pbos@webrtc.org
2ffb149c2c
Replace VideoFrameI420 with I420VideoFrame.
...
Gives one less struct/class for I420 video frames.
BUG=2657
R=mflodman@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5160 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-22 13:10:13 +00:00
andrew@webrtc.org
b0ed8f8a08
Don't reset the AEC filter in extended mode.
...
I don't believe I've witnessed this "feature" ever provide a benefit,
and have now collected some evidence of its harm when using the
extended filter mode. It can cause erroneous resets in two cases:
1. Some preprocessing noise suppression is enabled in the system (i.e.
"audio enhancements") that push the noise floor very low, possibly to
zero. If the filter is non-zero this condition can be triggered very
easily, and erroneously.
2. Non-zero energy in the filter before the peak impulse response can
cause a slight (and harmless) "pre-echo" in the error signal. This
becomes more significant as the peak is set further back in the filter.
This effect can cause needless resets during echo onsets.
In short, this isn't a great criterion for filter reset and has the
potential to cause serious harm. Ideally we would remove it entirely,
but in the interests of safety, can start with the extended mode.
BUG=1261
R=aluebs@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5159 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-22 06:39:42 +00:00
dwkang@webrtc.org
9e85c01ec8
Add exception handling when configuring MediaCodc in order to prevent break in the new sdk release.
...
BUG=2603
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5158 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-22 02:49:17 +00:00
henrik.lundin@webrtc.org
9fe3603dc1
Renaming ViEEncoderObserver::VideoSuspended
...
New name is ViEEncoderObserver::SuspendChange.
BUG=2436
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5157 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 23:00:40 +00:00
pbos@webrtc.org
484ee962b5
Protect reads of ViEEncoder::video_suspended_.
...
Does not fix an immediate bug, since this is the only method writing to
it there are no concurrent writes, but this should be more future-proof
by protecting all accesses.
BUG=2606
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4109006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5156 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 18:44:23 +00:00
fischman@webrtc.org
1977960866
AppRTCDemo(ios): remove codesigning hack now that gyp signs by default.
...
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4119005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5155 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 16:48:51 +00:00
stefan@webrtc.org
ef2d55461b
Increase size of pacer window to 500 ms as that better matches the encoder.
...
BUG=1812
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4129006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5154 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 14:37:11 +00:00
henrik.lundin@webrtc.org
331d4402fc
Connect pacer/padding to SuspendBelowMinBitrate
...
The suspend function must not be engaged unless padding is also enabled.
This CL makes the connection so that the pacer and padding is enabled
when SuspendBelowMinBitrate is.
Had to change the unit test to make it aware of the padding packets.
BUG=2606
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5153 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 14:05:40 +00:00
pbos@webrtc.org
ffe1b17b57
Lock access to ModuleRtpRtcpImpl::simulcast_.
...
Fixes race between RegisterSendPayload and SendOutgoingData.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4099006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5152 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 13:53:13 +00:00
pbos@webrtc.org
2c46f8d854
Rename DestroyStream methods to include Video.
...
Matches r5135 which renames CreateSendStream->CreateVideoSendStream for
instance.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4109005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5151 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 13:49:43 +00:00
henrik.lundin@webrtc.org
6f6ba6edee
Fix issues with sequence number wrap-around in jitter statistics
...
Wrap-arounds in sequence numbers (and in timestamps) were not always
treated correctly. This is fixed by introducing two helper functions
for correct comparisons, and by casting to the right word size.
Also added a new member variable to AutomodeInst_t. The new member keeps
track of when the first packet has been registered in the automode code.
This was previously done implicitly (and not very good) using the fact
that the lastSeqNo and lastTimestamp members were initialized to zero.
Two new unit tests were added as part of this CL.
NetEqDecodingTest.SequenceNumberWrap was failing before the fixes were
made; now it is ok.
BUG=2654
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5150 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 13:17:29 +00:00
pbos@webrtc.org
b3cc78de28
Add -Wnon-virtual-dtor warning for C++ code.
...
BUG=2659
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4119006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5149 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 11:42:02 +00:00
sprang@webrtc.org
72964bd4fb
Make interface destructor virtual
...
In summary, do this:
- ~FrameCountObserver() {}
+ virtual ~FrameCountObserver() {}
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4099005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5148 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 09:09:54 +00:00
asapersson@webrtc.org
8d02f5dc71
Added API for enabling/disabling RTCP Receiver Reference Time extension.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3419005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5147 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 08:57:04 +00:00
asapersson@webrtc.org
54a05518e2
Increase run-time for full stack test for the rtt to be added reliably to the delay measurement.
...
BUG=2592
R=holmer@google.com , phoglund@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5146 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 07:45:08 +00:00
braveyao@webrtc.org
425e1d0fb9
Typo in vie_autotest_win.cc
...
BUG=2637
TEST=AutoTest
R=mflodman@webrtc.org , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5145 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 02:17:01 +00:00
henrike@webrtc.org
a750044396
Fixes a crash in VoE when unregistering JNI hooks.
...
BUG=11695087
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5144 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 22:32:12 +00:00
wu@webrtc.org
364f204d16
Update talk to 56698267.
...
TBR=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/4119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5143 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 21:49:41 +00:00
sprang@webrtc.org
dc50aaeaa8
Interface changes to old api, for use by new api transition.
...
BUG=2589
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5142 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 16:47:07 +00:00
asapersson@webrtc.org
b24d33565c
Added ViE API for getting overuse measure.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3129005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5141 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 13:51:40 +00:00
pbos@webrtc.org
d29d4e9c08
Deliver I420VideoFrames from VideoRender module.
...
Performance issue and simplicity, this implementation skips conversion
to VideoEngine's frame format and then back again to I420VideoFrame.
BUG=2526
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5140 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 13:19:54 +00:00
asapersson@webrtc.org
1ae1d0c471
Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module).
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2383004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5139 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 12:46:11 +00:00
pbos@webrtc.org
27326b6a42
Rename newapi::Transport::SendRTP()->SendRtp().
...
Also fit rampup_tests.cc to use internal::TransportAdapter instead of
implementing its own.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5138 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 12:17:04 +00:00
pbos@webrtc.org
ce90eff345
Rename RTP-extension constants.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5137 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 11:48:56 +00:00
pbos@webrtc.org
53c8573525
Rename video streams' start/stop methods.
...
{Start,Stop}{Send,Receive}() -> {Start,Stop}{Sending,Receiving}().
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3609005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5136 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 11:36:47 +00:00
pbos@webrtc.org
5a63655ab0
Rename Call::Create{Receive,Send}Stream().
...
Renaming the methods to include Video. Long-term there will hopefully be
AudioSendStream/AudioReceiveStreams as well.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5135 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 10:40:25 +00:00
aluebs@webrtc.org
0b72f5863b
Add experimental noise suppression dummy API.
...
Add this flag to the voe_cmd_test.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5134 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-19 15:17:51 +00:00
sergeyu@chromium.org
5d85819dd2
Fix DesktopAndCursorComposer to restore frames to the original state.
...
Screen capturers may reuse frame buffers and they expect that the
frame content isn't changed by the frame consumer.
DesktopAndCursorComposer draws mouse cursor on generated frames and
it was releasing the frames with the mouse cursor on them. Fixed
it to restore frame content erasing mouse cursor before returning
desktop frames.
BUG=crbug.com/316297
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/3899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5133 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-19 02:15:47 +00:00
turaj@webrtc.org
7a05ae5c69
Adding back main() to the test. Now it is possible to choose between ACM1 and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded.
...
The main() was deleted in r4731.
BUG=
R=andrew@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2370004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5132 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 18:16:53 +00:00
pbos@webrtc.org
9c5fb76662
Exclude AV-sync test from Valgrind platforms.
...
Test is performance-dependent and was observed to never sync on the
linux_memcheck bot.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5131 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 16:22:50 +00:00
henrik.lundin@webrtc.org
ce8e0936d9
Rename AutoMute to SuspendBelowMinBitrate
...
Changes all instances throughout the WebRTC stack.
BUG=2436
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 12:18:43 +00:00
stefan@webrtc.org
28bf50f0ec
Fix test broken with r5128.
...
TBR=pbos@webrtc.org
BUG=2530
Review URL: https://webrtc-codereview.appspot.com/3979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5129 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 11:58:24 +00:00
stefan@webrtc.org
b082ade3db
Hook up audio/video sync to Call.
...
Adds an end-to-end audio/video sync test.
BUG=2530, 2608
TEST=trybots
R=henrika@webrtc.org , mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 11:45:11 +00:00
stefan@webrtc.org
4cfa6050f6
Fix breakage after introducing new test.
...
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3899005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5127 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15 13:15:56 +00:00
stefan@webrtc.org
69969e2e2f
Improve Call tests for RTX.
...
Also does some refactoring to reuse RtpRtcpObserver.
BUG=1811
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5126 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15 12:32:15 +00:00
henrik.lundin@webrtc.org
6e95d7afab
Increment RTP timestamps for padding packets
...
This CL makes the padding packets get their own RTP timestamps,
rather than having the same timestamp as the last sent video
packet. The purpose is to solve Issue 2611, where the overuse-
detector does not react to padding packets.
A test was implemented to verify that the padding packets do
get their own timestamps.
BUG=2611
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5125 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15 08:59:19 +00:00
pbos@webrtc.org
6488761f2e
Implement VideoSendStream::SetCodec().
...
Removing assertion that SSRC count should be the same as the number of
streams in the codec. It makes sense that you don't always use the same
number of streams under one call. Dropping resolution due to CPU overuse
for instance can require less streams, but the SSRCs should stay
allocated so that operations can resume when not overusing any more.
This change also means we can get rid of the ugly SendStreamState whose
content wasn't defined. Instead we use SetCodec to change resolution
etc. on the fly. Should something else have to be replaced on the fly
then that functionality simply has to be implemented.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3499005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5123 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-14 08:58:14 +00:00
sergeyu@chromium.org
183c727bca
Disable datachannel_unittest.cc
...
the test fails to compile because it uses incorrect gmock path (as
some other tests).
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5121 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 22:59:20 +00:00
sergeyu@chromium.org
a23f0ca4ba
Update talk to 56619788
...
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3839005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5120 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 22:48:52 +00:00
kjellander@webrtc.org
e8722856f9
Disable all vie_auto_tests on Linux for now (take 2)
...
Turns out OS_LINUX is not working in this context
(see http://review.webrtc.org/3539005/ )
WEBRTC_LINUX is the right define to use.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5119 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 15:51:49 +00:00
kjellander@webrtc.org
c8489852ec
Disable all automated vie_auto_tests on Linux for now
...
Since the switch from icewm to openbox window manager on
Linux in Chrome infra, causes the test to hang when
creating Windows.
TEST=trybots compile step
BUG=chromium:318760
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3539005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5118 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 15:44:54 +00:00
stefan@webrtc.org
9b82f5a6ed
Fix for RTX in combination with pacing.
...
Retransmissions didn't get sent over RTX when pacing was enabled since
the pacer didn't keep track of whether a packet was a retransmit or not.
BUG=1811
TEST=trybots
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5117 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 15:29:21 +00:00
turaj@webrtc.org
03f33709f8
Inject config when creating channels to override the existing one.
...
BUG=
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5116 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 00:02:48 +00:00
henrik.lundin@webrtc.org
e8433eb115
Reimplementing NetEq4's AudioVector
...
The current implementation using std::vector is too slow.
This CL introduces a new implementation, using a regular
array as data container.
In AudioMultiVector::ReadInterleavedFromIndex, a special case for
1 channel was implemented, to further reduce runtime. Finally,
AudioMultiVector::Channels was reimplemented.
The changes in this CL reduces the runtime of neteq4_speed_test
by 33%.
BUG=1363
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5115 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-12 13:15:02 +00:00
asapersson@webrtc.org
38599510df
Parse next RTCP XR report block after an unsupported block type.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5114 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-12 08:08:26 +00:00
minyue@webrtc.org
3e427263ee
Reducing opus_test runtime to pass Android test
...
BUG=2609
R=solenberg@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5111 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-11 22:03:52 +00:00
andrew@webrtc.org
e03cafaebc
MIPS optimizations for AECM audio processing module
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R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2279005
Patch from Ljubomir Papuga <lpapuga@mips.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5110 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-11 20:10:01 +00:00
andrew@webrtc.org
b0730108a2
Move audio_processing dependencies to a variable.
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R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5108 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-11 17:20:27 +00:00
pbos@webrtc.org
57eb858698
Remove ".." from include_dirs in build/common.
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BUG=1662
TEST=compile on trybots
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2332004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5107 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-11 10:20:27 +00:00
andrew@webrtc.org
6e908b3adf
Remove unnecessary include_dirs from audio_processing.
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TBR=aluebs
TESTED=trybots
Review URL: https://webrtc-codereview.appspot.com/3659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5106 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08 19:52:05 +00:00