The old OpenSSL threading hooks were removed in favor of the library knowing
about threads internally. Instead of CRYPTO_add, use FOO_up_ref wrappers that
don't require reaching into the type.
BUG=none
R=jiayl@webrtc.org, juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/54579004
Cr-Commit-Position: refs/heads/master@{#9324}
Use the current parameter names in the comment for SetNackMode().
Add a warning comment about the lifetime of the return value of
GetNackList().
R=stefan@webrtc.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/52599004
Cr-Commit-Position: refs/heads/master@{#9321}
The most important change is to prevent a potential buffer overflow in
NackList(). It cannot happen if the |size| argument passed to NackList()
is consistent with the |max_nack_list_size| argument passed to
SetNackSettings(), and there is an assertion to check that. But it is
good to defend against this in the release build because assert() is
compiled away in the release build.
Remove the unused |master| parameter to the VCMReceiver constructor.
Remove the unused State() getter method and the corresponding state_
member.
Remove the declarations for the nonexistent GenerateReceiverId()
method and the receiver_id_counter_ member.
Remove the unneeded data_buffer_ member of TestVCMReceiver. It was
assigned to packet.dataPtr and then immediately overwritten by
stream_generator_->GetPacket() or stream_generator_->PopPacket().
R=stefan@webrtc.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/51119004
Cr-Commit-Position: refs/heads/master@{#9318}
Reduces musical noise with minimal impact on interferer suppression.
This also unifies the treatment of "mean bins". The "end" bin is now
inclusive in the mean range as with the "start" bin.
Corrects interpretation of quantile.
BUG=chromium:490477
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50939004
Cr-Commit-Position: refs/heads/master@{#9317}
Changed Create() to return a pointer to the object rather than an error message, which is in line with how objects should be created.
BUG=441
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51939004
Cr-Commit-Position: refs/heads/master@{#9315}
Registers transport on construction removing the need for ViESender as a
hop and removing a potential deadlock by removing RegisterSendTransport.
BUG=1695, 2999
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/57449004
Cr-Commit-Position: refs/heads/master@{#9309}
Commit 7e0c7d49 ("Add support for external encoders in ACM") changed
things around so that we no longer recreate the speech encoder when
adding CNG or RED to an existing encoder. This isn't correct, since
those two expect to be in sync with the speech encoder they work with.
Solve the problem by resetting the speech encoder before hooking in
RED or CNG.
BUG=crbug/490368
R=jmarusic@webrtc.orgTBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53589004
Cr-Commit-Position: refs/heads/master@{#9307}
If srtp_create fails while adding streams, it deallocates the session
but doesn't clear the passed pointer which then could lead to a
double-free in the SrtpSession dtor.
The CL also adds locking for libsrtp initialization / shutdown.
BUG=4042
R=jiayl@webrtc.org, juberti@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47319004
Cr-Commit-Position: refs/heads/master@{#9300}
This change instroduces a mode where the Accelerate operation will be
more aggressive. When enabled, it will allow acceleration at lower
correlation levels, and possibly remove multiple pitch periods at
once.
The feature is enabled through NetEq::Config, and is off by
default. This means that bit-exactness tests are currently not
affected.
A unit test was added for the Accelerate class, with and without fast
mode enabled.
BUG=4691
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50039004
Cr-Commit-Position: refs/heads/master@{#9295}
Changed the WebRtcVad_Create() function to the more conventional format of returning the handle directly instead of an error code to take care of.
In addition NULL was changed to nullptr in the files where it applied.
Affected components:
* AGC
* VAD
* NetEQ
BUG=441, 3347
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51919004
Cr-Commit-Position: refs/heads/master@{#9291}
This reverts commit fc052055e939fa93d3ab92914e0dc8ed5e5d1d90.
since it was not committed correctly.
I committed it from a wrong machine, which did not have the correct patch.
BUG=
TBR=phoglund@webrtc.org,
Review URL: https://webrtc-codereview.appspot.com/56469005
Cr-Commit-Position: refs/heads/master@{#9289}
uint32_t parameters don't need to be passed by reference. The
VCMJitterBuffer destructor doesn't need to be virtual because the
class has no virtual methods.
R=stefan@webrtc.org
BUG=none
Review URL: https://webrtc-codereview.appspot.com/55499004
Cr-Commit-Position: refs/heads/master@{#9288}
In my previous cl, https://webrtc-codereview.appspot.com/52479004/, there is 'UnLock()' left when we switched to scoped lock, which will cause TSan warning sometimes.
===========================================================
WARNING: ThreadSanitizer: unlock of an unlocked mutex (or by a wrong thread) (pid=9981)
#0 pthread_mutex_unlock <null> (libjingle_peerconnection_unittest+0x00000046836f)
#1 webrtc::CriticalSectionPosix::Leave() webrtc/system_wrappers/source/critical_section_posix.cc:39:10 (libjingle_peerconnection_unittest+0x000000bc368d)
#2 ~CriticalSectionScoped webrtc/system_wrappers/interface/critical_section_wrapper.h:46:48 (libjingle_peerconnection_unittest+0x000000a61fcb)
#3 webrtc::AudioDeviceLinuxPulse::RecThreadProcess() webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc:3003 (libjingle_peerconnection_unittest+0x000000a61fcb)
===========================================================
BUG=3056
TEST=bots
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56439004
Cr-Commit-Position: refs/heads/master@{#9282}
BUG=4690
I have removed methods in VoE interfaces that were marked to be removed. I have removed them also in fake and mock implementations. I have also updated the callers in various ways:
1. Project win_test had some calls to the removed methods, but it turned out that the project is not used anymore, so I removed it entirely.
2. There were some calls to removed methods in jni methods. I have removed couple of jni methods as now they seem to do nothing.
3. With the remaining callers I just removed the calls to removed methods.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53519004
Cr-Commit-Position: refs/heads/master@{#9281}
It looks like our basictypes.h file in the overrides folder is including the file it is overriding due to include path precedence (Chrome's is lower than WebRTCs).
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51909004
Cr-Commit-Position: refs/heads/master@{#9280}
Add pylintrc file based on
https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc
bit tightened up quite a bit (the one in depot_tools is far
more relaxed).
Remove a few excluded directories from pylint check and fixed/
suppressed all warnings generated.
Add GN format check + formatted all GN files using 'gn format'.
Cleanup redundant rules in tools/PRESUBMIT.py
TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations.
Ran it again with a modification in webrtc/build/webrtc.gni, formatted
all the GN files and ran it again.
R=henrika@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50069004
Cr-Commit-Position: refs/heads/master@{#9274}
This is being done in preparation of moving base/logging.* to rtc_base_approved. base/stream.* has libjingle dependencies that webrtc can't use, so logging.* can't depend on streams. It does look like stream.* isn't used much, so cleaning that up as well as cleaning up usage of the actual stream support (now LogStream) in the logging code, is in order, but I'll leave that to another cl.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54529004
Cr-Commit-Position: refs/heads/master@{#9269}
I wanted to use Config::Get in Chromium code, but it triggered the following
warning:
../../third_party/webrtc/common.h:89:20: error: declaration requires an exit-time destructor [-Werror,-Wexit-time-destructors]
static const T def;
^
../../third_party/webrtc/common.h:110:10: note: in instantiation of function template specialization requested here
return default_value<T>();
^
I assume we don't hit this in webrtc because the warning is disabled.
This also switches to the RTC_ prefix from the deprecated LIBJINGLE_.
Needed due to this Chromium CL:
https://codereview.chromium.org/1148843004/R=andresp@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53459004
Cr-Commit-Position: refs/heads/master@{#9268}
tl;dr - non-continuous frames (due to padding) would get stuck as incomplete if the previous complete frame arrived and was decoded before the padding arrived.This fix re-checks the incomplete frame list for continuous frames after old packets arrive.
When padding is enabled and RTX is not, padding is sent as empty RTP packets tacked onto the end of completed frames (meaning: same timestamp, but after a packet with the marker bit set). Given the following set of circumstances, codified in the new unit test method, a frame can get permanently stuck in the incomplete frames list:
- Frame A decoded (packets 94-95). Next expected sequence number is 96.
- Frame C arrives (packets 100-101) and is marked complete. It isn't continuous, since it starts at 100, so it's placed in the incomplete frame list.
- Frame B arrives (packets 96-97) and is complete, since 97 has a marker bit. Turns out that packets 98-99 are padding, but the receiver doesn't know that.
- Frame B is decoded, removed from the decodable frames list, and last decoded state is updated.
- Packets 98-99 arrive. They hit the IsOldPacket check and update the last decoded state, but they don't trigger FindAndInsertContinuousFrames.
- Further packets/frames arrive and complete, but FindAndInsertContinuousFrames only runs on frames that are newer than the newly completed frame.
In this state, Frame C is permanently stuck as incomplete, so the jitter buffer overall is stuck until max NACK age (default: 450 packets), the max NACK list size (default: 200 packets), or a keyframe arrives and IsContinuous returns true for the keyframe.
(Before the November refactoring, Frame B wouldn't have to have been decoded for the bug to trigger; just having a complete continuous frame at any time before the padding arrived would cause this state, as FindAndInsertContinuousFrames was only called when the frame originally became continuous and was inserted into the decodable frames list. Post refactoring, the frame is removed/re-added to the decodable list on every padding packet that arrives)
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50959004
Cr-Commit-Position: refs/heads/master@{#9264}
