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1915 Commits

Author SHA1 Message Date
aluebs@webrtc.org
2693a54614 Add WEBRTC_BEAMFORMER define to BUILD.gn 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8034 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 23:26:13 +00:00
andrew@webrtc.org
8f27fcce79 Revert 8028 "Support associated payload type when registering Rt..." 
Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.

> Support associated payload type when registering Rtx payload type.
> 
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
> 
> BUG=4024
> R=pbos@webrtc.org, stefan@webrtc.org
> TBR=mflodman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/26259004
> 
> Patch from Changbin Shao <changbin.shao@intel.com>.

TBR=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 20:22:46 +00:00
andrew@webrtc.org
f3fd8e7cdf Add NEON intrinsics version for transform_neon 
WebRtcIsacfix_Time2SpecNeon and WebRtcIsacfix_Spec2TimeNeon are added.
TransformTest in modules_unittests is passed on ARM32/ARM64 platform.

Initially reviewed here:
https://webrtc-codereview.appspot.com/36449004/

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: I0920ff66a0a0f529707fd7e6619f91e271a47019

Review URL: https://webrtc-codereview.appspot.com/31309004

Patch from Yang Zhang <yang.zhang@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 18:29:37 +00:00
pbos@webrtc.org
2a169640a3 Support associated payload type when registering Rtx payload type. 
Major changes include,
- Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
- Receiver: Restore RTP packets by the new RTX-APT map.
- Sender: Send RTP packets by checking RTX-APT map.
- Add RTX payload type for RED in the default codec list.

BUG=4024
R=pbos@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26259004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 15:16:10 +00:00
kjellander@webrtc.org
8649fed1b8 GN: Fix Windows build. 
This required a tiny include fix in
src/third_party/winsdk_samples/src
which was committed in
https://code.google.com/p/webrtc/source/detail?r=7951

This incorporates contribution from vchigrin@yandex-team.ru
in https://webrtc-codereview.appspot.com/29299004/

BUG=261,1348,4105
R=pbos@webrtc.org
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8027 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-08 21:22:01 +00:00
bjornv@webrtc.org
758d6d431e audio_processing/aecm: Removed usage of macro WEBRTC_SPL_MUL_16_16 
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)

The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)

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

Review URL: https://webrtc-codereview.appspot.com/36639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8025 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-08 17:52:56 +00:00
bjornv@webrtc.org
dec649cbab audio_processing/ns: Replaced WEBRTC_SPL_MUL_16_16 macro with * 
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)

The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)

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

Review URL: https://webrtc-codereview.appspot.com/35619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8024 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-08 17:34:33 +00:00
bjornv@webrtc.org
5e5b32706a audio_processing/agc: Removed usage of macro WEBRTC_SPL_MUL_16_16 in legacy/agc 
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)

The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)

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

Review URL: https://webrtc-codereview.appspot.com/39389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8023 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-08 17:25:34 +00:00
pbos@webrtc.org
3663fb08ff Reenable dlclose() for InternalUnloadDll on TSan. 
Upstream TSan bug has been fixed and dlclose() no longer needs to be
excluded.

R=henrika@webrtc.org
BUG=3895

Review URL: https://webrtc-codereview.appspot.com/30099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8016 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 18:02:39 +00:00
aluebs@webrtc.org
fb7a039e9d Use array geometry in Beamformer 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8000 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-05 21:58:58 +00:00
aluebs@webrtc.org
e5a921a82d Use tmp files in file_utils_unittests 
The static file names were breaking when executing tests in parallel. This fixes it.

BUG=4138
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7997 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-05 18:45:22 +00:00
marpan@webrtc.org
c4ad157d8d Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9. 
BUG=4059

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7994 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-05 17:31:34 +00:00
bjornv@webrtc.org
bac0012120 Extend delay estimation window in AEC to 500 ms on all platforms 
On non-Android the delay estimator in audio_processing/aec has solely been used for logging purposes. The maximum possible observed delay has been 236 ms. We have seen longer delays for which the delay estimate at best ends up at 236 ms, but can also be 'random'. reported delays are clamped to 500 ms.
This cl extends the delay estimation window to match that.

BUG=4086, 3504, 4113
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7989 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-02 09:23:49 +00:00
bjornv@webrtc.org
3a70625caf audio_processing: Added back ATTRIBUTE_UNUSED lost in r7877 
BUG=N/A
TESTED=Now it builds with aec_debug_dump=1 on Mac
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7986 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-01 22:04:12 +00:00
andrew@webrtc.org
84d84471f5 Minor fixes regarding accumulator usage on MIPS platforms. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33729004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7979 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-29 17:08:44 +00:00
sprang@webrtc.org
46d4d29a75 Add field trial for screenshare bitrates when using temporal layers. 
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7976 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-23 15:19:35 +00:00
tkchin@webrtc.org
ed1a48b0cd Fix mac video capture leak. 
BUG=3878
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7971 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 20:51:02 +00:00
aluebs@webrtc.org
ae643ce280 Wire up Beamformer in AudioProcessing 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7969 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 19:57:34 +00:00
pbos@webrtc.org
5570769210 Remove the last getters from VideoReceiveStream stats. 
R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/32899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7965 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 15:45:03 +00:00
pbos@webrtc.org
d16e839c6d Rtp-Rtcp sender cleanup. 
Some setter functions from Rtp and Rtcp Sender never return negative values. Remove return results from those functions.

Also removed const on non-pointer/reference types for related files.

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

Review URL: https://webrtc-codereview.appspot.com/34469004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7962 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 13:49:55 +00:00
kjellander@webrtc.org
556caffb36 GN: Fix build for Mac 
BUG=4105
R=henrika@webrtc.org, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 13:28:37 +00:00
stefan@webrtc.org
11d8176cb3 Move updating nack bitrate inside UpdateNACKBitRate. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7960 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 09:52:24 +00:00
aluebs@webrtc.org
0c39e91cc8 Merge beamformer 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7958 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 22:22:04 +00:00
andrew@webrtc.org
1090a6eccf Remove obsolete target_arch == armv7. 
Also, use arm_version >= 7 so things will continue to work when building
for ARMv8 and higher targets.

BUG=3906
R=kjellander@webrtc.org, tkchin@webrtc.org, zhongwei.yao@arm.com

Review URL: https://webrtc-codereview.appspot.com/38379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7957 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 21:36:18 +00:00
asapersson@webrtc.org
cb79141eab Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc. 
When using rtx, receiver reports with two report blocks are received. The report blocks have the same remote ssrc and therefore the first report block was overwritten by the second report block when stored in the ReportBlockInfoMap.

Removed unused function ResetRTT.

BUG=4114
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33659005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7952 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 14:30:32 +00:00
pbos@webrtc.org
ce4e9a3562 Refactor some receive-side stats. 
Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 13:50:16 +00:00
sprang@webrtc.org
9b79197c80 Suppress REMB in bitrate ctrl if it seems lika a short network glitch. 
BUG=4082
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7948 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 11:53:59 +00:00
pbos@webrtc.org
f832a6d090 Remove _t from function pointer typedefs. 
_t are reserved in POSIX.

R=bjornv@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/34539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7947 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 09:56:09 +00:00
henrik.lundin@webrtc.org
eed7a22bbf Make an AudioEncoder subclass for iSAC redundant encoding 
Adding unit test, too.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36559005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7946 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 09:52:36 +00:00
pbos@webrtc.org
dd8f6f3d48 Rename rtpDumpPktHdr_t to RtpDumpPacketHeader. 
_t names are reserved in POSIX.

BUG=162
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7945 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 09:18:42 +00:00
pbos@webrtc.org
e468bc9e60 Rename _t struct types in audio_processing. 
_t names are reserved in POSIX.

R=bjornv@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/34509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7943 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 09:11:33 +00:00
henrik.lundin@webrtc.org
cab1291745 Fixing the memory leak in AudioEncoderCopyRedDeathTest.NullSpeechEncoder 
Re-enable the test and explicitly call delete on red, even though the
test should die in the AudioEncoderCopyRed cunstructor. Apparently,
things work a little differently under memcheck.

BUG=4108, 3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7942 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 06:58:42 +00:00
pbos@webrtc.org
eb544460e4 Rename _t struct types in audio_coding. 
_t names are reserved in POSIX.

R=henrik.lundin@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/34509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7933 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 15:23:29 +00:00
pbos@webrtc.org
e728ee03ba Remove or rename typedefs with _t prefixes. 
_t prefixes are reserved for additional typenames in POSIX.

R=henrik.lundin@webrtc.org, hta@webrtc.org, stefan@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/36559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7931 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 13:43:55 +00:00
sprang@webrtc.org
70f74f3f7b Add overshoot of target bitrate for screenshare with temporal layers. 
Set the codec target bitrate higher than TL0 but lower than TL1, making
sure frame rate is not too low (but still lower than TL1) and that
overshooting for complex scenes don't overly exceed TL1 bitrates.

BUG=4083
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7929 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 10:57:10 +00:00
kwiberg@webrtc.org
e102e8147b Enable the iSACfix AudioDecoder test (and make it work again) 
As far as I can tell, the test should have been enabled again once
https://code.google.com/p/webrtc/issues/detail?id=1353 was fixed, but
it wasn't, and has rotted a bit as a result. I'm not sure why the
number of encoded bytes have changed, but the output seems to be
correct (EncodeDecodeTest encodes, decodes, and compares the result
with the original).

The DecodePlc change is necessary because r7912 added support for that
to the iSACfix AudioDecoder.

BUG=1353, 3926
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7927 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 07:30:23 +00:00
andrew@webrtc.org
971bf557e2 Fix path to mock_agc.h 
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7924 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 22:28:20 +00:00
henrik.lundin@webrtc.org
a32487f97b Disable AudioEncoderCopyRedDeathTest.NullSpeechEncoder 
Fails linux memcheck.

BUG=4108
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7920 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 21:04:55 +00:00
andrew@webrtc.org
08df9b2841 Add a manageable command-line tool for AudioProcessing. 
This is the start of a replacement for the venerable and unwieldly
process_test.cc (aka audioproc). It will be limited to:
- Reading WAV or aecdebug protobuf files.
- Calling the float AudioProcessing interface.
- Requiring aecdebug files for running bi-directional stream
components (e.g. AEC).

This initial version only handles WAV files.

R=aluebs@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7918 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 20:57:15 +00:00
aluebs@webrtc.org
cf6d0b64ef Add 48kHz support to AGC 
Doing the same for the 16-24kHz band than was done in the 8-16kHz.
Results look and sound as nice.

Originally reviewed here:
https://webrtc-codereview.appspot.com/26339004/

BUG=webrtc:3146
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7917 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 20:56:09 +00:00
pbos@webrtc.org
451a133f44 Add AGC manager tests. 
R=bjornv@webrtc.org
BUG=4098

Review URL: https://webrtc-codereview.appspot.com/35539005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7914 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 14:48:47 +00:00
henrik.lundin@webrtc.org
c1c9291e9b Make an AudioEncoder subclass for RED 
This class only supports the simple case of payload duplication. That
is, one single encoder is used, and the redundant payload is a one-step
delayed payload.

BUG=3926
R=kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 13:41:36 +00:00
kwiberg@webrtc.org
88bdec8c3a AudioEncoder subclass for iSACfix 
This patch refactors AudioEncoderDecoderIsac so that it can share
almost all code with the very similar AudioEncoderDecoderIsacFix.

BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7912 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 12:49:37 +00:00
kjellander@webrtc.org
0198933b3d Cleanup: Remove 'const' qualifier from OnReceivedEstimatedBitrate(). 
This should fix the following error I'm seeing in Win8 GN trybot:

e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\bitrate_controller\bitrate_controller_impl.cc(78)
: error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\bitrate_controller\bitrate_controller_impl.cc(30)
: warning C4373:
'webrtc::BitrateControllerImpl::RtcpBandwidthObserverImpl::OnReceivedEstimatedBitrate':
virtual function overrides 'webrtc::RtcpBandwidthObserver::OnReceivedEstimatedBitrate',
previous versions of the compiler did not override when parameters only differed by const/volatile qualifiers
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\rtp_rtcp\interface\rtp_rtcp_defines.h(286)
: see declaration of 'webrtc::RtcpBandwidthObserver::OnReceivedEstimatedBitrate'

http://build.chromium.org/p/tryserver.chromium.win/builders/win8_chromium_gn_dbg/builds/23/steps/compile/logs/stdio

The above was triggered in CL https://codereview.chromium.org/802113002/

BUG=None
R=kjellander@google.com, kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37409004

Patch from Thiago Farina <tfarina@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7911 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 12:29:59 +00:00
asapersson@webrtc.org
d08d389ce8 Add field to counters for when first rtp/rtcp packet is sent/received. 
Use this time for histogram statistics (send/receive bitrates, sent/received rtcp fir/nack packets/min).

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7910 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 12:03:11 +00:00
bjornv@webrtc.org
b395a5ea65 audio_processing: Moved legacy AGC code to webrtc/modules/audio_processing/agc/legacy/ 
include/ is renamed to legacy/ and analog_agc.* and digital_agc.* moved into the directory.

BUG=
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7909 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 10:38:10 +00:00
pkasting@chromium.org
0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. 
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 22:09:40 +00:00
aluebs@webrtc.org
96a626262a Remove 20ms support in AGC 
Today, 10ms is the standard chunk length used in whole AudioProcessing, so this was only adding unnecessary complexity and maintainance.
Removing it doesn't change the bahavior in any use case of today.

R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7904 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 21:54:50 +00:00
pbos@webrtc.org
a7f77720cb Merge in AGC manager and AGC tools. 
R=bjornv@webrtc.org
BUG=4098

Review URL: https://webrtc-codereview.appspot.com/37379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7902 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 16:33:16 +00:00
bjornv@webrtc.org
903b4ae603 Removes unused test files by audio_processing/transient 
BUG=
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7901 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 16:13:05 +00:00