Commit Graph

352 Commits

Author SHA1 Message Date
d5ce2e63df Remove EventWrapper::Reset().
This simplifies the event wrapper which we've recently found issues in.
Also refactoring EndToEndTest.RespectsNetworkState to not depend on it.

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

Review URL: https://webrtc-codereview.appspot.com/41939004

Cr-Commit-Position: refs/heads/master@{#8366}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8366 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-13 14:58:38 +00:00
2b69eab077 Restructure GYP for vp9, opus and direct trace
This is needed to make the build more flexible for some use cases.

BUG=4185
R=andresp@webrtc.org, stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34099004

Cr-Commit-Position: refs/heads/master@{#8290}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8290 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 10:01:40 +00:00
f31f56d8d4 Remove default arguments in EncodedImageCallback.
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39719004

Cr-Commit-Position: refs/heads/master@{#8289}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8289 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 09:14:48 +00:00
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
7d2b6a9346 Enable Clang warning implicit-fallthrough and annotate the code.
BUG=4242
R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34899004

Cr-Commit-Position: refs/heads/master@{#8187}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8187 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 18:38:13 +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
9b64a6edd7 Adjust parameter in videoprocessor_integrationtest for VP9.
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/35919004

Cr-Commit-Position: refs/heads/master@{#8178}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8178 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 23:59:16 +00:00
dc8a9da386 Adjust qp-max settinhg in VP9 wrapper.
More closely matches the qp-max setting used in VP8.

TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/39709004

Cr-Commit-Position: refs/heads/master@{#8177}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8177 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 23:08:39 +00:00
d7e34e1086 Make it easier to use external libyuv + cleanup GYP files.
It is now easier to use an external libyuv library.
Fix some GYP errors.
Remove the temporary webrtc_base target (depends on
https://codereview.chromium.org/865603002/ being landed
first).

BUG=4185
R=andresp@webrtc.org, andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8154 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 19:17:26 +00:00
38d11b8529 Enable encoder multi-threading for VP9.
R=stefan@webrtc.org
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/41489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8150 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 15:21:36 +00:00
f18fba2f7b Implement SimulcastEncoderAdapter support.
R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/37589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8061 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 16:26:23 +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
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
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
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
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
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
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
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
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
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
f4c19480fc Remove jitter_estimate_test.h
BUG=2156
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7866 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 21:08:39 +00:00
f244760827 Add histograms for receive statistics:
- decoded frames per second ("WebRTC.Video.DecodedFramesPerSecond")
- percentage of delayed frames to rendered ("WebRTC.Video.DelayedFramesToRenderer")
- average delay (of delayed frames) to renderer ("WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs")

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

Review URL: https://webrtc-codereview.appspot.com/26319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7847 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 14:13:26 +00:00
9115cde6c9 Merge VP8 changes.
R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/35389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7841 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:36:40 +00:00
86b6d65ef1 Remove no longer used video codec test framework.
Moves one test to the vp8 unittests which might still be good to have.
Also does a bit of clean up in vp8 unittests.

R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7835 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 00:02:45 +00:00
fb01376eca Adjust some parameters for VP9 tests.
Needed for the next/upcoming libvpx roll.

BUG=

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7821 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 06:25:51 +00:00
001f3b9818 Adjust parameter in videoprocessor_integration_test for vp9.
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/33459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7791 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 02:00:12 +00:00
ceca014b8b Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeBitRateVP9.
BUG=4059

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7789 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 01:05:43 +00:00
84515f841d Roll chromium_revision 309cf65..24b4c73
Two VP9 tests needed to be disabled (see webrtc:4059) to make all tests pass.

Relevant changes:
* src/third_party/android_tools: ea50ccc..4c47ef6
* src/third_party/icu: dd72764..866ff69
* src/third_party/libvpx: 2e5ced5..429874c
* src/third_party/nss: 258342e..bb4e75a
* src/third_party/yasm/source/patched-yasm: c960eb1..4671120
* src/tools/gyp: 0a381c0..fe00999
* src/tools/swarming_client: 5b827c9..1d4965c
Details: 309cf65..24b4c73/DEPS

Clang version was not updated in this roll.

BUG=4059
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7778 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02 08:48:08 +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
91d928e737 Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader
This is in preparation for creating a new class RtpFileWriter which
will use the same RtpPacket struct.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7749 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 15:50:30 +00:00
4f16c874c6 Simplifying VideoReceiver and JitterBuffer.
Removing frame_buffers_ array and dual-receiver mechanism. Also adding
some thread annotations to VCMJitterBuffer.

R=stefan@webrtc.org
BUG=4014

Review URL: https://webrtc-codereview.appspot.com/27239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7735 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-24 09:06:48 +00:00
ad0e71c9a3 Update mock_frame_dropper.h to use size_t
This mock was missed in the work of
https://webrtc-codereview.appspot.com/23129004 since the file
is not currently used by any test in this repo.

BUG=chromium:81439
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7727 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-21 09:40:57 +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
e1745cbb7c Adjust parameter in vp9 rate control test.
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7638 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 02:55:53 +00:00
5f1e2e42a8 Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test.
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7637 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 02:02:28 +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
e451b756a8 Update rate control parameter in vp9 test.
TBR=hellner@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7599 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 21:26:08 +00:00
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
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
ed45896759 Adjust/increase rate control thresold for a vp9 test.
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7589 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 07:08:52 +00:00
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
776e6f289c Use external VideoDecoders in VideoReceiveStream.
Removes direct VideoCodec use from the new API, exposes VideoDecoders
through webrtc/video_decoder.h similar to VideoEncoders.

Also includes some preparation for wiring up external decoders in
WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they
were allocated internally or externally.

Additionally addresses a data race in VideoReceiver that was exposed with this change.

R=mflodman@webrtc.org, stefan@webrtc.org
TBR=pthatcher@webrtc.org
BUG=2854,1667

Review URL: https://webrtc-codereview.appspot.com/27829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 15:28:39 +00:00
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
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
3f8f5554a0 Disable TestVp8Impl.BaseUnitTest on MSan.
MemorySanitizer uses generic (non-optimized) libvpx which is not bit
exact. This may be a bug in upstream libvpx, but we're forced to disable
it now as it blocks launching the MSan bot.

R=stefan@webrtc.org
TBR=marpan@webrtc.org
BUG=3904

Review URL: https://webrtc-codereview.appspot.com/24089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7491 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 10:30:30 +00:00
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
c87b74717b Adjust/increase rate control thresold for a vp9 test.
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7423 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 17:55:57 +00:00
573c78e31c Add VP9 codec to VCM and vie_auto_test.
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