8673 Commits

Author SHA1 Message Date
peah
72a5645fdf Removed the deactivation of the level controller when there is a built-in AGC available
BUG=

Review-Url: https://codereview.webrtc.org/2240763002
Cr-Commit-Position: refs/heads/master@{#13853}
2016-08-22 19:09:02 +00:00
terelius
8c16520949 Method to parse event log directly from a string.
Switches the main parsing function for RtcEventLogs to take an istream instead of a file pointer. Adds wrappers that accept either a string or a filename.

Review-Url: https://codereview.webrtc.org/2253943006
Cr-Commit-Position: refs/heads/master@{#13852}
2016-08-22 18:35:54 +00:00
ehmaldonado
6c46eaa544 Add gtest as a dependency for neteq_quality_test_support.
Was removed in Patch Set 5 of https://codereview.webrtc.org/2252413002
but shouldn't have been, since it's actually required.

https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h?l=17

BUG=webrtc:6228
NOTRY=True

Review-Url: https://codereview.webrtc.org/2262173003
Cr-Commit-Position: refs/heads/master@{#13851}
2016-08-22 16:48:11 +00:00
stefan
d48717b455 Fix issue where the number of packets reported in tests/simulations sometimes are negative.
BUG=webrtc:6159

Review-Url: https://codereview.webrtc.org/2223033002
Cr-Commit-Position: refs/heads/master@{#13850}
2016-08-22 15:50:36 +00:00
kwiberg
4ec01d9c9d Fix trivial lint errors in FileRecorder and FilePlayer
Mostly, it's about replacing mutable reference arguments with pointer
arguments, and replacing C style casts with C++ style casts.

Review-Url: https://codereview.webrtc.org/2056653002
Cr-Commit-Position: refs/heads/master@{#13849}
2016-08-22 15:43:58 +00:00
danilchap
853ecb21f7 Style cleanup in UpdateTmmbr:
function names style updated,
unused return type removed.
Comment style fixed, redundant comments removed.
pass-by-pointer parameter changed to pass-by-value because can't be nullptr any more.

NOTRY=true
BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2258523005
Cr-Commit-Position: refs/heads/master@{#13848}
2016-08-22 15:26:22 +00:00
kwiberg
7f82fc988d WebRtcIlbcfix_Smooth: Fix UBSan fuzzer bug (left shift of 1 by 31 overflows)
scale1 == 31 if and only if w10 == 0. So even though 1 << scale1
overflows, we know that the result of the multiplication should be 0.
Handle that case.

BUG=chromium:615818

Review-Url: https://codereview.webrtc.org/2258543002
Cr-Commit-Position: refs/heads/master@{#13847}
2016-08-22 14:43:50 +00:00
danilchap
642e3bc75b [rtcp] TransportFeedback adjusted to match other rtcp packets:
Derived from rtcp::Rtpfb instead of directly from RtcpPacket
Does not depend on RTCPUtility.
Parse function takes CommonHeader.
TransportFeedback::BlockLength fixed to match size used by Create

BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/1847973003
Cr-Commit-Position: refs/heads/master@{#13846}
2016-08-22 14:37:00 +00:00
henrika
49810511c9 [Reland] Cleanup of the AudioDeviceBuffer class.
See https://codereview.webrtc.org/2256833003/

Contains a minor change to ensure that an external client builds.

TBR=magjed
BUG=NONE

Review-Url: https://codereview.webrtc.org/2269553004
Cr-Commit-Position: refs/heads/master@{#13845}
2016-08-22 12:56:17 +00:00
kjellander
83d79cd4a2 Revert of Add pps id and sps id parsing to the h.264 depacketizer. (patchset #5 id:80001 of https://codereview.webrtc.org/2238253002/ )
Reason for revert:
Breaks some h264 bitstream tests downstream. Reverting for now.

Original issue's description:
> Add pps id and sps id parsing to the h.264 depacketizer.
>
> BUG=webrtc:6208
>
> Committed: https://crrev.com/abcc3de169d8896ad60e920e5677600fb3d40180
> Cr-Commit-Position: refs/heads/master@{#13838}

TBR=sprang@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6208

Review-Url: https://codereview.webrtc.org/2265023002
Cr-Commit-Position: refs/heads/master@{#13844}
2016-08-22 12:34:43 +00:00
nisse
4381700fcc WebRtcVideoFrame constructor without transport_frame_id.
Needed as a substitute when eliminating the Copy method.

BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2262683002
Cr-Commit-Position: refs/heads/master@{#13843}
2016-08-22 10:55:01 +00:00
danilchap
e5b4141746 Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData
BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2249223005
Cr-Commit-Position: refs/heads/master@{#13842}
2016-08-22 10:39:31 +00:00
vopatop.skam
ff101d6171 iOS: add PlistBuddy location to path to avoid build errors
On OS X El Capitan, the system location of 'PlistBuddy' is:
"/usr/libexec/PlistBuddy"
and default system path environment variable is:
"PATH=/usr/local/bin:/usr/bin:/bin:/usr/sbin:/sbin"

NOTRY=True

Review-Url: https://codereview.webrtc.org/2262813002
Cr-Commit-Position: refs/heads/master@{#13841}
2016-08-22 09:27:38 +00:00
peah
4905f06878 Disable the software noise suppressor on iOS devices as that
functionality is always present in the hardware.

BUG=webrtc:6231

Review-Url: https://codereview.webrtc.org/2260173002
Cr-Commit-Position: refs/heads/master@{#13839}
2016-08-22 08:58:56 +00:00
stefan
abcc3de169 Add pps id and sps id parsing to the h.264 depacketizer.
BUG=webrtc:6208

Review-Url: https://codereview.webrtc.org/2238253002
Cr-Commit-Position: refs/heads/master@{#13838}
2016-08-22 08:20:43 +00:00
sakal
86ccd7bfba Revert of Add field_trial_default dependency to libjingle_peerconnection (patchset #3 id:40001 of https://codereview.webrtc.org/2120673004/ )
Reason for revert:
Breaks chromium.

Original issue's description:
> Add field_trial_default dependency to libjingle_peerconnection
>
> This is needed for webrtc::field_trial::FindFullName in peerconnection.cc
>
> NOTRY=True
>
> Committed: https://crrev.com/a7a01df2aebe7108afad208ccd0341c2f0bc7b3b
> Cr-Commit-Position: refs/heads/master@{#13836}

TBR=pthatcher@webrtc.org,pthatcher@chromium.org,kjellander@webrtc.org,arlolra@gmail.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2263063002
Cr-Commit-Position: refs/heads/master@{#13837}
2016-08-22 07:26:11 +00:00
arlolra
a7a01df2ae Add field_trial_default dependency to libjingle_peerconnection
This is needed for webrtc::field_trial::FindFullName in peerconnection.cc

NOTRY=True

Review-Url: https://codereview.webrtc.org/2120673004
Cr-Commit-Position: refs/heads/master@{#13836}
2016-08-22 06:48:14 +00:00
magjed
8177452698 iOS H264VideoToolBoxEncoder: Stop scaling native CVPixelBuffers
If the input to H264VideoToolBoxEncoder is a native CVPixelBuffer and
the quality scaler requests scaling, we fall back to a slow path where
the buffer is converted from NV12 to I420 on the CPU and then uploaded
to a native CVPixelBuffer again. It turns out this scaling is not needed
and that the H264VideoToolBoxEncoder can handle the scaling internally.

BUG=b/30939444

Review-Url: https://codereview.webrtc.org/2258103003
Cr-Commit-Position: refs/heads/master@{#13835}
2016-08-20 17:53:32 +00:00
henrika
d7a89dbe8b Revert of Cleanup of the AudioDeviceBuffer class (patchset #6 id:100001 of https://codereview.webrtc.org/2256833003/ )
Reason for revert:
Seems to break an external client.

Original issue's description:
> Cleanup of the AudioDeviceBuffer class.
>
> WebRTC works on 10ms buffer sizes in both directions but this class has contained
> support for any size (with some limits) and for changes on the fly. It makes no sense to maintain such code and we have no tests to test it. This CL ensures that only 10ms audio buffers are supported and that nothing can be changed on the fly.
>
> It also updates the style to follow the Google C++ style guide.
>
> Finally, I remove very old (not tested and not maintained) support for file
> handling since the code is never used. It was more or less dead code.
>
> BUG=NONE
> R=magjed@webrtc.org
>
> Committed: https://crrev.com/cf327b45b9f5738950d4fca2b6a7b6030d508cdf
> Cr-Commit-Position: refs/heads/master@{#13833}

TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=NONE

Review-Url: https://codereview.webrtc.org/2260183002
Cr-Commit-Position: refs/heads/master@{#13834}
2016-08-19 15:09:29 +00:00
henrika
cf327b45b9 Cleanup of the AudioDeviceBuffer class.
WebRTC works on 10ms buffer sizes in both directions but this class has contained
support for any size (with some limits) and for changes on the fly. It makes no sense to maintain such code and we have no tests to test it. This CL ensures that only 10ms audio buffers are supported and that nothing can be changed on the fly.

It also updates the style to follow the Google C++ style guide.

Finally, I remove very old (not tested and not maintained) support for file
handling since the code is never used. It was more or less dead code.

BUG=NONE
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/2256833003 .

Cr-Commit-Position: refs/heads/master@{#13833}
2016-08-19 14:38:07 +00:00
danilchap
da161d795c Reformat rtcp_receiver
git cl format --full

BUG=webrtc:5565
NOTRY=true

Review-Url: https://codereview.webrtc.org/2259213002
Cr-Commit-Position: refs/heads/master@{#13832}
2016-08-19 14:29:51 +00:00
ehmaldonado
861da3c662 Refactor neteq_test_support.
Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support.

BUG=webrtc:6228
NOTRY=True

Review-Url: https://codereview.webrtc.org/2252413002
Cr-Commit-Position: refs/heads/master@{#13831}
2016-08-19 14:02:31 +00:00
sakal
294fb050a0 Add a timeout for starting the camera on CameraCapturer.
This allows to at least get a camera error back if the camera thread freezes. Application can use this as a signal to restart the program.

R=magjed@webrtc.org

Review-Url: https://codereview.webrtc.org/2257123002
Cr-Commit-Position: refs/heads/master@{#13830}
2016-08-19 10:02:44 +00:00
ehmaldonado
bcba64a0fa GN: Add "//build/config/sanitizers:deps" as a dependency to executable targets.
When the sanitizer bots are switched to GN, this needs to be included as a dependency so that the executables can be compiled.

BUG=webrtc:6215
NOTRY=True

Review-Url: https://codereview.webrtc.org/2250893003
Cr-Commit-Position: refs/heads/master@{#13829}
2016-08-19 09:11:15 +00:00
kthelgason
4a85abb80e Add support for more resolutions on iOS/macOS
BUG=

Review-Url: https://codereview.webrtc.org/2231033002
Cr-Commit-Position: refs/heads/master@{#13828}
2016-08-19 08:24:49 +00:00
kjellander
ec5c9061c8 GN: Fix errors when some variables are set to non-default values.
BUG=webrtc:6223
TESTED=Passing generation with:
gn gen out/Default --args='rtc_build_expat=false rtc_build_json=false rtc_build_libyuv=false'
NOTRY=True

Review-Url: https://codereview.webrtc.org/2257753002
Cr-Commit-Position: refs/heads/master@{#13827}
2016-08-19 08:07:33 +00:00
kjellander
72333d2ca0 Add kjellander@webrtc.org to more BUILD.gn OWNERS files.
NOTRY=True

Review-Url: https://codereview.webrtc.org/2258983003
Cr-Commit-Position: refs/heads/master@{#13826}
2016-08-19 07:48:39 +00:00
vopatop.skam
96b6b8336a iOS: add type to peer connection local streams
BUG=

Review-Url: https://codereview.webrtc.org/2249173002
Cr-Commit-Position: refs/heads/master@{#13825}
2016-08-18 21:21:27 +00:00
Taylor Brandstetter
9b5306c4ef Adding test for unordered, fragmented SCTP message delivery.
This functionality broke after a recent usrsctp roll. This test would be
useful in catching issues that arise in the future.

BUG=633959
R=honghaiz@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2233033002 .

Cr-Commit-Position: refs/heads/master@{#13823}
2016-08-18 18:40:45 +00:00
peah
b5b30908dc Corrected the testvectors for the level controller
bitexactness test. The activation of the test will
be done in another CL.

BUG=

Review-Url: https://codereview.webrtc.org/2257733002
Cr-Commit-Position: refs/heads/master@{#13822}
2016-08-18 16:47:52 +00:00
isheriff
8df4d0e426 Add playout_delay_oracle_unittest as gn target
BUG=

Review-Url: https://codereview.webrtc.org/2256743002
Cr-Commit-Position: refs/heads/master@{#13821}
2016-08-18 14:53:44 +00:00
maxmorin
3a11933a63 Remove audio_device_test_func.
This code does not work and hasn't been used in a long time. It also
lacks a GN target. There's no reason to save it.

BUG=none

Review-Url: https://codereview.webrtc.org/2255173002
Cr-Commit-Position: refs/heads/master@{#13820}
2016-08-18 14:20:48 +00:00
peah
644fa96886 Added recording of the configuration for the AudioFrame API call
BUG=webrtc:6227

Review-Url: https://codereview.webrtc.org/2252043003
Cr-Commit-Position: refs/heads/master@{#13819}
2016-08-18 13:48:38 +00:00
minyue
7320866091 Reland of Adding audio to video_quality_test.
The original commit was https://codereview.webrtc.org/2136573002/.

BUG=

Review-Url: https://codereview.webrtc.org/2259783002
Cr-Commit-Position: refs/heads/master@{#13818}
2016-08-18 13:28:59 +00:00
danilchap
2b616397de Remove TMMBRSet class
by cleaning RTCPReceiveInfo class
and following cleaning of RTCPReceiver::BoundingSet function.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2254703003
Cr-Commit-Position: refs/heads/master@{#13817}
2016-08-18 13:17:48 +00:00
ossu
e1f5b4a7fe voice_engine: Removed old variants of Channel constructor and CreateChannel
These are no longer used internally and their interface is not to be
considered public. They were due to be changed in
https://codereview.webrtc.org/1993783002/ but remained due to a
misunderstanding.

Review-Url: https://codereview.webrtc.org/2082483003
Cr-Commit-Position: refs/heads/master@{#13816}
2016-08-18 11:23:04 +00:00
henrik.lundin
38d840c35a NetEq: Changing checked_cast to saturated_cast
The cast involves packet_len_samp, which is derived from the timestamps
and sequence numbers of incoming packets. Being values from the outside,
these should be treated as if any value is possible, making a
checked_cast unsuitable. Changing instead to saturated_cast to avoid
overflow with out-of-bounds values.

Review-Url: https://codereview.webrtc.org/2243403007
Cr-Commit-Position: refs/heads/master@{#13815}
2016-08-18 10:49:41 +00:00
kwiberg
96bbdd585e WebRtcSpl_SynthesisQMF: Fix UBSan fuzzer bug (left shift of negative value)
BUG=chromium:614033

Review-Url: https://codereview.webrtc.org/2253943002
Cr-Commit-Position: refs/heads/master@{#13814}
2016-08-18 10:17:10 +00:00
peah
e9a6acfbf5 Added missing unittest to the modules/BUILD.gn build file
NOTRY=True

BUG=

Review-Url: https://codereview.webrtc.org/2255093002
Cr-Commit-Position: refs/heads/master@{#13813}
2016-08-18 09:41:51 +00:00
kjellander
cb2d701946 Add kjellander as BUILD.gn OWNER in webrtc/modules
NOTRY=True

Review-Url: https://codereview.webrtc.org/2258593003
Cr-Commit-Position: refs/heads/master@{#13812}
2016-08-18 09:39:14 +00:00
danilchap
71fead2146 Reland of StartTimestamp generated randomly in RtpSender constructor (patchset #1 id:1 of https://codereview.webrtc.org/2248413002/ )
Reason for revert:
Reland: downstream code expectation about rtp_sender timestamp adjusted.

Original issue's description:
> Revert of StartTimestamp generated randomly in RtpSender constructor (patchset #4 id:60001 of https://codereview.webrtc.org/2241193002/ )
>
> Reason for revert:
> Breaks downstream code.
>
> Original issue's description:
> > StartTimestamp generated randomly in RtpSender constructor
> > instead of not-randomly at SetSendingState(true)
> > Renamed to timestamp_offset_ to better match meaning of the variable.
> >
> > R=asapersson@webrtc.org, terelius@webrtc.org
> >
> > Committed: https://crrev.com/4466782ae43e1b1125a55ee7e18abd10dd37cede
> > Cr-Commit-Position: refs/heads/master@{#13796}
>
> TBR=asapersson@webrtc.org,terelius@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/86c96948e340cf8b879bddb0c7293f3b5ad4dad4
> Cr-Commit-Position: refs/heads/master@{#13798}

TBR=asapersson@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2257083002
Cr-Commit-Position: refs/heads/master@{#13811}
2016-08-18 09:02:16 +00:00
ossu
d4e9f62ea7 Updated AudioDecoderFactory to list AudioCodecSpecs instead of SdpAudioFormats.
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2123923004
Cr-Commit-Position: refs/heads/master@{#13810}
2016-08-18 09:02:15 +00:00
magjed
235020dba6 Roll chromium_revision 915e47250f..e3860bd297 (412201:412289)
Change log: 915e47250f..e3860bd297
Full diff: 915e47250f..e3860bd297

No dependencies changed.
No update to Clang.

NOTRY=TRUE
TBR=
BUG=webrtc:6219

Review-Url: https://codereview.webrtc.org/2253973002
Cr-Commit-Position: refs/heads/master@{#13809}
2016-08-18 08:45:53 +00:00
sakal
010f092919 GN: Add Android support to video_engine_tests.
R=kjellander@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2246423002
Cr-Commit-Position: refs/heads/master@{#13808}
2016-08-18 07:42:05 +00:00
Honghai Zhang
fd16da290c Do not switch to a high-cost connection that is not receiving.
This prevents connection switching due to remote-side network down.

R=deadbeef@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2232563002 .

Cr-Commit-Position: refs/heads/master@{#13807}
2016-08-17 23:12:58 +00:00
tkchin
41a3287472 Nil out EAGLContext explicitly on RTCEAGLVideoView dealloc.
Theoretical fix to address some EAGLContext issues from other UIImageViews that could be active.

NOTRY=True
BUG=

Review-Url: https://codereview.webrtc.org/2259513002
Cr-Commit-Position: refs/heads/master@{#13806}
2016-08-17 23:03:09 +00:00
Alex Glaznev
869dab775c Disable Intel VP8 HW encoder.
Need to investigate dequeueOutputBuffer failure on Asus
Zenfones before re-enabling back.

BUG=b/30890961
R=jiayl@chromium.org

Review URL: https://codereview.webrtc.org/2249743007 .

Cr-Commit-Position: refs/heads/master@{#13805}
2016-08-17 22:41:22 +00:00
noahric
6a35590d14 Add code for dummy file audio to fallback to dummy audio.
BUG=

Review-Url: https://codereview.webrtc.org/2250853002
Cr-Commit-Position: refs/heads/master@{#13804}
2016-08-17 22:19:55 +00:00
Alex Glaznev
7c0f8ee67a Avoid null pointer exception if Android getCameraInfo fails.
BUG=b/30890971
R=magjed@webrtc.org, sakal@webrtc.org

Review URL: https://codereview.webrtc.org/2250283002 .

Cr-Commit-Position: refs/heads/master@{#13803}
2016-08-17 22:18:27 +00:00
noahric
d8a72f0ab2 Close input file in FileAudioDevice::StopRecording.
Also added some more logging, to help track down start/stop, start
failure, and the name of the file used.

BUG=

Review-Url: https://codereview.webrtc.org/2253763002
Cr-Commit-Position: refs/heads/master@{#13802}
2016-08-17 22:14:57 +00:00