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Author SHA1 Message Date
minyue@webrtc.org
0040a6ef97 This is a setup to solve 
https://code.google.com/p/webrtc/issues/detail?id=1906

In particular, we add an API to call Opus's set maximum bandwidth to prevent the encoder from coding audio content beyond this bandwidth so as to increase computation and transmission efficiency (without affecting sampling rate).

BUG=
R=henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6817 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 14:41:57 +00:00
asapersson@webrtc.org
84b9e1e9d9 Fix for retransmission. Base layer packets were not retransmitted. 
Issue introduced in r6669.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6816 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 11:59:42 +00:00
stefan@webrtc.org
e1c9caf6ee Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804. 
TEST=buildtools/linux64/gn gen out/Default
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-31 15:07:59 +00:00
stefan@webrtc.org
2ec560606b Add H.264 packetization. 
This also includes:
- Creating new packetizer and depacketizer interfaces.
- Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition.
- Created a Create() factory method for packetizers and depacketizers.

R=niklas.enbom@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-31 14:59:24 +00:00
stefan@webrtc.org
bfe6e08195 Add simulation of network effects to video_loopback tool. 
Also add support for uniform random packet loss to the fake network pipe.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6803 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-31 12:30:18 +00:00
andrew@webrtc.org
fdbe1442c5 Use C functions in aec for MIPS 
With GCC 4.9, the MIPS NDK toolchain has been changed to only support 16 spregs by default - the even-numbered ones. This has been changed to support the R6 MIPS architecture. While the old behaviour could be restored by adding "-modd-spreg", this would come with a performance hit because the kernel would emulate odd-numbered spregs and missing R2 instructions.
As a result of this change, the functions removed in this CL no longer compile as there are no longer enough spregs for the compiler to assign. So we are removing these functions and they will use the C implementation until the MIPS code is rewritten.

R=andrew@webrtc.org, ljubomir.papuga@gmail.com, pasko@chromium.org

Review URL: https://webrtc-codereview.appspot.com/16159005

Patch from Fabrice de Gans-Riberi <fdegans@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-29 14:39:10 +00:00
asapersson@webrtc.org
e75d78d32d Integrate rtcp packet class to rtcp receiver tests. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-29 08:21:50 +00:00
henrike@webrtc.org
1e7d60e451 merge_libs.py: fixes Windows breakage: there should be no space after "lib /OUT:". 
TBR=andrew@webrtc.org
BUG=b/15773179

Review URL: https://webrtc-codereview.appspot.com/16999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6793 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-29 04:45:23 +00:00
henrike@webrtc.org
961293d469 webrtc/base: FileModifyTime -> OlderThan as that's what it was ever used as. Needed for cl/70828325. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6787 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 21:58:50 +00:00
sergeyu@chromium.org
af9e7943d1 Fix compilation on windows with clang, indentation cleanups 
R=henrike@webrtc.org, thakis@chromium.org
TBR=hellner@chromium.org

Committed: https://code.google.com/p/webrtc/source/detail?r=6779

Review URL: https://webrtc-codereview.appspot.com/18849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6786 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 19:42:19 +00:00
henrike@webrtc.org
c7b8f39e2e Fixes "argument list too long" problem on Linux by using the "find" command instead of re-implementing one in python. 
R=pbos@webrtc.org
TBR=andrew@webrtc.org
BUG=b/15773179

Review URL: https://webrtc-codereview.appspot.com/18929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 18:36:55 +00:00
turaj@webrtc.org
1ebd2e96df Remove timestamp retreival warning/error. 
An error reported while retreiving playout timestamp if no RTP packet received, yet. This causes an overflow of errors/warnings in applications where few channel are created but only one is actively engaged in a conversation. Therefore, we don't find such logging informative (there is no check upon correctness of timestamp computaion only if a packet already received).

BUG=3545
TEST=manual with voe_cmd_test,try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 17:50:10 +00:00
sergeyu@chromium.org
2386882266 Revert "Fix compilation on windows with clang, indentation cleanups" 
This reverts commit f628eaedfeea97e13c63c78dd42f2b1c76723619.

TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 17:37:12 +00:00
sergeyu@chromium.org
a44fce5920 Fix compilation on windows with clang, indentation cleanups 
R=henrike@webrtc.org, thakis@chromium.org
TBR=hellner@chromium.org

Review URL: https://webrtc-codereview.appspot.com/18849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 17:28:25 +00:00
mflodman@webrtc.org
f9460688a6 Make sure padding is sent on the first sending RTP module. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-24 16:41:25 +00:00
stefan@webrtc.org
f24c4a3b8d Fix flaky ramp-up test. 
Don't require the first estimate to be less than the target bitrate. There are other tests verifying that BWE works, so it's enough for this test to measure the
time it takes to ramp-up.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6764 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 10:27:41 +00:00
minyue@webrtc.org
194fea7640 The lastest commit on this file was in 
https://webrtc-codereview.appspot.com/15529004/

The final patch set should have included this, but was missed.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6755 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-22 09:55:51 +00:00
andresp@webrtc.org
b0c8228755 Remove no longer used SkipEncodingUnusedStreams. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6753 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-22 07:17:17 +00:00
andresp@webrtc.org
5ab7616983 Remove remains of WEBRTC_NO_STL. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-22 06:48:58 +00:00
andrew@webrtc.org
ceafa8cce9 MIPS optimizations for ISAC (patch #2) 
Implemented functions:
- WebRtcIsacfix_CalculateResidualEnergy
- WebRtcIsacfix_Spec2Time
- WebRtcIsacfix_Time2Spec
- WebRtcIsacfix_HighpassFilterFixDec32
- WebRtcIsacfix_PCorr2Q32

Gain achieved: aprox. further 5% on top of patch#1 on ISAC encoding path.
The optimizations are bit-exact to the C code, with the excception of the
MIPS DSPr2 variant of the WebRtcIsacfix_Time2Spec function (the accuracy of
the WebRtcIsacfix_Time2Spec MIPS DSPr2 variant is same or better than C
variant). Code verification and improvement achieved have been determined
using the iSACFixtest application.

R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19749004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6749 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-21 16:43:13 +00:00
pbos@webrtc.org
3c10758b3b Check before send/receive rtp header extensions. 
BUG=1788
R=pbos@webrtc.org, tommi@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13949004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6744 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-20 15:27:35 +00:00
minyue@webrtc.org
f563e85ab0 This is to re-open an earlier CL 
https://webrtc-codereview.appspot.com/16619005/

which is reverted due to an issue in audio conference mixer.

This issue has been solved in
https://webrtc-codereview.appspot.com/20779004/

BUG=webrtc:3155
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18819005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 21:11:27 +00:00
tkchin@webrtc.org
ff50debd37 Runtime guard for iOS7 property. 
BUG=3487
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 17:17:59 +00:00
tkchin@webrtc.org
9343cf67a9 Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS. 
BUG=3581
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 17:13:28 +00:00
minyue@webrtc.org
026859b983 This is related to an earlier CL of enabling Opus 48 kHz. 
https://webrtc-codereview.appspot.com/16619005/

It was reverted due to a build bot error, which this CL is to fix. The problem was that when audio conference mixer receives audio frames all at 48 kHz and mixed them, it uses Audio Processing Module (APM) to do a post-processing. However the APM cannot handle 48 kHz input. The current solution is not to allow the mixer to output 48 kHz.

TEST=locally solved https://webrtc-codereview.appspot.com/16619005/

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 12:28:28 +00:00
pbos@webrtc.org
e9e4253a3c Sleep in ThreadTest thread functions. 
Prevents busy loops that really mess up Valgrind's thread scheduling,
this brings runtimes from up to minutes down to milliseconds.

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 10:12:50 +00:00
kwiberg@webrtc.org
e364ac902f AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float 
Specifically, when someone asks for a const pointer to the int16
version of the array, there's no need to invalidate the float version
of that array, and vice versa. (But obviously, invalidation still has
to happen when someone asks for a non-const pointer.)

R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6725 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 07:50:29 +00:00
andrew@webrtc.org
c145668dc8 Reduce runtime of RingBufferTest by a factor of 100. 
This test was needlessly long.

TBR=pbos

Review URL: https://webrtc-codereview.appspot.com/15029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6724 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 23:16:44 +00:00
wu@webrtc.org
4f5da030f1 Use _numMixedParticipants instead of audioFrameList->size() to determine if there're more than one participants. 
There are two audioFrameLists. The previous check wouldn't work correctly if each list had a single member.

TEST=chrome https://apprtc.appspot.com/?debug=loopback&video=false and verify e2e delay stats
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6723 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 22:19:21 +00:00
stefan@webrtc.org
8b94e3da0f Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled. 
This broke bandwidth estimation for calls without abs-send-time is enabled, but where RTX was.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 16:10:14 +00:00
aluebs@webrtc.org
4065988108 Remove unused ExperimentalNS API in AudioProcessing 
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6718 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 11:32:09 +00:00
kwiberg@webrtc.org
2b6bc8d84f AudioBuffer: Eliminate the SplitChannelBuffer class 
It's just a container for two IFChannelBuffers, and doesn't earn its
keep. The main problem is that the number of methods it needs that
just forward calls to either of its two IFChannelBuffers was already
large, and was about to grow.

R=aluebs@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6717 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 09:46:37 +00:00
aluebs@webrtc.org
2561d52460 Simplify AudioBuffer::mixed_low_pass_data API 
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6715 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 08:27:39 +00:00
kwiberg@webrtc.org
af93fc08a1 AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter 
R=aluebs@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6714 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 08:18:33 +00:00
kwiberg@webrtc.org
2ade42bd96 Add unit test for MediaFile WAV file writing 
R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 08:11:32 +00:00
tkchin@webrtc.org
4a472fb18d Fixes up rtc so that it compiles on iOS 8 SDK. 
Adds support for UIInterfaceOrientationUnknown (new with in SDK) and makes it the same as
UIInterfaceOrientationPortrait.

R=noahric@google.com, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13029004

Patch from David Maclachlan <dmaclach@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6712 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 00:21:59 +00:00
minyue@webrtc.org
c56ae63ea6 r6709 lacks a change in BUILD.gn 
BUG=
R=marpan@google.com, marpan@webrtc.org, pbos@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6710 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 22:18:49 +00:00
minyue@webrtc.org
74aaf29a0f Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module. 
The filter is an exponential filter borrowed from video coding module.

The method is written in a new class called PacketLossProtector (not sure if the name is nice), which can be used in the future for more sophisticated logic.

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

Review URL: https://webrtc-codereview.appspot.com/20809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6709 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 21:28:26 +00:00
tkchin@webrtc.org
2e3c97ddf5 Compile-time guard for iOS7 specific property. 
BUG=3487
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 19:59:05 +00:00
stefan@webrtc.org
4070b1db53 Print an info log instead of return an error if an external encoder is de-registered, but no corresponding internal encoder can be registered automatically. 
This is not an error case if for instance an external h.264 encoder is registered, but no internal implementation exists.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6704 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 11:20:40 +00:00
pbos@webrtc.org
63c60ed224 Remove old padding path in RTPSender. 
Removing RTPSender::SendPaddingAccordingToBitrate() as well as a couple
of arguments from SendPadData().

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6703 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 09:37:29 +00:00
kwiberg@webrtc.org
efb81d8d1f int16<->float conversions: Use size_t for array length argument, not int 
size_t is more appropriate for array lengths, since int might
theoretically be too small for a really large array. But more
importantly, if the caller's value is naturally of type size_t and the
function requires an int, VC++ will trigger warning C4267
(http://msdn.microsoft.com/en-us/library/6kck0s93.aspx) because the
implicit cast might be lossy, forcing the caller to do a manual cast.
Typing the function with size_t in the first place resolves the
problem.

R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6702 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 08:36:52 +00:00
kwiberg@webrtc.org
0fa6366ed1 Define convenient FATAL_ERROR() and FATAL_ERROR_IF() macros 
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 08:34:58 +00:00
kwiberg@webrtc.org
e8ea33ccb1 nrsh1 is written before tmp321 is read, so needs to be earlyclobber 
Otherwise, the compiler is allowed to put them in the same register
under the assumption that all inputs are read before any
(non-earlyclobber) output is written, which in this case would result
in nrsh2 being corrupted.

BUG=3439
R=aluebs@webrtc.org, ljubomir.papuga@gmail.com

Review URL: https://webrtc-codereview.appspot.com/16089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6700 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 08:26:48 +00:00
jiayl@webrtc.org
bac5f0fb56 Fix an invalid memory access due to typo in win/cursor.cc. 
BUG=crbug/391468
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/19949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 20:32:03 +00:00
tkchin@webrtc.org
122caa51b1 After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue. 
CL also replaces deprecated AudioSession calls with equivalent AVAudioSession ones.

BUG=3487
R=glaznev@webrtc.org, noahric@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6697 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 20:20:47 +00:00
tkchin@webrtc.org
42fe4350fe Remove Thread::RunningForChannelManager(). 
I haven't heard of this failing, so it should be safe to remove. Let me know if this isn't the case.

BUG=3388
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6695 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 17:52:43 +00:00
stefan@webrtc.org
89fd1e8e99 Improvements to the pacer where it lost some budget due to truncation errors. 
With this CL the resolution is increased to microseconds and proper rounding
is done in the Process() function. This means that we will be allowed to send
more than prior to r6664 as we previously truncated away parts of our budget.

We will also not lose budget due to inaccurate calculations in
TimeUntilNextProcess(), which was a regression in r6664.

BUG=cr/393950
TEST=out/Debug/webrtc_perf_tests --gtest_filter=RampUpTest.Simulcast
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6694 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 16:40:38 +00:00
pbos@webrtc.org
376b4ea93f Fix breakage introduced by r6691. 
ModuleRtpRtcpImpl returned incorrectly on RemoteNTP as the
RTCPReceiver::NTP changed return type.

BUG=
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6693 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 15:51:33 +00:00
pbos@webrtc.org
2f4b14e3f3 Make RTCP sender report send media bytes. 
r6654 changed RtpSender::Bytes() to return the number of bytes sent
instead of number of media bytes. This is used by VideoEngine for stats.
This change broke RTCP which sends this same count as the number of
payload bytes sent (excluding headers and padding).

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 15:25:39 +00:00