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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
75dd2885c5
Add an interface for accepting keypress signals to AudioProcessing.
...
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5529 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 20:52:30 +00:00
aa1278de46
Rename merged webrtc lib to libwebrtc_merged.a.
...
The name "libwebrtc.a" was conflicting with the newish webrtc target,
resulting in this error:
$ ./webrtc/build/gyp_webrtc merged_lib.gyp
$ ninja -C out/Debug
ninja: warning: multiple rules generate libwebrtc.a. builds involving
this target will not be correct; continuing anyway
ninja: error: dependency cycle: no_op -> libwebrtc.a -> no_op
BUG=b/12955740
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8409005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5528 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 18:22:29 +00:00
8685af7ea0
Remove "Too long processing time of Incoming frame" logspam.
...
This isn't indicative of anything actionable and spams android logcat with times
in the 10-30ms range several times per second.
BUG=2732
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5527 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 17:48:11 +00:00
a80be4b23c
Add boundary checking to supress gcc 4.8.3 warning.
...
BUG=2888
Test=try, voe_cmd_test
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5526 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 16:38:45 +00:00
fc320466d1
Remove ViE external encryption API.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8079005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5525 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 15:27:49 +00:00
82ebb463fd
Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file.
...
This patch removes generate_asm_header.gypi and uses libvpx's obj_int_extract and
unpack_lib_posix to generate offset header files.
It make the simliar feature's implementation consistent.
R=andrew@webrtc.org , fischman@webrtc.org , fischman@chromium.org
BUG=334447
Committed: https://code.google.com/p/webrtc/source/detail?r=5517
Review URL: https://webrtc-codereview.appspot.com/7769006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5524 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 04:48:27 +00:00
16c08f03da
Restore mixing integration tests.
...
These high level tests were disabled over time. Since they depend on
real-time results and the filesystem, they tended to be flaky on the
bots. We now give it a very generous 1 second to start up all channels
before verification and a further relaxed file length check. If we
continue to see problems, I will up the startup delay.
The restored tests would have caught the AGC bug fixed here:
https://code.google.com/p/webrtc/source/detail?r=5454
Add a new "real audio" stress test to exercise more code paths. This
would have caught the refactor bug fixed here:
https://code.google.com/p/webrtc/source/detail?r=5437
BUG=2164,2844
TESTED=git try. Verified it would have caught the aforementioned bugs
by reintroducing them.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5522 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 23:04:39 +00:00
a65abf9d3a
Revert "Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file."
...
This reverts commit 7686f0ddda717a9e776be0e219f039f68a10f9ed.
BUG=
TBR=andrew@webrtc.org , fischman@webrtc.org ,
Review URL: https://webrtc-codereview.appspot.com/8369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5520 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 19:26:26 +00:00
1f64f06784
Add stats of incoming frame delays for debugging bandwidth estimation.
...
BUG=crbug/338380
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5519 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 19:12:14 +00:00
