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Author SHA1 Message Date
magjed@webrtc.org
d4e7d49628 Scaler: Recycle allocations using buffer pool. 
BUG=1128
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49489004

Cr-Commit-Position: refs/heads/master@{#8828}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8828 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 12:28:14 +00:00
henrik.lundin@webrtc.org
09b6ff9460 Disable PLC for iSAC 
A codec's packet-loss concealer is called once from NetEq before
decoding the first packet after a packet loss. The purpose is not to
use the PLC output, but to prepare the state of the decoder such that
it may recover faster after the loss. However, this effect is not
achieved by calling iSAC's PLC. Also, there are some problems with the
fixed-point implementation of the PLC (see the associated bug).

BUG=4423
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42849004

Cr-Commit-Position: refs/heads/master@{#8827}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8827 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 12:24:14 +00:00
kjellander@webrtc.org
ee0c5af314 Remove unused version.py script. 
This should have been removed back in
https://webrtc-codereview.appspot.com/4899004

TBR=andrew@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/48629004

Cr-Commit-Position: refs/heads/master@{#8826}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8826 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 12:20:18 +00:00
jmarusic@webrtc.org
aa0bbab8ec Fix build failure 
There was a build failure due to including checks.h. Removed the include.
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48639004

Cr-Commit-Position: refs/heads/master@{#8825}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8825 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 11:43:14 +00:00
jmarusic@webrtc.org
a4bef3e6c0 AcmReceiver: use std::map instead of an array to keep the list of decoders 
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50419004

Cr-Commit-Position: refs/heads/master@{#8824}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8824 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 11:20:31 +00:00
pbos@webrtc.org
3335a4ffc8 Prevent asserting on unset start bitrate. 
Quick fix for crash in WebRtcBrowserTest.NegotiateOfferWithBLine.
Follow-up CL from stefan@ that changes this path is under review.

BUG=1788,1667
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51419005

Cr-Commit-Position: refs/heads/master@{#8823}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8823 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 09:49:12 +00:00
tommi@webrtc.org
38492c5b6f Re-land 8810 "- Add a SetPriority method to ThreadWr..." 
> Revert 8810 "- Add a SetPriority method to ThreadWrapper"
> Seeing if this is causing roll issues.
> 
> > - Add a SetPriority method to ThreadWrapper
> > - Remove 'priority' from CreateThread and related member variables from implementations
> > - Make supplying a name for threads, non-optional
> > 
> > BUG=
> > R=magjed@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/44729004
> 
> TBR=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/48609004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50459005

Cr-Commit-Position: refs/heads/master@{#8819}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8819 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-22 14:42:46 +00:00
tommi@webrtc.org
90a1cb4630 Revert 8810 "- Add a SetPriority method to ThreadWrapper" 
Seeing if this is causing roll issues.

> - Add a SetPriority method to ThreadWrapper
> - Remove 'priority' from CreateThread and related member variables from implementations
> - Make supplying a name for threads, non-optional
> 
> BUG=
> R=magjed@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/44729004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48609004

Cr-Commit-Position: refs/heads/master@{#8818}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8818 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-22 14:34:46 +00:00
braveyao@webrtc.org
346a64b9b5 Mac would force bluetooth playout working with 8kHz/1ch if capturing/rendering shares the same device, e.g. changing from 44.1kHz/2ch as default. 
So in the HandleStreamFormatChange() callback, we need to re-initiate the playout as same as what we do in InitPlayout(). Here we merely copy those codes out from InitPlayout() into a new SetDesiredPlayoutFormat() function for the invoking from the two places.
Previously, HandleStreamFormatChange only re-creates the AudioConverter, which is not enough. We also need to reset the buffer size and refresh the latency.

BUG=4240
TEST=Manual Test
R=andrew@webrtc.org, henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36029004

Cr-Commit-Position: refs/heads/master@{#8815}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-21 01:06:14 +00:00
wtc@chromium.org
4553941d32 Document the 'int' return value of Resampler methods. 
Remove an obsolete TODO comment.

R=andrew@webrtc.org
BUG=none

Review URL: https://webrtc-codereview.appspot.com/48589004

Cr-Commit-Position: refs/heads/master@{#8814}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 23:28:39 +00:00
andrew@webrtc.org
3200a64b3c Minor fix for MIPS Android build. 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47729004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

Cr-Commit-Position: refs/heads/master@{#8813}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 22:55:43 +00:00
pbos@webrtc.org
b9557a9bb7 Fix code to handle crashes for non-VP8. 
Unit tests will be submitted Monday, submitting this part to get the
Android bots green.

BUG=1667, 1788
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44789004

Cr-Commit-Position: refs/heads/master@{#8811}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8811 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 19:53:15 +00:00
tommi@webrtc.org
b6817d793f - Add a SetPriority method to ThreadWrapper 
- Remove 'priority' from CreateThread and related member variables from implementations
- Make supplying a name for threads, non-optional

BUG=
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44729004

Cr-Commit-Position: refs/heads/master@{#8810}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8810 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 15:52:43 +00:00
pbos@webrtc.org
a3209a2b27 Release buffer pool in Vp8DecoderImpl::Release(). 
Permits reusing an external VP8DecoderImpl instance from another
VideoReceiveStream without a thread-checker DCHECK blowing up. Also
releases buffers that would've been kept in memory even though the
decoder isn't configured.

BUG=
R=magjed@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50449004

Cr-Commit-Position: refs/heads/master@{#8807}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 13:36:25 +00:00
pbos@webrtc.org
8904290aca Make screenshare target bitrate experiment always on 
BUG=4083
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44699004

Patch from sprang@webrtc.org <sprang@webrtc.org>.

Cr-Commit-Position: refs/heads/master@{#8806}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 12:50:34 +00:00
perkj@webrtc.org
9f9ea7e5ab Clean up webrtc external capture. 
This cl removes the dependency to the external capture module if external capturing is used in webrtc.
It also removes two external capture methods that is not needed.
Further more it adds I420VideoFrame::Create that takes a pointer to packed memory as input.

R=magjed@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43879004

Cr-Commit-Position: refs/heads/master@{#8804}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 10:55:39 +00:00
pbos@webrtc.org
443ad403f5 Remove FullStackTest frame pointer handles. 
Simplifies code, speculative fix for a DCHECK crash in ForemanCifPlr5.

BUG=4451
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45809005

Cr-Commit-Position: refs/heads/master@{#8803}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8803 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 07:34:38 +00:00
pbos@webrtc.org
6231fb6dac Prevent crashes when copying a zero-size frame. 
BUG=4451
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44749004

Cr-Commit-Position: refs/heads/master@{#8802}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 07:33:11 +00:00
bjornv@webrtc.org
6069032ebb Refactor audio_coding/isac: removed usage of macro WEBRTC_SPL_LSHIFT_W32 
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44659004

Cr-Commit-Position: refs/heads/master@{#8801}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 07:03:41 +00:00
bjornv@webrtc.org
4ab23d0e8f Refactor audio_coding/ilbc: removes usage of macro WEBRTC_SPL_LSHIFT_W32 
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.

Also removes unnecessary casts to int32_t from int16_t.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48519004

Cr-Commit-Position: refs/heads/master@{#8800}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8800 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 06:01:43 +00:00
andrew@webrtc.org
bd8c865f43 Remove build-time beamformer flags. 
RealFourier is now unconditionally enabled since we can fall back to the
Ooura FFT. We no longer need to condition users on rtc_use_openmax_dl.

R=aluebs@webrtc.org, mgraczyk@google.com

Review URL: https://webrtc-codereview.appspot.com/50439004

Cr-Commit-Position: refs/heads/master@{#8799}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 00:28:42 +00:00
andrew@webrtc.org
04c50981f8 Add the Ooura FFT to RealFourier. 
We are using the Ooura FFT in a few places:
- AGC
- Transient suppression
- Noise suppression

The optimized OpenMAX DL FFT is considerably faster, but currently does
not compile everywhere, notably on iOS. This change will allow us to use
Openmax when possible and otherwise fall back to Ooura.

(Unfortunately, noise suppression won't be able to take advantage of it
since it's not C++. Upgrade time?)

R=aluebs@webrtc.org, mgraczyk@chromium.org

Review URL: https://webrtc-codereview.appspot.com/45789004

Cr-Commit-Position: refs/heads/master@{#8798}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8798 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 20:07:43 +00:00
henrika@webrtc.org
80d9aeeda5 Adds full-duplex unit test to AudioDeviceTest on Android 
BUG=NONE
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42709004

Cr-Commit-Position: refs/heads/master@{#8795}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 15:28:42 +00:00
tommi@webrtc.org
361981faa8 Use scoped_ptr for ThreadWrapper::CreateThread. 
BUG=
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45799004

Cr-Commit-Position: refs/heads/master@{#8794}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8794 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 14:45:42 +00:00
tommi@webrtc.org
27c0be9dfe Remove ThreadObj #define and kThreadMaxNameLength from thread_wrapper. 
BUG=
R=hbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47679004

Cr-Commit-Position: refs/heads/master@{#8792}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 14:36:43 +00:00
magjed@webrtc.org
17c64d1c96 Revert "Put ViEFrameProviderBase::DeliverFrame back in the critical section in ViECapturer::DeliverI420Frame" 
This reverts commit r8724.

Reason for revert: This was not the cause of the tsan issues.

BUG=1128
R=mflodman@webrtc.org, pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50389004

Cr-Commit-Position: refs/heads/master@{#8790}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8790 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 10:58:17 +00:00
tommi@webrtc.org
c7157da599 Use atomic operations for setting/reading the trace filter. 
The filter is currently being set and read by a number of threads and tripping up tsan.

Original review: https://webrtc-codereview.appspot.com/47609004/

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47659004

Cr-Commit-Position: refs/heads/master@{#8789}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 09:30:45 +00:00
jmarusic@webrtc.org
9afaee74ab Reland 8749: AudioEncoder: return EncodedInfo from Encode() and EncodeInternal() 
Old review at:
https://webrtc-codereview.appspot.com/43839004/

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45769004

Cr-Commit-Position: refs/heads/master@{#8788}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 08:51:20 +00:00
pbos@webrtc.org
d21406d333 Remove command-line tool 'video_coding_test'. 
Removes a lot of code that prevents refactoring VideoCodingModule. Tests
covering the module should be TEST_Fs, and this looks like like fairly
unused code in general.

Adds a 'rtp_player' binary which performs a small subset.

BUG=4391
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44559004

Cr-Commit-Position: refs/heads/master@{#8787}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8787 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 08:19:44 +00:00
tommi@webrtc.org
c4709a2930 Split C++ class from macro overrides to fix Chromium build 
BUG=chromium:468375
TBR=kjellander@webrtc.org,ajm@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51409004

Cr-Commit-Position: refs/heads/master@{#8786}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8786 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 07:26:21 +00:00
braveyao@webrtc.org
5506a93efd Expose ViECaptureImpl::DisconnectCaptureDevice() to JNI of WebRTCDemo and call it before releasing camera to deregister the corresponding framecallback. Also stop camera after stop remote rendering as the correct termination order. 
BUG=4448
TEST=Manual Test
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46649004

Cr-Commit-Position: refs/heads/master@{#8785}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8785 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 00:12:40 +00:00
kjellander@webrtc.org
2a8a46dacb vp8: Add missing call to SetUsageMessage(). 
Without it vp8_coder --help does not work.

BUG=None
TEST=ninja -C out/Debug && out/Debug/vp8_coder --help now shows the
usage message.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44649005

Patch from Thiago Farina <tfarina@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#8783}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8783 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 21:09:16 +00:00
minyue@webrtc.org
8f76cd25ec Renaming neteq_opus_fec_quality_test. 
neteq_opus_fec_quality_test has been modified to test more configurations of Opus than only FEC. It makes sense to rename it to neteq_opus_quality_test. This was planned in

https://webrtc-codereview.appspot.com/45619004/

but was forgotten. This CL handles it, and makes it easy for review.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45709004

Cr-Commit-Position: refs/heads/master@{#8782}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 20:44:26 +00:00
pbos@webrtc.org
143451d259 Base start bitrate on last observed bitrate. 
Instead of setting bitrates based on codec target settings (which may
have previously been capped by a codec max bitrate), fetch the last
bandwidth allocated for this channel. This fixes broken low start bitrates
due to QCIF being set as default codec in WebRtcVideoEngine2 which caps
the max bitrate to 200kbps.

BUG=1788
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43789004

Cr-Commit-Position: refs/heads/master@{#8780}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 14:40:52 +00:00
pbos@webrtc.org
5a477a0bc6 DCHECK frame parameters instead of return codes. 
We should never be creating video frames without width/height. If these
DCHECKs fire we should be fixing the calling code instead.

BUG=4359
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46639004

Cr-Commit-Position: refs/heads/master@{#8779}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 14:12:38 +00:00
stefan@webrtc.org
4346d92578 Use SendTimeHistory to keep track of send times in simulations. 
Use SendTimeHistory to keep track of send times in simulations.
Keep piggybacking send time in PacketInfo for now but use history in
order to be more in line with what we expect to do.

Landing this for sprang@. Original CL: https://review.webrtc.org/43559004/

TBR=sprang@webrtc.org
BUG=4308

Review URL: https://webrtc-codereview.appspot.com/48569004

Cr-Commit-Position: refs/heads/master@{#8778}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 13:42:48 +00:00
henrik.lundin@webrtc.org
f18993323d Removing henrik.lundin from OWNERS in video_coding/* 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45699004

Cr-Commit-Position: refs/heads/master@{#8777}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 09:56:21 +00:00
perkj@webrtc.org
af612d5e07 Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."" 
Original cl description:
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.

With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame

This cl was previously reverted in https://webrtc-codereview.appspot.com/46549004/.

Patchset 1 contains the original patch after rebase.
Patshet 2 fix webrtc_perf_tests reported in chromium:465306

Note that chromium:465287 is being fixed in https://webrtc-codereview.appspot.com/43829004/

BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47629004

Cr-Commit-Position: refs/heads/master@{#8776}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8776 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 09:51:44 +00:00
henrik.lundin@webrtc.org
6dba1ebd14 Make AudioDecoder stateless 
The channels_ member varable is removed from the base class, and the
associated accessor function is changed to Channels() which is a pure
virtual function.

R=jmarusic@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43779004

Cr-Commit-Position: refs/heads/master@{#8775}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8775 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 09:48:12 +00:00
henrik.lundin@webrtc.org
fc562e0a56 Delete ACMGenericCodec::Encode and use AudioEncoder::Encode directly 
Move timestamp conversion out of ACMGenericCodec. Also remove lock
from ACMGenericCodec since the instance is always protected by
acm_crit_sect_ in AudioCodingModuleImpl.

Restructuring the code in AudioCodingModuleImpl::Encode to streamline
the use of locks.

R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46479004

Cr-Commit-Position: refs/heads/master@{#8773}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8773 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 07:32:41 +00:00
tommi@webrtc.org
019955d770 Revert 8749 "We changed Encode() and EncodeInternal() return typ..." 
The reason is that this cl adds a static initializer so we can't roll webrtc into Chromium.
See audio_encoder.cc and 'sizes' regression here:
http://build.chromium.org/p/chromium/builders/Linux%20x64/builds/186

> We changed Encode() and EncodeInternal() return type from bool to void in this issue:
> https://webrtc-codereview.appspot.com/38279004/
> Now we don't have to pass EncodedInfo as output parameter, but can return it instead. This also adds the benefit of making clear that EncodeInternal() needs to fill in this info.
> 
> R=kwiberg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/43839004

TBR=jmarusic@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49449004

Cr-Commit-Position: refs/heads/master@{#8772}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8772 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 06:38:40 +00:00
tommi@webrtc.org
edd517bca1 Fix FYI build - add a missing include to event_tracer.h in system_wrappers. 
TBR=magjed@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/48559004

Cr-Commit-Position: refs/heads/master@{#8768}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 22:15:28 +00:00
guoweis@webrtc.org
54d072ea20 Add CVO support to video_coding layer. 
CVO is, instead of rotating frame on the capture side, to have renderer rotate the frame based on a new rtp header extension.

The change includes
1. encoder side needs to pass this from raw frame to the encoded frame.
2. decoder needs to copy it from rtp packet (only the last packet of a frame has this info) to decoded frame.

R=mflodman@webrtc.org
TBR=stefan@webrtc.org

BUG=4145

Review URL: https://webrtc-codereview.appspot.com/46429006

Cr-Commit-Position: refs/heads/master@{#8767}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 21:55:37 +00:00
pthatcher@webrtc.org
63a10978e1 Remove troublesome Windows line ending. 
R=decurtis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48549004

Cr-Commit-Position: refs/heads/master@{#8766}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 21:50:29 +00:00
tommi@webrtc.org
462dbcfc2a Fix bug in Transport where channel_.clear() was being called without a lock. 
Looks like this snuck in between misaligned braces.

Also switching to C++11 for loops, reducing lock scopes in a few places and removing locks in others.

BUG=4444
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43769004

Cr-Commit-Position: refs/heads/master@{#8765}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8765 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 21:40:26 +00:00
tkchin@webrtc.org
b493cb4497 Add storage alignment fix for opengles2.0 for iOS 
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40179004

Patch from Iurii Shevchuk <youwrk@gmail.com>.

Cr-Commit-Position: refs/heads/master@{#8764}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8764 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 20:18:42 +00:00
tkchin@webrtc.org
da4fcc494c Add minor fixes to video_capture_ios.mm in order to make it more robust. 
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46429005

Patch from Iurii Shevchuk <youwrk@gmail.com>.

Cr-Commit-Position: refs/heads/master@{#8763}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8763 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 20:13:49 +00:00
sprang@webrtc.org
779c3d16b9 Use ByteReader/ByteWriter instead of rtputility and manual shift/add. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41289004

Cr-Commit-Position: refs/heads/master@{#8761}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8761 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 16:44:54 +00:00
sprang@webrtc.org
09098dabd3 Fix screenshare loopback target bitrate which isn't correctly configured 
BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48539004

Cr-Commit-Position: refs/heads/master@{#8760}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8760 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 16:28:11 +00:00
tommi@webrtc.org
25819b8294 Revert 8753 "Use atomic operations for setting/reading the trace..." 
Caused VP9 test to fail on TSAN and doesn't build in some configuration due to
"../webrtc/base/criticalsection.h:181:12: error: cannot compile this atomic library call yet"
:-(

> Use atomic operations for setting/reading the trace filter.
> The filter is currently being set and read by a number of threads and tripping up tsan.
> 
> R=mflodman@webrtc.org
> BUG=
> 
> Review URL: https://webrtc-codereview.appspot.com/47609004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51369004

Cr-Commit-Position: refs/heads/master@{#8759}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8759 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 15:35:41 +00:00