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2714 Commits

Author SHA1 Message Date
jiayl@webrtc.org
f0fc72f70e Call PrintWindow for the first time of capturing to capture the window frames correctly. 
This will fix artifacts on the captured window frames, especially for cmd, which
sometimes leaks glimpss of other window's content.

BUG=
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/8989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5616 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-27 16:43:12 +00:00
andrew@webrtc.org
00073aafa8 Clean up CPU detection defines in SincResampler a little. 
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5615 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-27 04:12:34 +00:00
jiayl@webrtc.org
0231e801d6 Invalidate the whole screen when the frame size is changed. 
Otherwise we'll compare frames of different sizes and read into invalid
memory.

BUG=https://code.google.com/p/chromium/issues/detail?id=345498
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/9149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5614 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-26 18:54:57 +00:00
andrew@webrtc.org
2038920a2b Use scoped_ptr<T[]> in SincResampler to avoid .get()[] weirdness. 
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5613 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-26 18:14:54 +00:00
henrik.lundin@webrtc.org
c0e9aebe8f Add SetConfig method to FakeNetworkPipe and to DirectTransport 
This method allow the user to change the network configuration
during run-time. This is useful when testing how components react
to changing bandwidth.

BUG=2636
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5612 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-26 13:34:52 +00:00
aluebs@webrtc.org
bc1d22461b Add experimental noise suppression flag to audioproc test 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5608 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-25 16:50:22 +00:00
sprang@webrtc.org
050892a95b Missing include in experiments.h 
webrtc/typedefs.h should be included in webrtc/experiments.h since the
type uint32_t is being used and it is not indirectly included from this
file.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5607 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-25 09:17:43 +00:00
wu@webrtc.org
7f52a6ef2b Split the implementation of VP8Encoder|Decoder::Create into a seperated file 
(vp8_factory.cc).

R=fischman@webrtc.org, marpan@google.com, marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5606 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 23:56:39 +00:00
asapersson@webrtc.org
23caa2d8d6 Fix to get total number of sent and received rtcp packets. 
BUG=2638
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8979005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5601 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 09:27:38 +00:00
braveyao@webrtc.org
4f0801bd39 AviRecorder is missing a critical section. 
BUG=2885
TEST=AUTOTEST
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 09:19:36 +00:00
kjellander@webrtc.org
55fcd716f3 Disable libjingle_peerconnection_java_unittest 
Broken by libjingle roll in r5590.

TBR=henrike@webrtc.org
BUG=2960
TEST=git try --bot=linux_baremetal --revision=5597

Review URL: https://webrtc-codereview.appspot.com/9029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5598 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-23 18:47:27 +00:00
bjornv@webrtc.org
33af96c5c2 Removed unused mock methods in audio_processing 
TESTED=trybots,modules_unittests
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8999005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5597 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-21 23:56:05 +00:00
asapersson@webrtc.org
0f2809a5ac Add RTCP packet class. 
Adds packet types: sr, rr, bye, fir.

BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5592 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-21 08:14:45 +00:00
andrew@webrtc.org
c0907eff42 MIPS optimizations for AEC audio processing module 
The resulting output streams obtained by testing with audioproc test application
are bit-exact with generic C code output streams.

Performance gain achieved:
- mips32 ~ 17%
- mips32r2 ~ 20%
- mipsdsp & mipsdspr2 ~ 21%

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7359004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5591 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-21 00:13:31 +00:00
elham@webrtc.org
3f170dd309 Updated WebRTC version to 3.50 
TBR= wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5589 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 22:31:07 +00:00
andrew@webrtc.org
d617a44a4f Add an AlignedFreeDeleter and remove scoped_ptr_malloc. 
- Transition scoped_ptr_mallocs to scoped_ptr.
- AlignedFreeDeleter matches Chromium's version.

TESTED=try bots
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8969005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5587 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 21:08:36 +00:00
turaj@webrtc.org
d4d5be8781 Minor improvement in RoundToInt16 implementation. 
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5586 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 20:55:21 +00:00
asapersson@webrtc.org
a0a6df3910 Modified overuse detection thresholds. 
BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8949005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5585 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 17:37:37 +00:00
henrik.lundin@webrtc.org
04a691adac Removing a variable that was never read 
In NetEq4, the local variable discard_count in
PacketBuffer::DiscardOldPackets() was incremented but never read.
Removing it.

TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5584 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 15:27:00 +00:00
fbarchard@google.com
66061992fb ifdef the alsa code based on macro USE_X11 
BUG=none
TEST=try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 03:05:05 +00:00
turaj@webrtc.org
78f0db4710 Fix the break caused by r5579. 
TBR=tlegrand@google.com
BUG=

Review URL: https://webrtc-codereview.appspot.com/8939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5581 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 23:07:31 +00:00
turaj@webrtc.org
c2d69d3229 Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable. 
BUG=2944
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5579 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 20:31:17 +00:00
jiayl@webrtc.org
97e7a640d8 Make WindowCapturerLinux handling window resize events. 
We need to re-initialize the XServerPixelBuffer to the new size
when a window resize event is received.

BUG=https://code.google.com/p/chromium/issues/detail?id=339953
R=sergeyu@chromium.org, wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/8679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5578 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 17:28:41 +00:00
andresp@webrtc.org
242102517d Added architecture for compiling under chrome NaCl. 
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 13:55:02 +00:00
tina.legrand@webrtc.org
056287eee0 This CL separate all ACM tests with new and old implementation of ACM and NetEq. The reason is to debug an issue with failure on Android try bots. We need to see if the error only occurs with the new ACM/NetEq, or if it is a flakiness that affects both. 
BUG=issue2874
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5576 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 13:45:54 +00:00
asapersson@webrtc.org
8098e07478 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). 
Add counter to RTCP sender and RTCP receiver.
Add video api GetRtcpPacketTypes().

BUG=2638
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 11:59:02 +00:00
henrika@webrtc.org
b7a91fa95a Removes VoERTP_RTCP::InsertExtraRTPPacket. 
Reasons for removing:

- Feels like a complete hack IMHO.
- Not used by any client.
- Unclear functionality regarding time stamp, marker bit etc.
- Causes several issues in tests due to a bad design which mainly depends on the fact that this API "breaks" an ongoing data/packet flow and it complicates the threading model and creates risks for deadlock and memory corruption. Not worth trying to fix given the very unclear benefit of maintaining the API. Better to remove the API instead.
- We also see lots of TSan races and memcheck errors related to this API.

BUG=2296,2240
R=mflodman@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5574 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 08:58:08 +00:00
sergeyu@chromium.org
e384104166 Fix DesktopAndCursorComposer not to crash 
DesktopAndCursorComposer was crashing when screen/window
capturer returns a NULL frame due to an error.

BUG=crbug.com/344093
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5573 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 23:26:34 +00:00
andrew@webrtc.org
27c6980239 Move the volume quantization workaround from VoE to AGC. 
Voice engine shouldn't really have to manage this. Instead, have AGC
keep track of the last input volume, so that it can avoid getting stuck
under coarsely quantized conditions.

Add a test to verify the behavior.

TESTED=unittests, and observed that AGC didn't get stuck on a MacBook
where this problem can actually occur.

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5571 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 20:24:56 +00:00
solenberg@webrtc.org
00844d7bef Remove obsolete voe_unit_test. 
BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5570 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 18:50:50 +00:00
mflodman@webrtc.org
c320027d6a Don't print a warning if RTPPacketHistory::SetStorePacketStatus is called 
twice with the same settings.

Without this change, setting up a call with the new video API will
print a trace warning.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5566 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 14:51:00 +00:00
turaj@webrtc.org
2086e0fbf3 Remove unnecessary warnings. 
BUG=
TEST=try job
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8719005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5565 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 14:22:20 +00:00
solenberg@webrtc.org
a07923339b Remove external encryption API for VoE. 
BUG=
R=henrika@webrtc.org, henrikg@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 11:27:22 +00:00
sprang@webrtc.org
346094cb01 Incorrect overhead calculation when using FEC + RTP extension headers. 
When frames are fragmented inte multiple RTP packets in order to not
exceed a maximum packet size, the header overhead calculation must
take into account that FEC redundancy packets may use more than the
12 bytes of the basic RTP header. For example, a csrc list or extension
headers may be present.

BUG=2899
R=phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8769005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5562 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 08:40:33 +00:00
asapersson@webrtc.org
b60346e951 Reset estimate if no frame has been seen for a certain time (to avoid large jitter if stop sending). 
Add delay before start processing after a reset.

BUG=1577
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8699006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5561 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-17 19:02:15 +00:00
henrik.lundin@webrtc.org
340746aa13 Misc small nits in NetEq 
Fixing a few small things found recently. This is mostly cosmetics.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8749005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5558 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-17 11:37:16 +00:00
andrew@webrtc.org
f92aaff104 AudioProcessing is not a Module. 
Remove Module as the base class of AudioProcessing. The inherited
methods were all no-ops.

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/8779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5556 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-15 04:22:49 +00:00
bjornv@webrtc.org
e2fc13e42f Refactoring common_audio/signal_processing: Removed two macros used by isac only. 
Removed a macro for malloc() and one for free().  They are only used by the audio codec isac, where I replaced the macro with its implementation.
Further, the includes were updated with full paths and put in alphabetical order.

BUG=N/A
TESTED=trybots,module_tests,module_unittests
R=turaj@webrtc.org, turajs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5554 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-14 23:12:34 +00:00
stefan@webrtc.org
9075d519a2 Adding a critical section missing in r5543. 
This fixes a race caught by the linux tsan bot.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5551 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-14 09:45:58 +00:00
andrew@webrtc.org
38bf249049 Initialize output_will_be_muted_. 
TBR=aluebs

Review URL: https://webrtc-codereview.appspot.com/8659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5546 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-13 17:43:44 +00:00
asapersson@webrtc.org
8f690bc222 Increase overuse and normal use thresholds for Mac. 
BUG=1577
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5544 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-13 14:43:18 +00:00
stefan@webrtc.org
ae2563ae2f Fixes a race when writing to send_padding_. 
TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5543 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-13 13:48:38 +00:00
henrik.lundin@webrtc.org
fcfc6a990e Small refactoring of NetEq unittest for CNG with clock drift 
Converting the test to a method within the test fixture, and setting
up two tests that call this method. One for positive and one for
negative clock drift. The one with positive clock drift is disabled
for now since it does not pass, but will be re-enabled shortly.

This change is only made for NetEq4.

R=tlegrand@google.com

Review URL: https://webrtc-codereview.appspot.com/8599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5541 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-13 11:42:28 +00:00
andrew@webrtc.org
17342e5092 Add a method to inform AudioProcessing that its output will be muted. 
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5538 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-12 22:28:31 +00:00
jiayl@webrtc.org
de782180b0 Change the type of propagation delta from int64 to int. 
The delta value never exceeds the range of int. Changing it to int will save memory and copying cost.

BUG=2910
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5537 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-12 19:19:23 +00:00
andrew@webrtc.org
07b5950c12 Initialize key_pressed_. 
Was resulting in an error on Mac Asan:
[ RUN      ] ApmTest.DebugDump
[libprotobuf FATAL ../../third_party/protobuf/src/google/protobuf/message_lite.cc:224] CHECK failed: !coded_out.HadError():

TBR=aluebs

Review URL: https://webrtc-codereview.appspot.com/8539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5536 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-12 16:41:13 +00:00
andrew@webrtc.org
ce8e077cf0 Add a keypress field to the audioproc debug proto. 
Log the value in AudioProcessing, and unpack it to a new file in the
unpacking tool.

TESTED=
- The new tool can unpack old dumps.
- The old tool can unpack new dumps (without keypress.bool).
- Unpacking a new dump from voe_cmd_test produces a keypress.bool that
appears correct when examined.

R=aluebs@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5535 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-12 15:28:30 +00:00
pbos@webrtc.org
8118f1861f Set pacing bitrates in SetEncoder. 
Before the change no padding was allowed before the first remote bitrate
estimation was received. This bitrate estimation is based on what's
actually sent. In tests I set codec.startBitrate to 300 instead of
325, which incidentally means that only the first layer gets encoded.
As we only send ~150kbps instead of 300, the first REMB will
significantly pull down the remote bitrate estimate instead of keeping
the existing rate, even though there's no problem with the link.

This was detected in RampUpTest.PacingWithRtx,
(send_config.codec.startBitrate=300), which caused the tests to become a
lot slower, and flake out more. By allowing padding initially we're able
to keep our initial bitrate estimate.

R=stefan@webrtc.org
TEST=Running RampUpTest.WithPacingAndRtx with startBandwidth=300.
BUG=

Review URL: https://webrtc-codereview.appspot.com/8529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5534 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-12 14:50:29 +00:00
solenberg@webrtc.org
67e70442b5 Remove unused and not working voe_extended_test. 
BUG=2913
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5533 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-12 09:58:49 +00:00
andrew@webrtc.org
b659e2844d Reduce mixing threshold in test to avoid flakiness. 
Flake observed here:
http://chromegw/i/client.webrtc/builders/Win32%20Release%20%5Blarge%20tests%5D/builds/953/steps/voe_auto_test/logs/stdio

TBR=andresp

Review URL: https://webrtc-codereview.appspot.com/8489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5530 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-11 21:52:50 +00:00