026b892e72
Using << on an int8_t or uint8_t will output a character rather than a number.
...
Places that do this need to cast to int to get the desired behavior.
BUG=none
TEST=none
R=henrik.lundin@webrtc.org , pthatcher@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40579004
Cr-Commit-Position: refs/heads/master@{#8223}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8223 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 19:54:19 +00:00
7a37bfc240
Revert 8203 "Reducing locking in OveruseFrameDetector and increa..."
...
Broke tests in Chrome for some reason:
[ RUN ] WebRtcAecDumpBrowserTest.CallWithAecDump
[80131:1287:0129/074432:30561723987517:ERROR:vt_video_decode_accelerator.cc(132)] Failed to create VTDecompressionSession: codecOpenErr (-8973)
[80129:1287:0129/074432:30562276677373:INFO:CONSOLE(64)] "Looking at video in element remote-view-1", source: http://127.0.0.1:61401/media/webrtc_test_utilities.js (64)
[80129:1287:0129/074432:30562281435788:INFO:CONSOLE(64)] "Looking at video in element remote-view-2", source: http://127.0.0.1:61401/media/webrtc_test_utilities.js (64)
[80129:1287:0129/074432:30562315329399:INFO:CONSOLE(800)] "Negotiating call...", source: http://127.0.0.1:61401/media/peerconnection-call.html (800)
[80133:29187:0129/074432:30562402039578:FATAL:overuse_frame_detector.cc(388)] Check failed: processing_thread_.CalledOnValidThread().
0 libbase.dylib 0x000000010dfd688f base::debug::StackTrace::StackTrace() + 47
1 libbase.dylib 0x000000010dfd68e3 base::debug::StackTrace::StackTrace() + 35
2 libbase.dylib 0x000000010e030076 logging::LogMessage::~LogMessage() + 70
3 libbase.dylib 0x000000010e02f0c3 logging::LogMessage::~LogMessage() + 35
4 libcontent.dylib 0x000000011d8c0cd5 webrtc::OveruseFrameDetector::TimeUntilNextProcess() + 245
5 libcontent.dylib 0x000000011d31ddfd webrtc::ProcessThreadImpl::Process() + 525
6 libcontent.dylib 0x000000011d31d836 webrtc::ProcessThreadImpl::Run(void*) + 38
7 libcontent.dylib 0x000000011d10c390 webrtc::ThreadPosix::Run() + 288
8 libcontent.dylib 0x000000011d10c076 webrtc::StartThread(void*) + 38
9 libsystem_pthread.dylib 0x00007fff8e667899 _pthread_body + 138
10 libsystem_pthread.dylib 0x00007fff8e66772a _pthread_struct_init + 0
11 libsystem_pthread.dylib 0x00007fff8e66bfc9 thread_start + 13
> Reducing locking in OveruseFrameDetector and increasing constness.
>
> I also added a few TODOs there to see what we can do to reduce the chance of contention.
> To catch regressions, I've started using the ThreadChecker class on the processing thread but it might also be a good idea to add similar checks for other known threads such as the thread we receive frames on. I'm sure we can reduce locking even further.
>
> BUG=2822
> R=asapersson@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/33129004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34079004
Cr-Commit-Position: refs/heads/master@{#8206}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8206 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 16:09:07 +00:00
18e758526d
Reducing locking in OveruseFrameDetector and increasing constness.
...
I also added a few TODOs there to see what we can do to reduce the chance of contention.
To catch regressions, I've started using the ThreadChecker class on the processing thread but it might also be a good idea to add similar checks for other known threads such as the thread we receive frames on. I'm sure we can reduce locking even further.
BUG=2822
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33129004
Cr-Commit-Position: refs/heads/master@{#8203}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8203 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 12:35:19 +00:00
a26f511dd2
Remove frame copy in ViEExternalRendererImpl::RenderFrame
...
Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.
BUG=1128,4227
R=mflodman@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8136
Review URL: https://webrtc-codereview.appspot.com/36489004
Cr-Commit-Position: refs/heads/master@{#8199}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8199 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 11:45:43 +00:00
a907e01c63
Adding constness.
...
Make a few member variables in the Transport class officially const so that it's clear that locking isn't needed for access. There are getters for some of these (e.g. content_name()) that don't have locking or checking, so making the variables const is at least a way to guard against regressions. Also making the clock_ member in overuse_frame_detector.h const for clarity that it doesn't require a lock for access.
No code change.
Review URL: https://webrtc-codereview.appspot.com/35949004
Cr-Commit-Position: refs/heads/master@{#8186}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8186 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 17:33:45 +00:00
37c0559c1e
Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets).
...
Don't copy codec specific header for empty packets in the jitter buffer.
BUG=3135
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37659004
Cr-Commit-Position: refs/heads/master@{#8184}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8184 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 13:58:40 +00:00
273fbbb921
Update StreamDataCounter with FEC bytes.
...
Add histograms stats for send/receive FEC bitrate:
- "WebRTC.Video.FecBitrateReceivedInKbps"
- "WebRTC.Video.FecBitrateSentInKbps"
Correct media payload bytes in StreamDataCounter to not include FEC bytes.
Fix stats for rtcp packets sent/received per minute (regression from r7910).
BUG=crbug/419657
R=holmer@google.com , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8164 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 12:17:29 +00:00
7519de519e
Revert 8136 "Remove frame copy in ViEExternalRendererImpl::Rende..."
...
> Remove frame copy in ViEExternalRendererImpl::RenderFrame
>
> Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.
>
> BUG=1128
> R=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/36489004
TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8144 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 21:20:41 +00:00
182ea46fac
Remove frame copy in ViEExternalRendererImpl::RenderFrame
...
Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.
BUG=1128
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8136 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 11:50:13 +00:00
cfd82dfc11
Split packets/bytes in StreamDataCounter into RtpPacketCounter struct.
...
Prepares for adding FEC bytes to the StreamDataCounter.
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8122 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 09:39:59 +00:00
e7358eabbc
Only report fraction of lost packets if report_block_stats has been updated.
...
R=holmer@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8108 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 09:00:19 +00:00
cbacd9e3bf
Bump to version 41.
...
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8104 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 18:52:01 +00:00
2624b1ed23
Remove unused private data member engine_id_
...
BUG=chromium:447445
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8088 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 07:54:29 +00:00
0800db74b9
Add percentage of fec packets and recovered media packets to histogram stats:
...
- "WebRTC.Video.ReceivedFecPacketsInPercent"
- "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec"
BUG=crbug/419657
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8072 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 07:40:20 +00:00
86e1e487e7
Move system_wrappers.gyp files to the proper directory.
...
Build targets should not refer to non-subpaths as was happening before when
source/system_wrappers.gyp refers to ../interface/ files.
R=kjellander@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 09:30:52 +00:00
0b0c24177b
Only return Rtx mode in RTXSendStatus().
...
There is no need to return 'ssrc' and 'payloadtype' inside this function
since they are never used.
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38569004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8049 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 14:15:15 +00:00
16825b1a82
Use int64_t more consistently for times, in particular for RTT values.
...
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t. Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org , holmer@google.com , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:51:21 +00:00
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
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
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
823c9b8e36
Add histograms stats for sent/received fraction loss for a stream:
...
- "WebRTC.Video.SentPacketsLostInPercent"
- "WebRTC.Video.ReceivedPacketsLostInPercent"
BUG=crbug/419657
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8020 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 07:50:56 +00:00
215bbbdcdd
Fix for log typo in ViEExternalCodecImpl::RegisterExternalReceiveCodec.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7993 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-05 14:56:02 +00:00
50f7db8a77
Remove unneccessary lock causing a potential deadlock.
...
TBR=pbos@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/28359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7967 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 17:55:20 +00:00
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
742386a136
Enable payload-based padding by default and remove the API.
...
BUG=1812
R=mflodman@webrtc.org , pbos@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7964 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 15:33:17 +00:00
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
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
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
45a272ab22
Change aggregated fraction loss to be calculated from the cumulative loss and extended sequence number diff between the current and the last report block of two get stats calls.
...
Previously it was derived from the fraction loss of the current report (which could be based on a received report block in between two get stats calls).
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7928 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 10:27:57 +00:00
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
d021bbbc9e
Fix vp9 setting in vie loopback test.
...
If vp9 codec was selected then videoCodec.codecSpecific.VP8.numberOfTemporalLayers was being set.
TBR=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/37389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7907 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 00:21:47 +00:00
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
8084f9500f
Change LastProcessedRtt (used in the rtp/rtcp module) to return the average RTT (instead of max RTT) to get a smooth estimate of the nack interval.
...
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7863 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 11:04:13 +00:00
0df371549f
Cast payload type to int in logs.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7861 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 10:30:45 +00:00
97d0489058
Add video send bitrates to histogram stats:
...
- total bitrate ("WebRTC.Video.BitrateSentInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateSentInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateSentInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateSentInKbps")
- retransmitted bitrate ("WebRTC.Video.RetransmittedBitrateInKbps")
Add retransmitted bytes to StreamDataCounters.
Change in UpdateRtpStats to also update counters for retransmitted packet.
BUG=crbug/419657
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7838 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 09:47:53 +00:00
edeea91803
Change all system clock types to int64_t in bitrate_controller.
...
They are both compared to int64_t types inside the class, and is being called
with int64_t types. Could possibly cause bugs.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7832 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 19:46:23 +00:00
273a414b0e
Report encoded frame size in VideoSendStream.
...
Implements reporting transmitted frame size in WebRtcVideoEngine2.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=4033
Review URL: https://webrtc-codereview.appspot.com/33399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 15:23:21 +00:00
83b5200f95
Add framerate for complete received frames to histogram stats:
...
"WebRTC.Video.CompleteFramesReceivedPerSecond".
BUG=crbug/419657
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7762 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-28 10:17:13 +00:00
d952c40c7e
Add receive bitrates to histogram stats:
...
- total bitrate ("WebRTC.Video.BitrateReceivedInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateReceivedInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateReceivedInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateReceivedInKbps")
BUG=crbug/419657
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27189005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7756 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-27 07:38:56 +00:00
8562f23acb
OWNERS: Remove tomasl@ and mallinath@
...
mallinath@ has left the team and tomasl@ says he doesn't
know why he's owner in webrtc/test/channel_transport
R=henrika@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7736 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-24 10:05:05 +00:00
9334ac2d78
Use vector of CSRCs for DeliverFrame & SetCSRCs.
...
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28029004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7734 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-24 08:25:50 +00:00
4591fbd09f
Use size_t more consistently for packet/payload lengths.
...
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
049e4ece30
Change default values for CpuOveruseOptions.
...
Enabled method based on encode time and modified values for the low (60->55) and high threshold (90->85).
Moved DelayedEncoder to fake_encoder.h and added configuration for the delay.
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7722 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 10:19:46 +00:00
ece3890d3a
Report total bitrate for all streams in GetStats.
...
This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.
R=stefan@webrtc.org , xians@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/27179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 11:52:04 +00:00
49ff40e32e
Make SetREMBData accept vector of SSRCs.
...
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7697 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 14:42:37 +00:00
52bb521b47
Update isolate files for Android APK tests.
...
This should speed up test execution on Android since only
the files needed by the test will be processed (instead
of the whole data + resources directories).
A few files for modules_unittests had to be explicitly added
for Android, since they were previously a part of the
add-whole-directories entries for the resources and data
directories.
BUG=webrtc:3741
TEST=Passing android+android_rel trybots.
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7694 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 08:35:05 +00:00
a367aeab82
Bump to version 40
...
TBR=niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/26109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7683 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 16:23:15 +00:00
cd621a8657
Add thread annotations to overuse_frame_detector class.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7678 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 09:40:19 +00:00
6a364fe11b
Remove uses of build date/time.
...
Uses of __DATE__ and __TIME__ are blocking deterministic Chromium
builds. We're not really making use of these, and if anything they're
likely to be misleading as it's impossible to distinguish between a new
revision and a freshly-built old branch.
R=mflodman@webrtc.org , tnakamura@webrtc.org
BUG=3983
Review URL: https://webrtc-codereview.appspot.com/27039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7635 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 17:40:28 +00:00
0bae1fab4a
Wire up bandwidth stats to the new API and webrtcvideoengine2.
...
Adds stats to verify bandwidth and pacer stats.
BUG=1788
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 14:05:29 +00:00
