32452b20b8
Make ReconfigureVideoEncoder use current bitrate.
...
Prevents bitrate drops when changing resolution etc.
R=stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/24069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7493 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 12:15:24 +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
b69ea9a35a
common_audio: Replaced invalid operand in min_max_operations_neon.S"
...
Vector Move immediate can not load #0x7FFF. Changed to us vdup from already loaded register.
BUG=N/A
TESTED=ios and android trybots
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7477 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20 14:08:35 +00:00
b35b136480
Make avg_{psnr,ssim}_threshold_ const.
...
Triggered warning on next clang version being rolled as these variables
are annotated to be protected by crit_.
R=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/24949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7475 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20 09:14:38 +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
28100cb388
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
...
BUG=N/A
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 22:03:39 +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
0371a37f85
Moving creating TURN configration to the host machine instead of the bots - rtcBot
...
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7468 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 16:43:50 +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
c221db6165
Test names changed from e.g) testOneWayVideo/chrome=>chrome to testOneWayVideo/chrome-chrome.
...
Because the symbol ">" is interpreted as special command for output to file in bash commands.
TBR= andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7465 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 09:13:43 +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
458c2c3b06
Improve rtcbot to load all test files at start and allow them to registerTests
...
via: registerBotTest. After loading all tests main.js starts running the
requested one on the command arguments.
R=houssainy@google.com
Review URL: https://webrtc-codereview.appspot.com/29779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7461 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-16 07:36:37 +00:00
9aed002090
Add ability to include a larger time span (in addition to encode time) for measuring the processing time of a frame.
...
Controlled by setting enable_extended_processing_usage. Enabled by default.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7460 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-16 06:57:12 +00:00
d1ba6d9cbf
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
...
BUG=3379
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27709005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 17:30:28 +00:00
3e2f8ff36c
Selecting bot_type changed to be specified in the test file
...
Selecting bot_type changed to be specified in the test file instead of
specify it in the running command.
Now we can write test for rtcBot that run one bot on chrome for android
and the other bot on chrome for desktop.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7458 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 15:01:11 +00:00
e93cbd13d5
Fix data races in ThreadTest.ThreeThreadsInvoke.
...
R=henrike@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/26819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7457 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 14:54:56 +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
fab5439112
common_audio: Removed version API from signal_processing
...
The Signal Processing version API is not used anymore.
BUG=3353
R=kwiberg@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7451 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 04:38:42 +00:00
a73a678e25
Remove -1 from Call::Config::start_bitrate_bps.
...
Instead initialize it to a good default value. The code does the same,
but we don't have to check explicitly for -1.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/23989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7445 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 11:52:10 +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
c216b9aeaf
Add a packet loss full stack test to the new API.
...
Remove all full stack tests for the old API.
BUG=3750
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7442 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 10:38:49 +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
7f7b0a1cdd
Re-enable ThreadCheckerDeathTest.MethodNotAllowedOnDifferentThreadInDebug (missed when enabling other base tests).
...
BUG=3836
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/24909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7425 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 21:41:55 +00:00
4ddbbed16e
Disable SendsAndReceivesVP9 test for now.
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Fails on linux memcheck and DrMemory.
Will re-enable on next libvpx roll.
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7424 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 21:25:20 +00:00
c87b74717b
Adjust/increase rate control thresold for a vp9 test.
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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.
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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.
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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
