3672 Commits

Author SHA1 Message Date
kjellander
5aa2d344d7 Revert of Use initial bitrates for software VP8. (patchset #3 id:40001 of https://codereview.webrtc.org/1893313002/ )
Reason for revert:
Likely broke Chromium:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Tester/builds/26838
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win10%20Tester/builds/2224

Original issue's description:
> Use initial bitrates for software VP8.
>
> Makes the software encoder start at VGA as well, since ~300k isn't good
> enough to produce a good HD stream.
>
> BUG=webrtc:5678
> R=glaznev@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/e1da27e543bdb1983638118172a4efd599ca51b5
> Cr-Commit-Position: refs/heads/master@{#12428}

TBR=stefan@webrtc.org,glaznev@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5678

Review URL: https://codereview.webrtc.org/1898183002

Cr-Commit-Position: refs/heads/master@{#12430}
2016-04-19 15:18:50 +00:00
Peter Boström
e1da27e543 Use initial bitrates for software VP8.
Makes the software encoder start at VGA as well, since ~300k isn't good
enough to produce a good HD stream.

BUG=webrtc:5678
R=glaznev@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1893313002 .

Cr-Commit-Position: refs/heads/master@{#12428}
2016-04-19 13:53:22 +00:00
nisse
cc23b7c1ea Delete unused methods SetStartImage and SetTimeoutImage.
Declared in webrtc::VideoRender, implemented in IncomingVideoStream.
This cl also eliminates some of the few uses of
webrtc::VideoFrame::CopyFrame.

BUG=webrtc:5682

Review URL: https://codereview.webrtc.org/1885323002

Cr-Commit-Position: refs/heads/master@{#12427}
2016-04-19 13:19:47 +00:00
Per
ba7dc723b0 Add rotation to EncodedImage and make sure it is passed through encoders.
This fix a potential race where the rotation information of a sent frame does not match the encoded frame.

BUG=webrtc:5783
TEST= Run ApprtcDemo on IOs and Android with and without capture to texture and both VP8 and H264.
R=magjed@webrtc.org, pbos@webrtc.org, tkchin@webrtc.org
TBR=tkchin_webrtc // For IOS changes.

Review URL: https://codereview.webrtc.org/1886113003 .

Cr-Commit-Position: refs/heads/master@{#12426}
2016-04-19 13:01:32 +00:00
kwiberg
0fa0a97cf3 NetEq: Simplify DecoderDatabase::DecoderInfo
By eliminating one of the two constructors, handling decoder ownership
with a unique_ptr instead of a raw pointer, and making all member
variables const (except one, which is made private instead).

BUG=webrtc:5801

Review URL: https://codereview.webrtc.org/1899733002

Cr-Commit-Position: refs/heads/master@{#12425}
2016-04-19 12:03:51 +00:00
peah
f3669661bd Removed the issue with the leading semicolon in the audio
processing module experiment description that was present
when AEC3 was not activated and when RefinedAdaptiveFilter
was activated.

BUG=webrtc:5778, webrtc:5777

Review URL: https://codereview.webrtc.org/1899123002

Cr-Commit-Position: refs/heads/master@{#12424}
2016-04-19 10:40:15 +00:00
Danil Chapovalov
ee6e4272a4 Fixed undefined shift in parsing Tmmbr, Tmmbn and Remb
BUG=chromium:603483
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1888793003 .

Cr-Commit-Position: refs/heads/master@{#12423}
2016-04-19 10:15:21 +00:00
kjellander
470dd37b41 Roll chromium_revision 212f976fef..61ed337cfe (387882:388120)
https://codereview.chromium.org/1826693002 enables some
more Clang warnings which were fixed.

Change log: 212f976fef..61ed337cfe
Full diff: 212f976fef..61ed337cfe

No dependencies changed.
No update to Clang.

TBR=
NOTRY=True

Review URL: https://codereview.webrtc.org/1896953004

Cr-Commit-Position: refs/heads/master@{#12422}
2016-04-19 10:03:31 +00:00
pbos
a96b60b3a6 Move frame_callback.h to common_video/include.
BUG=webrtc:4243
R=kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1902543002

Cr-Commit-Position: refs/heads/master@{#12419}
2016-04-19 04:12:57 +00:00
Peter Boström
b9e77097ed Add QVGA to thresholds for initial quality.
Makes QualityScaler start at QVGA for <250k initial bitrates. Useful in
combination with overriding max bitrates to a max lower than that for
connections where we know that the max bitrate is capped below where VGA
is useful.

BUG=webrtc:5678
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1900483004 .

Cr-Commit-Position: refs/heads/master@{#12416}
2016-04-18 20:46:06 +00:00
danilchap
d6b851a1bd Fixed memleak when two voip blocks present in single rtcp packet.
BUG=chromium:603894

Review URL: https://codereview.webrtc.org/1901593002

Cr-Commit-Position: refs/heads/master@{#12413}
2016-04-18 17:54:13 +00:00
ossu
264087f45a A few small cleanups of stuff caught by lint
Review URL: https://codereview.webrtc.org/1871003002

Cr-Commit-Position: refs/heads/master@{#12412}
2016-04-18 15:07:33 +00:00
ossu
2903ba5ff3 Reland Remove the deprecated EncodeInternal interface from AudioEncoder
Remove the deprecated EncodeInternal interface from AudioEncoder

Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.

BUG=webrtc:5591

Review URL: https://codereview.webrtc.org/1881003003

Cr-Commit-Position: refs/heads/master@{#12409}
2016-04-18 13:14:42 +00:00
Stefan Holmer
54728bab25 Remove process thread checker from BWE.
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1898723002 .

Cr-Commit-Position: refs/heads/master@{#12408}
2016-04-18 13:06:16 +00:00
Peter Boström
2c8a2964fd Tune QP-based quality thresholds.
Increases measure time for downscale back to 5 seconds, this is required
to not over-react on hand-waving or quick device rotations.

Also increase max thresholds for QP a bit to not overreact when quality
still looks somewhat OK. Min thresholds for H264 seemed very low and are
increased to be sure that we can go back up again. The window is still
quite big with the increased max QP.

Also changes libvpx thresholds to use the same thresholds as the
encoder, they were excessively low before and wouldn't adapt on bad QPs
at all before (but rely on >60% framedropping based on bitrates to go
down).

BUG=webrtc:5678
R=stefan@webrtc.org
TBR=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1894083002 .

Cr-Commit-Position: refs/heads/master@{#12403}
2016-04-18 10:58:17 +00:00
asapersson
5265fedffe Add histogram stats for average QP per frame for VP9 (for sent video streams):
- "WebRTC.Video.Encoded.Qp.Vp9"
- "WebRTC.Video.Encoded.Qp.Vp9.S0"
- "WebRTC.Video.Encoded.Qp.Vp9.S1"
- "WebRTC.Video.Encoded.Qp.Vp9.S2"

BUG=

Review URL: https://codereview.webrtc.org/1870043002

Cr-Commit-Position: refs/heads/master@{#12402}
2016-04-18 09:58:52 +00:00
Peter Boström
8056acc6f5 Use bitstream-level QP for libvpx VP8 quality.
BUG=webrtc:5678
TBR=marpan@webrtc.org

Review URL: https://codereview.webrtc.org/1888843002 .

Cr-Commit-Position: refs/heads/master@{#12401}
2016-04-18 09:17:43 +00:00
asapersson
a186288fd0 Revert of Update histogram "WebRTC.Video.OnewayDelayInMs" to use the estimated one-way delay. (patchset #4 id:60001 of https://codereview.webrtc.org/1688143003/ )
Reason for revert:
The delay stats are high.

Original issue's description:
> Update histogram "WebRTC.Video.OnewayDelayInMs" to use the estimated one-way delay.
> Previous logged delay was: network delay (rtt/2) + jitter delay + decode time + render delay.
>
> Make capture time in local timebase available for decoded VP9 video frames (propagate ntp_time_ms from EncodedImage to decoded VideoFrame).
>
> BUG=
>
> Committed: https://crrev.com/5249599a9b69ad9c2d513210d694719f1011f977
> Cr-Commit-Position: refs/heads/master@{#11901}

TBR=stefan@webrtc.org,pbos@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:603838

Review URL: https://codereview.webrtc.org/1893543003

Cr-Commit-Position: refs/heads/master@{#12400}
2016-04-18 07:41:09 +00:00
kjellander
e532aec252 Add isolate files for Android tests
BUG=chromium:583318
TESTED=Passing runs with:
GYP_DEFINES='test_isolation_mode=prepare OS=android' webrtc/build/gyp_webrtc
ninja -C out/Release
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1882963003

Cr-Commit-Position: refs/heads/master@{#12397}
2016-04-18 03:08:28 +00:00
jackychen
e42c0ae040 Display moving object detection result on Nexus for debugging.
Review URL: https://codereview.webrtc.org/1890183003

Cr-Commit-Position: refs/heads/master@{#12390}
2016-04-16 17:44:23 +00:00
peah
594a877f2d Cleaned up the EchoSuppression method in the AEC so that it
does not have to use the aec state as an input.

Furthermore, the debug dump output of e_fft was removed as
it is not really used in any analysis scripts.

BUG=webrtc:5298

Review URL: https://codereview.webrtc.org/1883293003

Cr-Commit-Position: refs/heads/master@{#12387}
2016-04-16 11:04:04 +00:00
peah
0332c2db39 Added support in the AEC for refined filter adaptation.
The following algorithmic functionality was added:
-Add support for an exact regressor power to be computed
 which avoids the issue with the updating of the filter
 sometimes being unstable.
-Lowered the fixed step size of the adaptive filter to 0.05
 which significantly reduces the sensitivity of the
 adaptive filter to near-end noise, nonlinearities,
 doubletalk and the unmodelled echo path tail. It also
 reduces the tracking speed of the adaptive filter but the
 chosen value proved to give a sufficient tradeoff for the
 requirements on the adaptive filter.

To allow the new functionality to be selectively applied the following was done:
-A new Config was added for selectively activating the functionality.
-Functionality was added in the audioprocessing  and echocancellationimpl classes
 for passing the activation of the functionality down to the AEC algorithms.

To make the code for the introduction of the functionality clean,
the following refactoring was done:
-The selection of the step size was moved to a single place.
-The constant for the step size of the adaptive filter in extended filter mode was
 made local.
-The state variable storing the step-size was renamed to a more describing name.

When the new functionality is not activated, the changes
have been tested for bitexactness on Linux.

TBR=minyue@webrtc.org
BUG=webrtc:5778, webrtc:5777

Review URL: https://codereview.webrtc.org/1887003002

Cr-Commit-Position: refs/heads/master@{#12384}
2016-04-15 18:23:36 +00:00
Per
83d0910694 Move Ownership of RtpModules to VideoSendStream from VieChannel and remove use of vie_channel and vie_receiver from video_send_stream.
The purpose of this refactoring is a first step of separating the encoder parts from the RTP transport.

BUG=webrtc:5687
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1864313003 .

Cr-Commit-Position: refs/heads/master@{#12377}
2016-04-15 12:59:21 +00:00
kwiberg
6ca0a31708 We no longer use compilers that can't =default move construction and assignment
Review URL: https://codereview.webrtc.org/1891483006

Cr-Commit-Position: refs/heads/master@{#12376}
2016-04-15 12:25:03 +00:00
nisse
26acec4772 Delete method webrtc::VideoFrame::native_handle.
Instead, use the corresponding method on VideoFrameBuffer. In the process,
reduce code duplication in frame comparison functions used in
the test code.

Make FramesEqual use FrameBufsEqual. Make the latter support texture frames.

The cl also refactors VideoFrame::CopyFrame to use I420Buffer::Copy. This
has possibly undesired side effects of never reusing the frame buffer of
the destination frame, and producing a frame buffer which may use different
stride than the source frame.

BUG=webrtc:5682

Review URL: https://codereview.webrtc.org/1881953002

Cr-Commit-Position: refs/heads/master@{#12373}
2016-04-15 10:43:45 +00:00
sprang
3911c26bc0 Add support for writing raw encoder output to .ivf files.
Also refactor GenericEncoder to use these file writers, and remove use
of preprocessor to enable file writing.

BUG=

Review URL: https://codereview.webrtc.org/1853813002

Cr-Commit-Position: refs/heads/master@{#12372}
2016-04-15 08:24:21 +00:00
peah
7789fe7ab1 Added a protobuf field for the audio processing module to store the status of temporary experimental features that
are active in the module and its submodules.

BUG=webrtc:5778, webrtc:5777

Review URL: https://codereview.webrtc.org/1886233003

Cr-Commit-Position: refs/heads/master@{#12371}
2016-04-15 08:19:47 +00:00
stefan
1112b2bc68 Fix bug when the BWE times out due to no incoming packets.
Both InterArrival and OveruseEstimator should be timed out at the same time since otherwise the overuse filter may take a long time to converge.

BUG=webrtc:5773

Review URL: https://codereview.webrtc.org/1886783002

Cr-Commit-Position: refs/heads/master@{#12364}
2016-04-14 15:08:20 +00:00
Peter Boström
00b62b0849 Remove QualityScaler kDefaultLowQpDenominator.
This denominator doesn't make any semantic sense, it's better to use
real thresholds for when things look "good" or "bad" rather than
fractions of a max QP.

BUG=webrtc:5678
R=danilchap@webrtc.org

Review URL: https://codereview.webrtc.org/1855393005 .

Cr-Commit-Position: refs/heads/master@{#12363}
2016-04-14 15:04:10 +00:00
Peter Boström
926dfcdf5e Make QualityScaler not downscale below QVGA.
Applies to all platforms, not only Android now, 160x90 video looks
awful and there's no real point to going below QVGA.

BUG=webrtc:5678
R=danilchap@webrtc.org
TBR=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1867643003 .

Cr-Commit-Position: refs/heads/master@{#12362}
2016-04-14 12:48:18 +00:00
danilchap
7c9426cf38 Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
Review URL: https://codereview.webrtc.org/1877253002

Cr-Commit-Position: refs/heads/master@{#12359}
2016-04-14 10:05:37 +00:00
noahric
91efeecab8 Remove VERBOSE logs from frame_dropper.cc.
They are way too verbose, ~100 lines of log per second.

BUG=

Review URL: https://codereview.webrtc.org/1888453004

Cr-Commit-Position: refs/heads/master@{#12356}
2016-04-14 04:01:48 +00:00
noahric
d4badbcb6d Fix SetRates for encoders with internal sources.
An earlier change moved SetRates to happen on every input frame, but
encoders with internal sources don't receive input frames, so they
weren't getting updated bitrate (and framerate) signals.

BUG=

Review URL: https://codereview.webrtc.org/1682253005

Cr-Commit-Position: refs/heads/master@{#12354}
2016-04-13 21:59:57 +00:00
aluebs
2fae89ed0d Disable Intelligibility Enhancer for high SNRs
Review URL: https://codereview.webrtc.org/1878133002

Cr-Commit-Position: refs/heads/master@{#12352}
2016-04-13 18:24:11 +00:00
peah
21a395ddf7 Moved the aec_rdft*.c files to be build using C++
BUG=webrtc:5298
NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1881153003

Cr-Commit-Position: refs/heads/master@{#12346}
2016-04-13 14:53:57 +00:00
Peter Boström
eda7926d97 Add pbos@webrtc.org to video_coding OWNERS.
BUG=
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1865303002 .

Cr-Commit-Position: refs/heads/master@{#12345}
2016-04-13 12:59:48 +00:00
peah
3eeb2e89b3 Moved the audioprocessing unittest to the audio_processing folder
where the other audioprocessing unittests are located.

BUG=webrtc:5298

Review URL: https://codereview.webrtc.org/1846323002

Cr-Commit-Position: refs/heads/master@{#12343}
2016-04-13 11:10:09 +00:00
pbos
cbac40d321 Reland of Make QualityScaler more responsive to downgrades. (patchset #1 id:1 of https://codereview.webrtc.org/1880103002/ )
Reason for revert:
Regressed behavior is actually desirable (go down to 360p instead of producing super-bad 720p).

Original issue's description:
> Revert of Make QualityScaler more responsive to downgrades. (patchset #3 id:40001 of https://codereview.webrtc.org/1830593003/ )
>
> Reason for revert:
> Speculative revert: want to see if this causes the regression in https://crbug.com/602621
>
> Original issue's description:
> > Make QualityScaler more responsive to downgrades.
> >
> > Permits going from HD to QVGA in 6 seconds instead of 10. Also adds
> > windows for going up quickly in the beginning of a call (before any
> > downscaling happens due to bad quality).
> >
> > BUG=webrtc:5678
> > R=glaznev@webrtc.org, stefan@webrtc.org
> >
> > Committed: https://crrev.com/85829fd90cc4e7a91c9857921b19e8fc126aeb60
> > Cr-Commit-Position: refs/heads/master@{#12219}
>
> TBR=glaznev@webrtc.org,stefan@webrtc.org,pbos@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5678
> NOTRY=true
>
> Committed: https://crrev.com/19b4fecf08e3fe215e431a260fb673553c15e569
> Cr-Commit-Position: refs/heads/master@{#12331}

TBR=glaznev@webrtc.org,stefan@webrtc.org,phoglund@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:602621, webrtc:5678

Review URL: https://codereview.webrtc.org/1887493003

Cr-Commit-Position: refs/heads/master@{#12341}
2016-04-13 09:51:10 +00:00
jackychen
afaae0d151 External VNR speed improvement.
Improved visual quality with 3x times speed-up.
Change list:
 1. Remove second chance filter in temporal denoising filter to mitigate trailing artifact.
 2. Add swap buffer to save one whole-frame memcpy.
 3. Do noise estimation on every N blocks.
 4. Adopt a faster moving object detection algorithm (change the structure).
 5. Refactor the for loops and PositionCheck().
 6. Refactor the function ReduceFalseDetection (RFD).
 7. Fix a bug in TrailingBlock() which causes a mismatch.
 8. Change unit test to support swap buffer test.
 9. Remove CopyMem8x8, use memcpy to copy U/V plane which can be optimized future.
 10. Remove DenoiseMetrics.

Review URL: https://codereview.webrtc.org/1871853003

Cr-Commit-Position: refs/heads/master@{#12340}
2016-04-13 06:03:11 +00:00
peah
bdb7af692f Changed the delay estimator to be built using C++
BUG=webrtc:5724
NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1878613002

Cr-Commit-Position: refs/heads/master@{#12336}
2016-04-12 21:47:46 +00:00
guidou
63911495de Revert of Fix screen capturers to initialize on the same thread on which Start() is called. (patchset #3 id:80001 of https://codereview.webrtc.org/1861893002/ )
Reason for revert:
This is preventing a WebRTC roll into Chromium. See https://codereview.chromium.org/1877263003/

Original issue's description:
> Fix screen capturers to initialize on the same thread on which Start() is called.
>
> Previously screen capturers were initialized when they are created.
> This means that in the CRD host they were initialized on the thread
> that's different from the thread on which they are used. Because of this
> on Linux the host was using XErrorTrap() on two different threads and
> this is not supported. Now ScreenCapturer implementations always
> initialize themselves on the thread on which Start() is called.
>
> Also added ThreadChecker to make sure the capturers are always called
> from the same thread.
>
> BUG=600432
>
> Committed: https://crrev.com/e8d4b7d8a3d298438a2ebd9ee8d5aa71f42cf033
> Cr-Commit-Position: refs/heads/master@{#12285}

TBR=jamiewalch@chromium.org,sergeyu@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=600432

Review URL: https://codereview.webrtc.org/1882083002

Cr-Commit-Position: refs/heads/master@{#12334}
2016-04-12 18:42:07 +00:00
phoglund
19b4fecf08 Revert of Make QualityScaler more responsive to downgrades. (patchset #3 id:40001 of https://codereview.webrtc.org/1830593003/ )
Reason for revert:
Speculative revert: want to see if this causes the regression in https://crbug.com/602621

Original issue's description:
> Make QualityScaler more responsive to downgrades.
>
> Permits going from HD to QVGA in 6 seconds instead of 10. Also adds
> windows for going up quickly in the beginning of a call (before any
> downscaling happens due to bad quality).
>
> BUG=webrtc:5678
> R=glaznev@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/85829fd90cc4e7a91c9857921b19e8fc126aeb60
> Cr-Commit-Position: refs/heads/master@{#12219}

TBR=glaznev@webrtc.org,stefan@webrtc.org,pbos@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5678
NOTRY=true

Review URL: https://codereview.webrtc.org/1880103002

Cr-Commit-Position: refs/heads/master@{#12331}
2016-04-12 16:06:02 +00:00
ossu
164bc4bbd3 Revert of Remove the deprecated EncodeInternal interface from AudioEncoder (patchset #4 id:60001 of https://codereview.webrtc.org/1864993002/ )
Reason for revert:
Broke import. Implementations of the old interface still exists somewhere.

Original issue's description:
> Remove the deprecated EncodeInternal interface from AudioEncoder
>
> Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.
>
> BUG=webrtc:5591
>
> Committed: https://crrev.com/5222d315dbea8f3563c100cc9f2451907f70b05f
> Cr-Commit-Position: refs/heads/master@{#12329}

TBR=kwiberg@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5591

Review URL: https://codereview.webrtc.org/1883543002

Cr-Commit-Position: refs/heads/master@{#12330}
2016-04-12 10:58:10 +00:00
ossu
5222d315db Remove the deprecated EncodeInternal interface from AudioEncoder
Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.

BUG=webrtc:5591

Review URL: https://codereview.webrtc.org/1864993002

Cr-Commit-Position: refs/heads/master@{#12329}
2016-04-12 10:31:03 +00:00
sprang
afe1f74c04 Make sure temporal layered screenshare frames are sent in at least 2s.
If a very large frame is sent (high res slide change) when the available
send bitrate is very low, the it might take many seconds before any new
frames are emitted as the accrued debt will take time to pay off.

Add a bailout, so that if a frame hasn't been sent for 2 seconds, cancel
the debt immediately, even if the target bitrate is then exceeded.

BUG=webrtc:5750

Review URL: https://codereview.webrtc.org/1869003002

Cr-Commit-Position: refs/heads/master@{#12328}
2016-04-12 09:45:20 +00:00
peah
b97526ed65 Corrected include of C++ header file in AECM that was done using external inclusion
BUG=

Review URL: https://codereview.webrtc.org/1876143002

Cr-Commit-Position: refs/heads/master@{#12327}
2016-04-12 08:20:47 +00:00
Marco
7f315883f0 vp8-intergationtest: Adjust a parameter in resize test.
Needed for new libvpx roll.

TBR=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1877113003 .

Cr-Commit-Position: refs/heads/master@{#12324}
2016-04-12 04:02:51 +00:00
tkchin
efdd930dc9 Fix RTCAudioSession crash in removeDelegate.
BUG=

Review URL: https://codereview.webrtc.org/1877643002

Cr-Commit-Position: refs/heads/master@{#12320}
2016-04-11 19:01:06 +00:00
tkchin
9ff9c6540b iOS h264: Request keyframe after coming back from background.
BUG=

Review URL: https://codereview.webrtc.org/1877613002

Cr-Commit-Position: refs/heads/master@{#12319}
2016-04-11 18:40:36 +00:00
danilchap
90a1351072 Fixed rtcp rpsi parsing of invalid packets.
Added packet type RpsiItem to destinguish parsed rpsi header and Rpsi body
  preventing handling two half-valid (header-only) rpsi packets as one valid,
  making test parser calculate rpsi packet once instead of twice.
Added check padding bits doesn't exceed native bit string length.
Marking rpsi received moved after it is validated.

BUG=600977

Review URL: https://codereview.webrtc.org/1880443002

Cr-Commit-Position: refs/heads/master@{#12318}
2016-04-11 17:05:06 +00:00