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Author SHA1 Message Date
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
asapersson@webrtc.org
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
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
marpan@webrtc.org
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
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
asapersson@webrtc.org
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
asapersson@webrtc.org
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
asapersson@webrtc.org
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
stefan@webrtc.org
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
pbos@webrtc.org
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
asapersson@webrtc.org
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
asapersson@webrtc.org
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
kjellander@webrtc.org
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
pbos@webrtc.org
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
pkasting@chromium.org
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
asapersson@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
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
kjellander@webrtc.org
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
tnakamura@webrtc.org
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
asapersson@webrtc.org
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
pbos@webrtc.org
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
stefan@webrtc.org
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
sprang@webrtc.org
dcebf2daa7 Reworked paced sender queue 
Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage.

Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these.

Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long.

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

Review URL: https://webrtc-codereview.appspot.com/27869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 16:27:16 +00:00
stefan@webrtc.org
f37145f685 Enables AIMD control by default. 
BUG=1788
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 09:08:21 +00:00
marpan@webrtc.org
4765ca55f9 Roll chromium_revision: 28d1981..d3db2ff 
Pick up the libvpx roll: https://codereview.chromium.org/674753002

Summary of changes (28d1981..d3db2ff/DEPS):
* third_party/android_tools 36bf7ac..ea50ccc
* third_party/boringssl 7ea8481..751e889
* third_party/icu 8ac906f..d8b2a9d
* third_party/libvpx efe9712..2e5ced5
* third_party/usrsctp/usrsctplib
* tools/gyp 1990:1991
* tools/swarming_client a57d7db..bcb3bc3

Clang is not updated in this roll.

Made the change getchar() --> getc(stdin) as seems like getchar() isn't supported on android anymore.
(getchar() was causing the error: undefined reference to '__srget')

Update rate control parameter in vp9 test.

R=andrew@webrtc.org
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/23229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7598 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 20:10:26 +00:00
asapersson@webrtc.org
96dc685143 Add stats for video: 
- number of sent/received RTCP NACK/FIR/PLI per minute
- percentage of unique sent/received NACK requests
- percentage of discarded/duplicated packets by the jitter buffer
- permille of sent/received key frames

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

Review URL: https://webrtc-codereview.appspot.com/24959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7592 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 14:40:38 +00:00
marpan@webrtc.org
5b88317820 Add VP9 codec to VCM and vie_auto_test. 
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.

This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in:
see https://code.google.com/p/webrtc/issues/detail?id=3932

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

Review URL: https://webrtc-codereview.appspot.com/31829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-01 06:10:48 +00:00
kjellander@webrtc.org
78c222bfae Update all .isolate files for the new format. 
R=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27809004

Patch from Marc-Antoine Ruel <maruel@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 18:08:09 +00:00
asapersson@webrtc.org
7f10513efc Remove unused code in overuse detector. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7557 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 10:05:21 +00:00
asapersson@webrtc.org
580d367b14 Add macros and APIs for webrtc histograms. 
BUG=crbug/419657

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

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

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

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

R=mflodman@webrtc.org

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7505 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 11:57:05 +00:00
henrike@webrtc.org
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
asapersson@webrtc.org
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
stefan@webrtc.org
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
stefan@webrtc.org
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
marpan@webrtc.org
573c78e31c Add VP9 codec to VCM and vie_auto_test. 
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.
Passes trybots.

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

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

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

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

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

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-10 09:42:53 +00:00
minyue@webrtc.org
2c0cdbce22 Estimating NTP time with a given RTT. 
RemoteNtpTimeEstimator needed user to give a remote SSRC and it intended to call RtpRtcp module to obtain RTT, to be able to calculate Ntp time.

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

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

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

BUG=

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 10:52:43 +00:00
kjellander@webrtc.org
f21ea918ad GN: Add common configs to all targets. 
This is needed to ensure we have the same build with GN
as with GYP, since GYP includes the common.gypi on a global level.
Several fixes has been needed in the past because some code have
been built without the right defines.

BUG=3441
R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/28589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-28 17:37:22 +00:00
henrikg@webrtc.org
ca110b808f Mark virtual overrides of ViENetwork and VoENetwork as such. 
This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.

BUG=none
TEST=none
R=henrikg@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7305 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-26 11:09:08 +00:00
tnakamura@webrtc.org
61e811faa0 Bump to version 39 
TBR=niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/24749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7299 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-25 15:28:20 +00:00
pbos@webrtc.org
38344ed280 Move thread_annotations.h to webrtc/base/. 
R=andresp@webrtc.org, mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 06:05:00 +00:00
andresp@webrtc.org
ab071daab8 Split video_render_module implementation into default and internal implementation. 
Targets must now link with implementation of their choice instead of at "gyp"-time.

Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default render implementation:
- video_engine_tests
- video_loopback
- video_replay
- anything dependent on webrtc_test_common

Targets linking with internal render implementation:
- vie_auto_test
- video_render_tests
- libwebrtcdemo-jni
- video_engine_core_unittests

GN changes:
- Not many since there is almost no test definitions.

Work-around for chromium:
- Until chromium has updated libpeerconnection to link with video_capture_impl and video_render_impl, webrtc target automatically depends on it. This should fix the FYI bots and not require a webrtc roll to fix.

Re-enable android tests by reverting 7026 (some tests left disabled).

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3770
R=kjellander@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7217 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 08:58:15 +00:00
andresp@webrtc.org
a74eda1b6f Split video_capture_module specific implementation (external vs internal capture) 
into its own targets. Dependencies must link directly with the desired one.

Targets linking with libjingle_media:
 - internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default/external capture implementation:
 - anything dependent on webrtc_test_common
 - anything dependent on video_engine_core

Targets linking with internal capture implementation:
 - vie_auto_test
 - anything dependent on webrtc_test_renderer

GN changes:
 - Not many since there is almost no test definitions.

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.

BUG=3768
R=glaznev@webrtc.org
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 11:50:19 +00:00
andresp@webrtc.org
85ef770d92 Split video engine android initialization into each internal module initialization. 
This is to later on allow targets to pick at link time if to include the external or internal implementation. In order to do that the video_engine cannot compile different based on which option is picked later on.

BUG=3768,3770
R=glaznev@webrtc.org, stefan@webrtc.org
TBR=henrike@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7208 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 11:44:51 +00:00
stefan@webrtc.org
6071b0636d Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such. 
This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.

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

-- This is was reviewed in https://webrtc-codereview.appspot.com/19309004/, but
-- a new cl was needed to resolve a small conflict before committing.

BUG=none
TEST=none
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 07:42:33 +00:00
henrike@webrtc.org
cc774a69cb Mark all virtual overrides in the hierarchies of RtpDump and 
VCMPacketizationCallback as such.

This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.

This also marks all other such overrides in the affected files.

BUG=none
TEST=none
R=henrike@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7161 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 22:45:54 +00:00
henrik.lundin@webrtc.org
1972ff8a6e Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE. 
This will make a subsequent change I intend to do safer, where I'll change the
return type of one of the base Module functions, by breaking the compile if I
miss any overrides.

This also highlighted a number of unused functions (in many cases apparently
virtual "overrides" of no-longer-existent base functions).  I've removed some of
these.

This also highlighted several cases where "virtual" was used unnecessarily to
mark a function that was only defined in one class.  Removed "virtual" in those
cases.

BUG=none
TEST=none
R=andrew@webrtc.org, henrik.lundin@webrtc.org, mallinath@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7146 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 06:20:28 +00:00