1002 Commits

Author SHA1 Message Date
nisse
284542b882 Make OverheadObserver::OnOverheadChanged count RTP headers only
This lets the RTP code be unaware of lower layers, and the
SetTransportOverhead method is deleted from RTPSender and RtpRtcp.

Instead, that method is added to CongestionController and
TransportFeedbackAdapter, where it is more appropriate.

BUG=wertc:6847

Review-Url: https://codereview.webrtc.org/2589743002
Cr-Commit-Position: refs/heads/master@{#15995}
2017-01-10 16:58:32 +00:00
nisse
af916899cc Move VideoFrame and related declarations to webrtc/api/video.
Moves webrtc/common_video/rotation.h and parts of
webrtc/common_video/include/video_frame_buffer.h and
webrtc/video_frame.h, and adds to a new GN target api:video_frame_api.

BUG=webrtc:5880

Review-Url: https://codereview.webrtc.org/2517173004
Cr-Commit-Position: refs/heads/master@{#15993}
2017-01-10 15:44:26 +00:00
philipel
8054c7ecf8 EndToEndTest.ReceivesAndRetransmitsNack now handles reordering.
Since packets can be received out of order NACKs could be triggered before any
packet was actually dropped. This cause the test to never drop packets which in
turn caused the block of code which set the |observation_complete_| event to
never execute.

BUG=webrtc:2845

Review-Url: https://codereview.webrtc.org/2613443002
Cr-Commit-Position: refs/heads/master@{#15990}
2017-01-10 13:19:19 +00:00
johan
0d1b2b6880 Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
Add RTC_DEPRACATed anonymous unions to not break downstream projects.

Orignal issue's description:
> commit 0ad21111fcc57a7e978edba3c4263f0062d7f9ff
> Author: danilchap <danilchap@webrtc.org>
> Date:   Mon Dec 19 09:36:33 2016 -0800
>
>     Revert of Rename RTPVideoHeader.isFirstPacket to
>     .is_first_packet_in_frame. (patchset #1 id:1 of
>     https://codereview.webrtc.org/2574943003/ )
>
>     Reason for revert:
>     breaks downstream project.
>
>     Can you make this change in a compatible way using anonymous
>     union:
>     union {
>       bool is_first_packet_in_frame;
>       RTC_DEPRECATED bool isFirstPacket;
>     };
>     (unfortunetly this this treak breaks braced initialization in
>     rtp_rtcp_impl_unittest.cc,
>     so that should be rewritting in a more classic way)
>
>     Original issue's description:
>     > Rename RTPVideoHeader.isFirstPacket to
>     > .is_first_packet_in_frame.
>     >
>     > Name should represent the actual meaning.
>     >
>     > BUG=None
>     >
>     > Review-Url: https://codereview.webrtc.org/2574943003
>     > Cr-Commit-Position: refs/heads/master@{#15684}
>     > Committed:
>     > efde908380
>
>     TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org
>     # Skipping CQ checks because original CL landed less than 1 days
>     ago.
>     NOPRESUBMIT=true
>     NOTREECHECKS=true
>     NOTRY=true
>     BUG=None
>
>     Review-Url: https://codereview.webrtc.org/2589783003
>     Cr-Commit-Position: refs/heads/master@{#15686}
>

BUG=None

Review-Url: https://codereview.webrtc.org/2614503002
Cr-Commit-Position: refs/heads/master@{#15987}
2017-01-10 12:21:35 +00:00
palmkvist
c7e7e079a4 Lower bad call logging severity
BUG=webrtc:6814

Review-Url: https://codereview.webrtc.org/2584723003
Cr-Commit-Position: refs/heads/master@{#15964}
2017-01-09 15:23:33 +00:00
ehmaldonado
021eef32f4 Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ )
Reason for revert:
Try again.

Original issue's description:
> Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ )
>
> Reason for revert:
> Probably cause unexpected performance regression
> BUG=chromium:678569
>
> Original issue's description:
> > Refactor webrtc_perf_tests into several source_sets.
> >
> > BUG=webrtc:6954
> >
> > Review-Url: https://codereview.webrtc.org/2609403002
> > Cr-Commit-Position: refs/heads/master@{#15902}
> > Committed: 0b5a26a576
>
> TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6954
>
> Review-Url: https://codereview.webrtc.org/2613913002
> Cr-Commit-Position: refs/heads/master@{#15916}
> Committed: 5fbcd228f0

TBR=kjellander@webrtc.org,danilchap@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:678569

Review-Url: https://codereview.webrtc.org/2615873002
Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 15:09:50 +00:00
danilchap
5fbcd228f0 Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ )
Reason for revert:
Probably cause unexpected performance regression
BUG=chromium:678569

Original issue's description:
> Refactor webrtc_perf_tests into several source_sets.
>
> BUG=webrtc:6954
>
> Review-Url: https://codereview.webrtc.org/2609403002
> Cr-Commit-Position: refs/heads/master@{#15902}
> Committed: 0b5a26a576

TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6954

Review-Url: https://codereview.webrtc.org/2613913002
Cr-Commit-Position: refs/heads/master@{#15916}
2017-01-05 12:57:49 +00:00
ehmaldonado
0b5a26a576 Refactor webrtc_perf_tests into several source_sets.
BUG=webrtc:6954

Review-Url: https://codereview.webrtc.org/2609403002
Cr-Commit-Position: refs/heads/master@{#15902}
2017-01-04 17:13:58 +00:00
asapersson
6966bd51fc ReceiveStatisticsProxy:
- add some unit tests
- change SampleCounter members from int to int64_t

BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2577073002
Cr-Commit-Position: refs/heads/master@{#15882}
2017-01-03 08:44:06 +00:00
brandtr
20d4547e5d Make VideoSendStreamTest.DoesNotUtilizeUlpfecForH264WithNackEnabled useful again.
This test was inadvertently made useless in https://codereview.webrtc.org/2558303002/.
Prior to that CL, at least 100 media packets -- but no ULPFEC packets -- had to
be sent before the test claimed that ULPFEC was not utilized. This CL reinstates
that behaviour, but instead of counting packets we expect the test to time out
when ULPFEC should not be utilized.

This solution is simple in code, but increases the test duration from 70 ms
to 10000 ms. Note, however, that the 70 ms duration was for a test that didn't
test anything interesting...

TESTED=Using gtest-parallel locally: 2000 test runs without flakes.
BUG=webrtc:6851

Review-Url: https://codereview.webrtc.org/2592693002
Cr-Commit-Position: refs/heads/master@{#15859}
2017-01-02 08:34:27 +00:00
asapersson
fe50b4d750 Make class of static functions in rtp_to_ntp.h:
- UpdateRtcpList
- RtpToNtp

class RtpToNtpEstimator
- UpdateMeasurements
- Estimate

List with rtcp measurements is now private.

BUG=none

Review-Url: https://codereview.webrtc.org/2574133003
Cr-Commit-Position: refs/heads/master@{#15762}
2016-12-22 15:53:51 +00:00
danilchap
bf5f5297c5 Disable flaky VideoSendStreamTest.RemoveOverheadeFromBandwidth
BUG=webrtc:6886
NOTRY=True
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2596223002
Cr-Commit-Position: refs/heads/master@{#15761}
2016-12-22 15:51:54 +00:00
brandtr
b29e652b10 Revert "Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ )"
Problem fixed: RTP header extensions were not properly set in tests.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2593963003
Cr-Commit-Position: refs/heads/master@{#15741}
2016-12-21 14:37:18 +00:00
brandtr
504b95eff8 Avoid creating receiver_time outliers in the VideoAnalyzer.
Prior to this change, the receiver_time metric had huge outliers
whenever FlexFEC was enabled. This was due to a measurement problem,
where the time of the incoming packet was incorrectly set to zero.
This happened for packets that were lost in transit, but recovered
through FEC.

This CL fixes this problem by simply not recording samples where the
incoming packet time is undefined. The CL also removes the possibility
of timestamp collisions in the data structures.

TESTED=Ran './webrtc_perf_tests --gtest_filter="*ForemanCifPlr5H264Flexfec*" | grep receiver_time' locally 10 times, without experiencing any outliers.
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2596793002
Cr-Commit-Position: refs/heads/master@{#15735}
2016-12-21 10:54:35 +00:00
stefan
64427e563e Add back video_replay. Disappeared in the gn conversion.
BUG=webrtc:6323

Review-Url: https://codereview.webrtc.org/2595533002
Cr-Commit-Position: refs/heads/master@{#15715}
2016-12-20 15:26:58 +00:00
philipel
022b54e86a Wire up H264 fmtp sprop-parameter-sets with H264SpsPpsTracker.
To avoid making this CL large unittests will be added in a followup CL.

BUG=webrtc:5948

patch from issue 2570073002 at patchset 20001 (http://crrev.com/2570073002#ps20001)

Review-Url: https://codereview.webrtc.org/2565173009
Cr-Commit-Position: refs/heads/master@{#15710}
2016-12-20 12:15:59 +00:00
nisse
5206667dad Delete unused method PayloadRouter::MaxPayloadLength.
Documentation was also unclear, it seems it returned the RTP packet
size including RTP headers.

BUG=None.

Review-Url: https://codereview.webrtc.org/2588343002
Cr-Commit-Position: refs/heads/master@{#15707}
2016-12-20 11:12:04 +00:00
danilchap
0ad21111fc Revert of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. (patchset #1 id:1 of https://codereview.webrtc.org/2574943003/ )
Reason for revert:
breaks downstream project.

Can you make this change in a compatible way using anonymous union:
union {
  bool is_first_packet_in_frame;
  RTC_DEPRECATED bool isFirstPacket;
};
(unfortunetly this this treak breaks braced initialization in rtp_rtcp_impl_unittest.cc,
so that should be rewritting in a more classic way)

Original issue's description:
> Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
>
> Name should represent the actual meaning.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2574943003
> Cr-Commit-Position: refs/heads/master@{#15684}
> Committed: efde908380

TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2589783003
Cr-Commit-Position: refs/heads/master@{#15686}
2016-12-19 17:36:33 +00:00
johan
efde908380 Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
Name should represent the actual meaning.

BUG=None

Review-Url: https://codereview.webrtc.org/2574943003
Cr-Commit-Position: refs/heads/master@{#15684}
2016-12-19 16:32:24 +00:00
asapersson
66d4b37414 Move histogram for number of pause events to per stream:
"WebRTC.Call.NumberOfPauseEvents" -> "WebRTC.Video.NumberOfPauseEvents"

Recorded if a certain time has passed (10 sec) since the first media packet was sent.

Moved to per stream to know when media has started and to prevent logging stats for calls that was never in use.

Add histogram for percentage of paused video time for sent video streams:
"WebRTC.Video.PausedTimeInPercent"

BUG=b/32659204

Review-Url: https://codereview.webrtc.org/2530393003
Cr-Commit-Position: refs/heads/master@{#15681}
2016-12-19 14:50:53 +00:00
kthelgason
0cd27ba088 Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ )
Reason for revert:
Bug affecting perf tests has been fixed. The issue was that I had accidentally disabled cpu overuse adaptation based on the encoders ScalingSettings, not just quality-based scaling.

Original issue's description:
> Revert of Properly report number of quality downscales in stats. (patchset #11 id:220001 of https://codereview.webrtc.org/2564373002/ )
>
> Reason for revert:
> Breaks perf tests
>
> Original issue's description:
> > Properly report number of quality downscales in stats.
> >
> > A regression was introduced in 876222f that caused these stats to
> > be reported incorrectly. This used to be only implemented for VP8
> > but should now be available for all codecs.
> >
> > BUG=webrtc:6860
> >
> > Review-Url: https://codereview.webrtc.org/2564373002
> > Cr-Commit-Position: refs/heads/master@{#15673}
> > Committed: 0c8c538835
>
> TBR=asapersson@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:6860
>
> Review-Url: https://codereview.webrtc.org/2586783003
> Cr-Commit-Position: refs/heads/master@{#15678}
> Committed: fe04bd43cc

TBR=asapersson@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:6860

Review-Url: https://codereview.webrtc.org/2588743002
Cr-Commit-Position: refs/heads/master@{#15680}
2016-12-19 14:32:16 +00:00
kthelgason
fe04bd43cc Revert of Properly report number of quality downscales in stats. (patchset #11 id:220001 of https://codereview.webrtc.org/2564373002/ )
Reason for revert:
Breaks perf tests

Original issue's description:
> Properly report number of quality downscales in stats.
>
> A regression was introduced in 876222f that caused these stats to
> be reported incorrectly. This used to be only implemented for VP8
> but should now be available for all codecs.
>
> BUG=webrtc:6860
>
> Review-Url: https://codereview.webrtc.org/2564373002
> Cr-Commit-Position: refs/heads/master@{#15673}
> Committed: 0c8c538835

TBR=asapersson@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:6860

Review-Url: https://codereview.webrtc.org/2586783003
Cr-Commit-Position: refs/heads/master@{#15678}
2016-12-19 14:17:30 +00:00
brandtr
696c9c6b64 Add multithreaded fake encoder and corresponding FlexFEC VideoSendStreamTest.
This test would have found the issue that was fixed in
https://codereview.webrtc.org/2562983002/.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2573453002
Cr-Commit-Position: refs/heads/master@{#15675}
2016-12-19 13:47:28 +00:00
kthelgason
0c8c538835 Properly report number of quality downscales in stats.
A regression was introduced in 876222f that caused these stats to
be reported incorrectly. This used to be only implemented for VP8
but should now be available for all codecs.

BUG=webrtc:6860

Review-Url: https://codereview.webrtc.org/2564373002
Cr-Commit-Position: refs/heads/master@{#15673}
2016-12-19 13:04:40 +00:00
philipel
20d05a9f71 Now expect the correct number of streams in EndToEndTest.GetStats.
The rtx streams were not included in the number of expected streams
but the test passed most of the time anyway due to how the checking was done.
Flake was caused when the number of registered streams jumped passed the
number of expected send streams excluding the number of rtx streams.

BUG=webrtc:6879

Review-Url: https://codereview.webrtc.org/2580343002
Cr-Commit-Position: refs/heads/master@{#15671}
2016-12-19 12:17:27 +00:00
asapersson
fb6ad3b196 Add full stack tests:
- ForemanCif30kbpsWithoutPacketLoss
- ForemanCif30kbpsWithoutPacketLossH264

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2585703002
Cr-Commit-Position: refs/heads/master@{#15658}
2016-12-16 14:54:01 +00:00
kthelgason
7e70fe2b96 Fix wrong log message.
When scaling up or down the reason for scaling was flipped
in the logs.

BUG=None
R=perkj@webrtc.org
NOTRY=true

Review-Url: https://codereview.webrtc.org/2563683002
Cr-Commit-Position: refs/heads/master@{#15652}
2016-12-16 11:46:48 +00:00
mflodman
8d66245ccc Adding Åsa and Erik as video owners.
Also, removing pbos from the owner list.

BUG=none
R=asapersson@webrtc.org, sprang@webrtc.org

Review-Url: https://codereview.webrtc.org/2583793002 .
Cr-Commit-Position: refs/heads/master@{#15648}
2016-12-16 10:12:31 +00:00
mflodman
d79f97b542 Fixing loopback video test by reconfiguring the encoder to correct size.
Same as https://codereview.webrtc.org/2480753002, but with a small fix.

BUG=none

Review-Url: https://codereview.webrtc.org/2578143002
Cr-Commit-Position: refs/heads/master@{#15639}
2016-12-15 15:24:38 +00:00
philipel
721d402d71 Create VideoReceiver with external VCMTiming object.
In order for the VCMTiming object to be correctly updated with decoding timings
when running the WebRTC-NewVideoJitterBuffer experiment the VCMTiming object
has to be available in both the VideoReceiver and the video_coding::FrameBuffer
class. Therefore the VCMTiming object is created in VideoRecieveStream and
then passed to VideoReceiver/video_coding::FrameBuffer as they are constructed.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2575473004
Cr-Commit-Position: refs/heads/master@{#15638}
2016-12-15 15:11:01 +00:00
nisse
df2ceb88a8 Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
Reason for revert:
Fixing perf tests.

Original issue's description:
> Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
>
> Reason for revert:
> Crashes perf tests, e.g.,
>
> ./out/Debug/webrtc_perf_tests --gtest_filter='FullStackTest.ScreenshareSlidesVP8_2TL_VeryLossyNet'
>
> dies with an assert related to rtc::Optional.
>
> Original issue's description:
> > Delete VideoFrame default constructor, and IsZeroSize method.
> >
> > This ensures that the video_frame_buffer method never can return a
> > null pointer.
> >
> > BUG=webrtc:6591
> >
> > Committed: https://crrev.com/bfcf561923a42005e4c7d66d8e72e5932155f997
> > Cr-Commit-Position: refs/heads/master@{#15574}
>
> TBR=magjed@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:6591
>
> Committed: https://crrev.com/0989fbcad2ca4eb5805a77e8ebfefd3af06ade23
> Cr-Commit-Position: refs/heads/master@{#15597}

TBR=magjed@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:6591

Review-Url: https://codereview.webrtc.org/2574183002
Cr-Commit-Position: refs/heads/master@{#15633}
2016-12-15 14:30:00 +00:00
nisse
0989fbcad2 Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
Reason for revert:
Crashes perf tests, e.g.,

./out/Debug/webrtc_perf_tests --gtest_filter='FullStackTest.ScreenshareSlidesVP8_2TL_VeryLossyNet'

dies with an assert related to rtc::Optional.

Original issue's description:
> Delete VideoFrame default constructor, and IsZeroSize method.
>
> This ensures that the video_frame_buffer method never can return a
> null pointer.
>
> BUG=webrtc:6591
>
> Committed: https://crrev.com/bfcf561923a42005e4c7d66d8e72e5932155f997
> Cr-Commit-Position: refs/heads/master@{#15574}

TBR=magjed@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:6591

Review-Url: https://codereview.webrtc.org/2574123002
Cr-Commit-Position: refs/heads/master@{#15597}
2016-12-14 10:06:49 +00:00
nisse
bfcf561923 Delete VideoFrame default constructor, and IsZeroSize method.
This ensures that the video_frame_buffer method never can return a
null pointer.

BUG=webrtc:6591

Review-Url: https://codereview.webrtc.org/2541863002
Cr-Commit-Position: refs/heads/master@{#15574}
2016-12-13 14:08:39 +00:00
palmkvist
349092befe Logging basic bad call detection
BUG=webrtc:6814

Review-Url: https://codereview.webrtc.org/2474913002
Cr-Commit-Position: refs/heads/master@{#15568}
2016-12-13 10:46:06 +00:00
johan
07e276c8d5 Rename RtpStreamReceiver::SetCodec() to ::AddCodec().
AddCodec represents better what this function actually does.

BUG=None

Review-Url: https://codereview.webrtc.org/2573593003
Cr-Commit-Position: refs/heads/master@{#15565}
2016-12-13 10:23:43 +00:00
brandtr
65a1e77202 Try to deflake VideoSendStream tests with ULPFEC.
The changes here are the same as in https://codereview.webrtc.org/2523993002/:
- reduce packet loss from 50% to 5%, to allow the BWE to ramp up better.
- Do not wait for 100 packets to be sent before letting the test pass.

BUG=webrtc:6851

Review-Url: https://codereview.webrtc.org/2558303002
Cr-Commit-Position: refs/heads/master@{#15542}
2016-12-12 09:55:09 +00:00
brandtr
3536463e7e Only store sequence numbers for media stream in FlexFEC end-to-end test.
This should remove the test flakiness, as before this change there
could be collisions from sequence numbers coming from two sequence
number spaces (the media SSRC and the FlexFEC SSRC). The probability
of collisions was low, and hence the flakes were far between.

This change also reduces the packet loss to 5% (down from ~50%), in
order for the BWE to have an easier time to ramp up.

BUG=webrtc:6825
R=philipel@webrtc.org, mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/2554403003
Cr-Commit-Position: refs/heads/master@{#15512}
2016-12-09 14:51:43 +00:00
kthelgason
33ce88926a Reland of Bump up scaling limit for MediaCodec. (patchset #1 id:1 of https://codereview.webrtc.org/2562963002/ )
Reason for revert:
Fixed perf tests.

Original issue's description:
> Revert of Bump up scaling limit for MediaCodec. (patchset #3 id:40001 of https://codereview.webrtc.org/2566533002/ )
>
> Reason for revert:
> Failed on the perf tests.
>
> Original issue's description:
> > Bump up scaling limit for MediaCodec.
> >
> > Wait until MediaCodec is better tested at these low
> > resolutions, and until some fallback mechanism is in place
> > before lowering this threshold.
> >
> > BUG=webrtc:6837
> >
> > Committed: https://crrev.com/3e9b1330467edf6b5af609b375c15efb9e6b4933
> > Cr-Commit-Position: refs/heads/master@{#15498}
>
> TBR=magjed@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6837
>
> Committed: https://crrev.com/1cd0a0ab43846072b1e2f37c953ecd770feb5963
> Cr-Commit-Position: refs/heads/master@{#15500}

TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6837

Review-Url: https://codereview.webrtc.org/2567613002
Cr-Commit-Position: refs/heads/master@{#15505}
2016-12-09 11:54:08 +00:00
asapersson
a90799d5fb Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
Reason for revert:
Increase in encode time larger than expected.

Original issue's description:
> Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
>
> Error resilience is currently always enabled for VP9 which reduces quality.
>
> BUG=webrtc:6783
>
> Committed: https://crrev.com/4eb03c76fa2320534d669fda2aabf800e7a6f579
> Cr-Commit-Position: refs/heads/master@{#15390}

TBR=stefan@webrtc.org,marpan@webrtc.org,mflodman@webrtc.org,marpan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6783

Review-Url: https://codereview.webrtc.org/2554403006
Cr-Commit-Position: refs/heads/master@{#15501}
2016-12-09 10:35:30 +00:00
kthelgason
1cd0a0ab43 Revert of Bump up scaling limit for MediaCodec. (patchset #3 id:40001 of https://codereview.webrtc.org/2566533002/ )
Reason for revert:
Failed on the perf tests.

Original issue's description:
> Bump up scaling limit for MediaCodec.
>
> Wait until MediaCodec is better tested at these low
> resolutions, and until some fallback mechanism is in place
> before lowering this threshold.
>
> BUG=webrtc:6837
>
> Committed: https://crrev.com/3e9b1330467edf6b5af609b375c15efb9e6b4933
> Cr-Commit-Position: refs/heads/master@{#15498}

TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6837

Review-Url: https://codereview.webrtc.org/2562963002
Cr-Commit-Position: refs/heads/master@{#15500}
2016-12-09 10:30:50 +00:00
kthelgason
3e9b133046 Bump up scaling limit for MediaCodec.
Wait until MediaCodec is better tested at these low
resolutions, and until some fallback mechanism is in place
before lowering this threshold.

BUG=webrtc:6837

Review-Url: https://codereview.webrtc.org/2566533002
Cr-Commit-Position: refs/heads/master@{#15498}
2016-12-09 09:46:46 +00:00
brandtr
f7c6d7231c Rename RtpStreamReceiver::IsFecEnabled to RtpStreamReceiver::IsUlpfecEnabled.
BUG=webrtc:5654
R=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2548523002
Cr-Commit-Position: refs/heads/master@{#15487}
2016-12-08 16:25:51 +00:00
nisse
10daf861b9 Simplify an always true condition.
Also deletes one call to CongestionController::pacer.

BUG=None

Review-Url: https://codereview.webrtc.org/2542113003
Cr-Commit-Position: refs/heads/master@{#15479}
2016-12-08 14:24:35 +00:00
brandtr
1cfbd6003b Generalize FlexfecReceiveStream::Config.
- Adding information about RTCP and RTP header extensions.
- Renaming flexfec_payload_type -> payload_type and
  flexfec_ssrc -> remote_ssrc.

BUG=webrtc:5654
R=stefan@webrtc.org, philipel@webrtc.org

Review-Url: https://codereview.webrtc.org/2542413002
Cr-Commit-Position: refs/heads/master@{#15477}
2016-12-08 12:18:05 +00:00
ossu
f515ab8c3f Moved call.h and most of api/call/* into call/
BUG=webrtc:6716

Review-Url: https://codereview.webrtc.org/2550273003
Cr-Commit-Position: refs/heads/master@{#15460}
2016-12-07 12:53:04 +00:00
hta
9aa96889a3 Reland of H.264 packetization mode 0 (try 3) (patchset #1 id:1 of https://codereview.webrtc.org/2558453002/ )
Reason for revert:
Fixed timeouts in slow tests

Original issue's description:
> Revert of H.264 packetization mode 0 (try 3) (patchset #13 id:490001 of https://codereview.webrtc.org/2528343002/ )
>
> Reason for revert:
> Failures on the Linux Memcheck bot
>
> Original issue's description:
> > This approach passes packetization mode to the encoder as part of
> > a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.
> >
> > BUG=600254
> >
> > Committed: https://crrev.com/e59647b991f61cf1cf61b020356705e6c0f81257
> > Cr-Commit-Position: refs/heads/master@{#15437}
>
> TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=600254
>
> Committed: https://crrev.com/243a0a7a7fd6b5da1e32df31f1bfbb6a68dc09f3
> Cr-Commit-Position: refs/heads/master@{#15441}

TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=600254

Review-Url: https://codereview.webrtc.org/2558463002
Cr-Commit-Position: refs/heads/master@{#15445}
2016-12-06 13:36:13 +00:00
hta
243a0a7a7f Revert of H.264 packetization mode 0 (try 3) (patchset #13 id:490001 of https://codereview.webrtc.org/2528343002/ )
Reason for revert:
Failures on the Linux Memcheck bot

Original issue's description:
> This approach passes packetization mode to the encoder as part of
> a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.
>
> BUG=600254
>
> Committed: https://crrev.com/e59647b991f61cf1cf61b020356705e6c0f81257
> Cr-Commit-Position: refs/heads/master@{#15437}

TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=600254

Review-Url: https://codereview.webrtc.org/2558453002
Cr-Commit-Position: refs/heads/master@{#15441}
2016-12-06 12:22:05 +00:00
hta
e59647b991 This approach passes packetization mode to the encoder as part of
a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.

BUG=600254

Review-Url: https://codereview.webrtc.org/2528343002
Cr-Commit-Position: refs/heads/master@{#15437}
2016-12-06 10:22:54 +00:00
brandtr
d654a9b6f0 Reduce number of FlexFEC VideoSendStreamTests and lower packet loss.
The intention is to make the tests less flaky.

BUG=webrtc:6744

Review-Url: https://codereview.webrtc.org/2552713002
Cr-Commit-Position: refs/heads/master@{#15421}
2016-12-05 13:38:27 +00:00
asapersson
4eb03c76fa Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
Error resilience is currently always enabled for VP9 which reduces quality.

BUG=webrtc:6783

Review-Url: https://codereview.webrtc.org/2532053002
Cr-Commit-Position: refs/heads/master@{#15390}
2016-12-02 16:58:02 +00:00