11 Commits

Author SHA1 Message Date
philipel
e21be1db4c Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
Reason for revert:
Fixes has landed.

Original issue's description:
> Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
>
> Reason for revert:
> We are not certain this is the behavior we want.
>
> Original issue's description:
> > Fix the video buffer size should take rtt into consideration
> >
> > BUG=webrtc:8010
> >
> > Review-Url: https://codereview.webrtc.org/2980413002
> > Cr-Commit-Position: refs/heads/master@{#19285}
> > Committed: f1e08d0b58
>
> TBR=sprang@webrtc.org,gustavogb@gmail.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:8010
>
> Review-Url: https://codereview.webrtc.org/3002033002
> Cr-Commit-Position: refs/heads/master@{#19442}
> Committed: bdbc8895f3

TBR=sprang@webrtc.org,gustavogb@gmail.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8010

Review-Url: https://codereview.webrtc.org/3016633002
Cr-Commit-Position: refs/heads/master@{#19944}
2017-09-25 13:37:12 +00:00
Niels Möller
b0573bca16 Reorganize config of RTP header extensions for video receive streams.
Bug: webrtc:6847
Change-Id: Iae2386e55520601883379fc7802a5c5246be935e
Reviewed-on: https://webrtc-review.googlesource.com/2001
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19943}
2017-09-25 11:51:20 +00:00
brandtr
2c30120fac Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ )
Reason for revert:
Breaks KitKat/Lollipop perf bots.

Original issue's description:
> Add full stack tests for MediaCodec encoder.
>
> * Add audio_ prefix to CallTest::{en,de}coder_factory_.
> * Let VideoQualityTest only instantiate encoders using encoder factories.
> * Add HW encoder factories to VideoQualityTest.
> * Add full stack tests:
>   - sqcif7 at  30 kbps: MediaCodec and libvpx.
>   - 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx.
>
> BUG=webrtc:8219
>
> Review-Url: https://codereview.webrtc.org/3005253002
> Cr-Commit-Position: refs/heads/master@{#19923}
> Committed: https://webrtc.googlesource.com/src/+/2cefac6c1685abfcd7b90fdef8e926f1c2b79bfa

TBR=sprang@webrtc.org,asapersson@webrtc.org,kjellander@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:8219

Review-Url: https://codereview.webrtc.org/3016593002
Cr-Commit-Position: refs/heads/master@{#19926}
2017-09-22 11:30:08 +00:00
brandtr
2cefac6c16 Add full stack tests for MediaCodec encoder.
* Add audio_ prefix to CallTest::{en,de}coder_factory_.
* Let VideoQualityTest only instantiate encoders using encoder factories.
* Add HW encoder factories to VideoQualityTest.
* Add full stack tests:
  - sqcif7 at  30 kbps: MediaCodec and libvpx.
  - 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx.

BUG=webrtc:8219

Review-Url: https://codereview.webrtc.org/3005253002
Cr-Commit-Position: refs/heads/master@{#19923}
2017-09-22 07:46:25 +00:00
brandtr
7cd28b9172 Set protected_by_flexfec flag properly in tests.
BUG=none

Review-Url: https://codereview.webrtc.org/3010003002
Cr-Commit-Position: refs/heads/master@{#19921}
2017-09-22 07:26:25 +00:00
Karl Wiberg
73b60b82ee Remove the redundant method GetPayloadSpecifics
It's in the way of a refactoring.

Also change PayloadTypeToPayload---the method all callers can use instead---to return Optional<Payload> instead of const Payload* (for thread safety reasons: an object that protects itself with a mutex shouldn't be handing out pointers to parts of itself). 

BUG=webrtc:8159

Change-Id: I7ef0d545077ffdea016b309f2165e3c4955a2928
Reviewed-on: https://webrtc-review.googlesource.com/2360
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19917}
2017-09-21 20:19:55 +00:00
asapersson
8d75ac7e3f Add stats for forced software encoder fallback for VP8.
Stats added for number of forced SW fallback changes per minute and percentage of time fallback is enabled for sent video streams:

- "WebRTC.Video.Encoder.ForcedSwFallbackChangesPerMinute.Vp8"
- "WebRTC.Video.Encoder.ForcedSwFallbackTimeInPercent.Vp8"

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/3012863002
Cr-Commit-Position: refs/heads/master@{#19862}
2017-09-15 13:41:15 +00:00
Mirko Bonadei
7120742701 Adding NOLINT for typedefs.h and common_types.h
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.

The cpplint complaint is:
Include the directory when naming .h files  [build/include] [4]

This CL disables the error but we have to remove these two headers
from the root directory.

NOPRESUBMIT=true

Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
2017-09-15 13:03:51 +00:00
oprypin
743117f304 Disable FullStackTest.LargeRoomVP8_*thumb on iOS
It fails for the same reason as on Android

BUG=webrtc:7301

Review-Url: https://codereview.webrtc.org/3017473002
Cr-Commit-Position: refs/heads/master@{#19857}
2017-09-15 12:24:24 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00