5272eb8d83
Don't register iSAC-swb and iSAC-fb in NetEqDecodingTest.
...
Android bots break due to r5164. This CL patches that issue.
BUG=
TEST=modules_unittests on local device.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5166 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-23 00:11:32 +00:00
e839da02c1
Fix MouseCursor to MouseCursorShape conversion in ScreenCapturerWin.
...
BUG=crbug.com/322596
R=dcaiafa@chromium.org , wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/4279005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5165 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-22 20:39:16 +00:00
78b41a09bd
Fix issues with sequence number wrap-around in jitter statistics.
...
Related CL for NetEq 3 is https://code.google.com/p/webrtc/source/detail?r=5150
Jitter statistics was not very sensitive to timestamp warp-around, and NetEqDecodingTest.TimestampWrap *DID NOT* fail before fixes applied. However, we still keep the test.
The criteria for the tests are not satisfied for first few packets, before any wrap-around happens. We could either relax the bound or ignore the first few packets. We chose the latter.
BUG=2662
TEST=modules_unittests,trybots
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5164 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-22 20:27:07 +00:00
1e8c93c953
Distinguish instances of ACM1 from ACM2 by a version string. This is fpr testing purposes and will be removed when the experiment is done and ACM1 is fade out.
...
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4069006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5161 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-22 17:04:49 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
e03cafaebc
MIPS optimizations for AECM audio processing module
...
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
b0730108a2
Move audio_processing dependencies to a variable.
...
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
57eb858698
Remove ".." from include_dirs in build/common.
...
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
6e908b3adf
Remove unnecessary include_dirs from audio_processing.
...
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
48df38114d
Fix for making sure that the packet in order checks are done prior to updating the last received packet state.
...
Without this fix all packets are considered out-of-order by the rtp receiver, causing the last received state
in the rtp receiver to never get valid.
Also makes sure that only valid timestamps and receive times are used for audio/video sync.
BUG=2608
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5102 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08 15:18:52 +00:00
bde3056567
Fix for video_processor_intergration_tests to run in parallel.
...
BUG=2601.
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5091 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-06 20:59:29 +00:00
7a36cb408b
Add missing dependencies to .isolate files
...
Also fix invalid paths in video_engine_tests.isolate.
TEST=trybots passing compile step (no .isolate use is deployed on them yet)
BUG=chromium:300017
R=pbos@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3399005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5084 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-05 14:28:57 +00:00
b8cb85b348
Fix broken build on x86 Android
...
BUG=2545
R=fischman@webrtc.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3019004
Patch from Lu Quiang <qiang.lu@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5081 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 19:06:08 +00:00
766154aa1d
Removed unused code.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5073 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 08:35:50 +00:00
5dd2ecb32d
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
...
This reverts commit f4ca3808bd9ec2293ec205f2f4a7d9739ce1f2df.
TBR=niklas.emblom@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/3269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5071 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 23:41:04 +00:00
74e6e8458e
Remove extra copy in VideoCaptureImpl::IncomingFrameI420
...
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an
existing memory buffer without taking ownership. Use this to remove an extra
copy in VideoCaptureImpl::IncomingFrameI420.
BUG=1128
BUG=chromium:310271
TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux
TBR=fischman@webrtc.org , mflodman@webrtc.org , mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3239005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5070 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 21:48:16 +00:00
d705649edf
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
...
This reverts commit 99f9743fe39066ba93b41f2b0a417696cbbd06fb.
Revert while build breakage is fixed.
BUG=None
TBR=niklas.emblom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5069 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 21:20:15 +00:00
1a4ed0d70c
Remove extra copy in VideoCaptureImpl::IncomingFrameI420
...
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an
existing memory buffer without taking ownership. Use this to remove an extra
copy in VideoCaptureImpl::IncomingFrameI420.
BUG=1128
TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux
R=fischman@webrtc.org , mflodman@webrtc.org , mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5068 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 20:32:28 +00:00
58cd31665c
Address Clag Analyzer issues.
...
Following are the issues related to NetEq 4, discovered by Clang Analyzer.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b44b95.html#EndPath
Valid; perhaps unlikely, addressed.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-6beef6.html#EndPath
Valid, addressed.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2e3883.html#EndPath
Valid; Addressed
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-293659.html#EndPath
Valid; Addressed.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b875cd.html#EndPath
Valid; Addressed.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/index.html
Not valid;
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-86f2ed.html#EndPath
Not Valid; the assert statement will be short-circuited, however I also added a check of nullity of |packet|.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-3a5669.html#EndPath
Not Valid: |energy_input| and |energy_expand| are both non-negative, therefore if-statement condition on line 226 is not satisfied unless |energy_input| >= 1. Therefore |energy_input| cannot be zero after normalization to 14-bits, i.e. operations on lines 228 & 229.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2f914f.html#EndPath
Valid; addressed.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2332b1.html#EndPath
Valid; addressed.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-de8dea.html#EndPath
Not valid; |out_len| is set when Process() is called, however, it makes sense to initialize to zero when declaring |out_len|.
https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b671a3.html#EndPath
Valid; addressed.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2729005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5064 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 15:15:55 +00:00
7d6bd22019
Propagate estimated RTT from receivers to rtt observer.
...
BUG=1613
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5063 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 12:14:34 +00:00
773e72797f
Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc
...
Chromium issue:
https://code.google.com/p/chromium/issues/detail?id=310146
BUG=2551
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2759004
Patch from Daniel Nicoara <dnicoara@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5061 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 01:51:21 +00:00
dce70ccb0b
Add delay limit to ChokeFilter.
...
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3079005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5058 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-30 19:18:07 +00:00
d6e46638ec
Logging for BWE test framework.
...
BUG=
R=stefan@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5055 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-30 16:06:26 +00:00
55e1723713
Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN.
...
BUG=2515
TEST=reproduced locally on linux and verified the fix resolves the issue.
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5048 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-29 04:40:09 +00:00
0aeb22e32c
Adding tl0idx consideration for continuity
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5046 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 22:26:14 +00:00
1a3a6e5340
Removing the threshold from the auto-mute APIs
...
The threshold is now set equal to the minimum bitrate of the
encoder. The test is also changed to have the REMB values
depend on the minimum bitrate from the encoder.
BUG=2436
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5040 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 10:16:14 +00:00
fe5d36b6fe
Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.
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We will do some refactoring of video engine and would like to use the
same rtcp stats struct there. Both video and audio seem to use 8bit
fraction lost, so that is changed in the struct as well.
BUG=
R=henrik.lundin@webrtc.org , kjellander@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5039 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 09:21:07 +00:00
c94abd313e
Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h
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R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5036 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-25 18:15:09 +00:00
0729460acb
Added a "interleaved_" flag to webrtc::AudioFrame.
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And also did some format refactoring on the AudioFrame class, no change on the functionalities on those format refactoring code.
BUG=
TEST=compile
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5032 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-25 12:50:46 +00:00
b56d0e383e
Change the low-bitrate handling in BitrateControllerImpl
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Changing to using strategy classes rather than having two different
derived classes of BitrateControllerImpl. This enables run-time switching
of the strategy, which is now possible through a new API. The reason is
that it must fit the current design of ViE.
BUG=2436
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5028 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-24 09:24:06 +00:00
37bb4974e7
Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals.
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R=juberti@google.com , mikhal@webrtc.org , stefan@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5027 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 23:59:45 +00:00
22858d4785
Add an extended filter option to audioproc.
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R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2609005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5024 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 14:07:17 +00:00
042e91c2b2
Fix for incorrect RTT estimation. A too low RTT value could be estimated.
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R=andrew@webrtc.org , holmer@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5023 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 13:58:31 +00:00
