just the methods that I have touched. My vote - go for the whole class.
R=henrika@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51579004
Cr-Commit-Position: refs/heads/master@{#8983}
With this change, the currently used encoder is held in a scoped_ptr.
iSAC is a special case, since the encoder instance is also a decoder
instance, so it may have to be available also if another send codec is
used. This is accomplished by having a separate scoped_ptr for iSAC.
Remove mirror ID from ACM codec database functions, and remove unused
functions from the database.
COAUTHOR=kwiberg@webrtc.org
BUG=4228
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48729004
Cr-Commit-Position: refs/heads/master@{#8982}
Buildbot Android Tests (L Nexus9)(dbg) consistently fails on Instrumentation test libjingle_peerconnection_android_unittest (VideoCapturerAndroidTest) after this CL was landed.
This reverts commit f4acf46c863f2d516b09b00b39608de7e506ac65.
BUG=
TBR=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45079004
Cr-Commit-Position: refs/heads/master@{#8981}
- Do not handle more than one camera switch request at a time
to avoid blocking camera thread with multiple switch requests.
- Add a callback to notify when camera switch has been done.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46859004
Cr-Commit-Position: refs/heads/master@{#8978}
ViEChannels without default encoders doesn't register a receive codec by
default. This makes VideoReceiver::Decode return early, causing a
high-priority thread to effectively be busy looping. This would be
expected to wreck more havoc in a more cross-platform manner than it has
visibly done. On Windows XP however it manages to bring the whole
machine to a grinding halt forcing a reboot if CPU usage hits 100%.
BUG=chromium:470013
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48049004
Cr-Commit-Position: refs/heads/master@{#8976}
Also prevents that we try to restore audio mode when it has not been changed.
TBR=glaznev
BUG=NONE
TEST=AppRTCDemo and verify that volume control switches from "Ringtone to Phone" mode when call starts and switches back to Ringtone mode when call ends.
Review URL: https://webrtc-codereview.appspot.com/46879004
Cr-Commit-Position: refs/heads/master@{#8975}
It is just a pass through to webrtc/video_frame.h. Updated the callers
to include webrtc/video_frame.h instead and removed i420_video_frame.h.
This should fix pbos' TODO in i420_video_frame.h.
Tested on Linux with the following command lines:
$ rm -rf out/
$ ./webrtc/build/gyp_webrtc
$ ninja -C out/Debug
BUG=None
TEST=see above
R=magjed@webrtc.org, pbos@webrtc.org, tommi@webrtc.orgTBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46819004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8973}
Using the single stream bwe is really bad for the screenshare
test case in particular, but would probably help in other
cases as well so enabling it by default in CallTest setup.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43089004
Cr-Commit-Position: refs/heads/master@{#8971}
The macro is defined as
#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
(WEBRTC_SPL_MUL_16_16(a, b) >> (c))
where the latter macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) \
((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
- minor cleanups like remove of unnecessary parentheses and style changes
- removed commented code lines used during development
- excluded fft.c since there are neon optimizations used and a removal may cause a performance regression
BUG=3348, 3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48799004
Cr-Commit-Position: refs/heads/master@{#8967}
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 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}
