Commit Graph

5363 Commits

Author SHA1 Message Date
7e19a11a71 Break out WebRtcNs_ComputeDdUpdate function in ns_core
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7514 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:54:33 +00:00
f8ea0d5518 Break out WebRtcNs_UpdateNoise function in ns_core
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7513 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:49:42 +00:00
799e88ae19 Break out FFT function in ns_core
This is done in order to make the code more readible and maintainable.
This introduces an error of only +1 and -1.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7512 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:36:42 +00:00
8454ad88ed Break out ComputeSnr function in ns_core
This is done in order to make the code more readible and maintainable.
The output is bit-exact.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7511 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:34:14 +00:00
580d367b14 Add macros and APIs for webrtc histograms.
BUG=crbug/419657

Code that links system_wrappers.gyp:system_wrappers should either:
- provide implementations for the APIs, or
- link with default implementations in system_wrappers.gyp:system_wrappers_default.

R=andresp@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7508 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 12:57:56 +00:00
82462aade0 Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate.
Also wires up a finch experiment to control this.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7505 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 11:57:05 +00:00
ad553a2731 Let video_loopback use internal VCM capturers.
R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7503 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 08:24:02 +00:00
3f8f5554a0 Disable TestVp8Impl.BaseUnitTest on MSan.
MemorySanitizer uses generic (non-optimized) libvpx which is not bit
exact. This may be a bug in upstream libvpx, but we're forced to disable
it now as it blocks launching the MSan bot.

R=stefan@webrtc.org
TBR=marpan@webrtc.org
BUG=3904

Review URL: https://webrtc-codereview.appspot.com/24089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7491 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 10:30:30 +00:00
76960d5f74 For FIR packet, payload length is zero, so SendToNetwork function is failing.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7490 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 09:47:14 +00:00
67cf1d742b Break out WebRtcNs_Windowing function in ns_core
This is done in order to make the code more readible and maintainable.
This introduces only +1 and -1 errors.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7488 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 22:35:40 +00:00
0e7099244c Break out WebRtcNs_Energy function in ns_core
This is done in order to make the code more readible and maintainable.
This generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7487 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 22:14:10 +00:00
7634c09406 Break out WebRtcNs_IFFT function in ns_core
This is done in order to make the code more readible and maintainable.
This creates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7486 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 21:27:00 +00:00
333e2556ed Break out WebRtcNs_UpdateBuffer function in ns_core
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7483 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 20:33:09 +00:00
def1e97ed2 Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests
BUG=3926
R=kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7481 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 12:48:29 +00:00
78ea06dd34 audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
Removed usage of trivial macro.

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

Review URL: https://webrtc-codereview.appspot.com/29789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7480 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 07:17:24 +00:00
913f7b8d5e Fix for glitches in ACM when switching desired output sample rate
The problem was that if the output sample rate is changed such from one
where no resampling is needed to a rate that requires resampling, the
first output from the resampler will contain an onset period. The
solution provided in this CL is to keep a copy of the last output frame
in ACM, and if the resampler is engaged, it will be primed with this
old frame before resampling the current frame.

BUG=3919
R=bjornv@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7479 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 06:54:23 +00:00
2abebe7baf audio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7474 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20 08:26:41 +00:00
a5ce7bbe17 audio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7473 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20 08:24:54 +00:00
b1dac33cac Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
BUG=3932
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/27779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 18:54:46 +00:00
f7030d4ed7 Query Android device orientation on every camera frame received.
Remove orientation listener from Android camera, since device
orientation change events are not well synchronized with actual
device display orientation. Plus these event may not be delivered
at all if device is in stationary position causing initial camera
frames appear rotated.

BUG=
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7467 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 16:25:06 +00:00
264e66f7a5 Add encoded_timestamp to AudioEncoder base class
BUG=3926
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7464 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-16 21:16:07 +00:00
9ea6f8a84d New interface class AudioEncoder
This class will be the base for new C++ wrapper classes for all
encoders.

BUG=3926
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7463 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-16 11:26:24 +00:00
f87c0aff7f audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
Also includes a typo in a comment.
Affects
* aecm
* hpf

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

Review URL: https://webrtc-codereview.appspot.com/29769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7456 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 12:51:23 +00:00
f02ba9be54 audio_processing/agc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
Affects AGC only.

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

Review URL: https://webrtc-codereview.appspot.com/31709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7455 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 11:16:48 +00:00
8dc00d76af audio_processing/ns: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
Affects fixed point version of Noise Suppression.

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

Review URL: https://webrtc-codereview.appspot.com/26809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7454 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 09:31:40 +00:00
99e561f6a6 Extend AcmSwitchingOutputFrequencyOldApi with more frequencies
Also reducing test duration, since the issue is triggered anyway.
The tests that are not failing are now enabled.

BUG=3919
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7453 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 08:50:00 +00:00
eb24b04f16 Add periodic logging of received RTP headers and estimated clock offsets for e2e delay.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7444 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 11:40:13 +00:00
81a78930ee New ACM test to trigger audio glitch when switching output sample rate
This CL implements a new unit test. The test is designed to trigger
a problem in ACM where switching the desired output frequency creates
a short discontinuity in the output audio. The problem itself is not
solved in this CL, but the failing test is disabled for now.

BUG=3919
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7443 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 10:49:58 +00:00
a57678a70e Workarounds for a bug in VS2013.3 linker when PGO is turned on.
See crbug.com/421607 for more details about this. This CL solve a linker bug when the PGO is turned on, without changing the behaviour or the performances.

BUG=crbug.com/421607
R=kwiberg@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26789005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7441 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 09:40:04 +00:00
b6af4283ca Adjust speech probability in NS when echo
The average speech probability for the higher band is multiplied by the quotient of the process and analyze powers, to avoid thinking that suppressed echo is speech. In order to do this both magnitudes, alanyze and process, needed to be stored. This also was used to calculate different previous STSA estimates for analyze and process.
This CL was tested on two long team member recordings (bjornv and kwiberg) and the noisiest (5) recordings from the QA set.

BUG=webrtc:3763
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7437 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 20:48:05 +00:00
bc1a4578e0 common_audio: Removed macro WEBRTC_SPL_RSHIFT_W16
Replaced the trivial right shift macro at remaining 4 places and removed from signal_processing.

Affected components:
* vad
* aecm

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

Review URL: https://webrtc-codereview.appspot.com/25849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7434 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 14:00:43 +00:00
a3722b643d iSAC tests: Type buffers as uint8_t[] to avoid casts
The iSAC interface functions now expect uint8_t arrays, so change some
arrays to be of that type instead of casting at each point of use.

R=bjornv@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7433 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 13:29:04 +00:00
d4fe824862 audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>
The implementation of WEBRTC_SPL_RSHIFT_W16 is simply >>. This CL removes the macro usage in audio_processing and signal_processing.

Affected components:
* aecm
* agc
* nsx

Indirectly affecting (through signal_processing changes)
* codecs/cng
* codecs/isac/fix
* codecs/isac/main

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

Review URL: https://webrtc-codereview.appspot.com/28699005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7432 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 13:01:13 +00:00
396a5e0001 WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[]
This patch changes WebRtcIsac_Decode, WebRtcIsac_DecodeRcu, and
WebRtcIsacfix_Decode so that they read the encoded data from a uint8
array instead of a uint16 array.

BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7431 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 11:23:24 +00:00
3f7f899a15 WebRtcIsac_UpdateBwEstimate et al.: Type byte streams as uint8, not uint16
This patch changes the signature of WebRtcIsac_UpdateBwEstimate,
WebRtcIsacfix_UpdateBwEstimate, and WebRtcIsacfix_UpdateBwEstimate1 so
that they expect the encoded data to be uint8 arrays, not uint16,
which is more natural. The implementations of the functions are left
unchanged for now.

BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7430 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 11:07:06 +00:00
1172988c79 Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[]
The affected functions are

  WebRtcIsacfix_ReadFrameLen
  WebRtcIsacfix_GetNewBitStream
  WebRtcIsacfix_ReadBwIndex

and

  WebRtcIsac_ReadFrameLen
  WebRtcIsac_GetNewBitStream
  WebRtcIsac_ReadBwIndex
  WebRtcIsac_GetRedPayload

BUG=909
R=aluebs@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7429 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 10:53:42 +00:00
c502df54f8 Merge the supporting to UYVY on Linux video capture in crbug/410202 to webrtc standalone.
BUG=3765
TEST=Manual
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7427 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 02:13:00 +00:00
651c05e4fc Release _inputSendPin & _outputCapturePin before _captureFilter & _sinkFilter since they should depend on the filters.
The previous steps work fine for all the webcam, but have problem on SplitCam driver as in the issue report.
Anyway it's always good to de-initial with the reversing order to initialization.

BUG=3845
TEST=Manual
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7426 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 02:11:55 +00:00
c87b74717b Adjust/increase rate control thresold for a vp9 test.
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7423 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 17:55:57 +00:00
573c78e31c Add VP9 codec to VCM and vie_auto_test.
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.
Passes trybots.

R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7422 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 16:44:47 +00:00
3cefbc99f4 Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
This also marks all virtual overrides of other classes in the same files. 

This will make a subsequent change I intend to do safer, where I'll change the 
argument types of the base Transport functions, by breaking the compile if I 
miss any overrides. 

This also highlighted a number of unused functions. I've removed some of these. 

TBR=mflodman@webrtc.org, pkasting@chromium.org
BUG=none 
TEST=none

Review URL: https://webrtc-codereview.appspot.com/28709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 09:42:53 +00:00
e46bc77e94 Reland 28629004: adding new AEC dump start interface for chrome.
This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
http://msdn.microsoft.com/en-us/library/ms235460.aspx

R=andresp@webrtc.org, andrew@webrtc.org, bjornv@webrtc.org, henrike@webrtc.org, henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7418 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 08:36:56 +00:00
4bd2db9a55 Opus wrapper: Use const for inputs and uint8[] for byte streams
About half of the functions already followed the desired pattern; this
patch fixes the other half.

BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7409 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 11:21:10 +00:00
2c0cdbce22 Estimating NTP time with a given RTT.
RemoteNtpTimeEstimator needed user to give a remote SSRC and it intended to call RtpRtcp module to obtain RTT, to be able to calculate Ntp time.

When RTT cannot be directly obtained from the RtpRtcp module with the specified SSRC, RemoteNtpTimeEstimator would fail.

This change allows RemoteNtpTimeEstimator to calculate NTP with an external RTT estimate.

An immediate benefit is that capture_start_ntp_time_ms_ can be obtained in a Google hangout call.

BUG=

TEST=chromium + hangout call
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7407 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 10:52:43 +00:00
c803907d87 Removing useless packets when inserting them (NetEq)
This is to save the buffer.

Some old code may become unnecessary, and will be removed in a separate CL.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7406 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 10:49:54 +00:00
3ea35fdb1b common_audio: Removed macro WEBRTC_SPL_LSHIFT_W16
The macro was a trivial << operation and where used has been replaced by <<. Affected components are
* ilbc
* isacfix

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

Review URL: https://webrtc-codereview.appspot.com/22919005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7404 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 08:47:02 +00:00
f71785cd3b audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>
Replaced trivial shift macro with >>. The actual implementation of the macro is simply >>.

Affected codecs:
* ilbc
* isac/fix

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

Review URL: https://webrtc-codereview.appspot.com/23889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7396 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-08 15:36:30 +00:00
5e3d7c78de Change name of a NetEq internal member variable
In the StatisticsCalculator class, the member last_report_timestamp_
was unfortunately named. It's now been changed to
timestamps_since_last_report_, which describes it more accurately.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7393 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-08 12:10:53 +00:00
a9e363e721 Roll chromium_revision c264a05..fc668e2 (297113:298195)
Mainly to pick up https://codereview.chromium.org/619723006/
to fix our MSan bot.

Summary of changes (git diff c264a05..fc668e2 DEPS):
* third_party/boringssl 01fe820..c7dd5f30
* third_party/usrsctp/usrsctplib 8975bd5..d5685d4
* tools/swarming_client 79940aee..33d442a

Clang updated 216630:217949 (git diff c264a05..fc668e2 tools/clang/scripts/update.sh)
This caused TSan v2 to hit an assert in libjingle_peerconnection_unittest
and libjingle_media_unittest, which is why the dlclose call
had to be disabled for now (webrtc:3895).

BUG=3895
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7385 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 12:49:34 +00:00
9103953b58 Fix neteq_rtpplay so that empty SSRC is valid
In r7380, the command line flag --ssrc was added to neteq_rtpplay.
However, it was not possible to omit that flag, since the validation
did not accept an empty string. This CL fixes that.

TBR=kwiberg@webrtc.org
BUG=2692

Review URL: https://webrtc-codereview.appspot.com/24869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7382 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 07:18:36 +00:00