985 Commits

Author SHA1 Message Date
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
sprang
1a646ee522 Wire up BitrateAllocation to be sent as RTCP TargetBitrate
This is the video parts of https://codereview.webrtc.org/2531383002/
Wire up BitrateAllocation to be sent as RTCP TargetBitrate

BUG=webrtc:6301

Review-Url: https://codereview.webrtc.org/2541303003
Cr-Commit-Position: refs/heads/master@{#15359}
2016-12-01 14:34:18 +00:00
kthelgason
5e13d41124 Remove limit on how often quality scaling downscales
When starting from 720p this is necessary to achieve acceptable
quality at low bitrates.

BUG=webrtc:6495

Review-Url: https://codereview.webrtc.org/2538913003
Cr-Commit-Position: refs/heads/master@{#15356}
2016-12-01 11:59:56 +00:00
magjed
dd40702357 Move VideoDecoder::Create() logic to separate internal video decoder factory
The goal with this CL is to move implementation details out from the
webrtc root (webrtc/video_decoder.h) to simplify the dependency graph.
Another goal is to streamline the creation of VideoDecoders in
webrtcvideoengine2.cc; it will now have two factories of the same
WebRtcVideoDecoderFactory type, one internal and one external.

Specifically, this CL:
 * Removes webrtc::VideoDecoder::DecoderType and use webrtc::VideoCodecType
   instead.
 * Removes 'static VideoDecoder* Create(DecoderType codec_type)' and
   moves the create function to the internal decoder factory instead.
 * Removes video_decoder.cc. webrtc::VideoDecoder is now just an
   interface without any static functions.

BUG=webrtc:6743

Review-Url: https://codereview.webrtc.org/2521203002
Cr-Commit-Position: refs/heads/master@{#15350}
2016-12-01 08:27:35 +00:00
brandtr
aa354c9512 Rename full_stack.cc to full_stack_tests.cc.
Also rename the accompanying plot file.

BUG=None

Review-Url: https://codereview.webrtc.org/2529293006
Cr-Commit-Position: refs/heads/master@{#15349}
2016-12-01 08:20:24 +00:00
brandtr
93c5d030fc Start gathering perf data for VP8 + FlexFEC.
This is to assess the performance penalty of the (current)
lack of integration with FlexFEC and BWE.

This CL also enables send-side BWE for the following tests:
- foreman_cif_net_delay_0_0_plr_0_VP9
- foreman_cif_net_delay_0_0_plr_0_H264
- foreman_cif_delay_50_0_plr_5_VP9
- foreman_cif_delay_50_0_plr_5_H264
- foreman_cif_delay_50_0_plr_5_H264_flexfec
- foreman_cif_delay_50_0_plr_5_H264_ulpfec
Perf alerts on these tests are therefore expected.

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

Review-Url: https://codereview.webrtc.org/2534203004
Cr-Commit-Position: refs/heads/master@{#15339}
2016-11-30 15:50:13 +00:00
nisse
13d38fbe90 Delete all of the video_processing module but the denoiser code.
It is unused since cl https://codereview.webrtc.org/2386573002.

The new denoiser implementation and its tests are kept for now. This
code is also unused, but there are some plans to take this code into
use in the not too distant future.

BUG=None

Review-Url: https://codereview.webrtc.org/2496153002
Cr-Commit-Position: refs/heads/master@{#15338}
2016-11-30 15:44:59 +00:00
asapersson
076c0118c5 Change unit of logged bitrate stats in bytes/s to bits/s.
Multiplier added to ToString method in AggregatedStats.

BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2535323003
Cr-Commit-Position: refs/heads/master@{#15330}
2016-11-30 13:17:21 +00:00
minyue
78b4d56535 Relanding "Pass time constant to bwe smoothing filter."
An earlier attempt to land this was in https://codereview.webrtc.org/2518923003/

It was failed because it removed an API. This CL fixes this.

BUG=webrtc:6443, webrtc:6303

Review-Url: https://codereview.webrtc.org/2536753002
Cr-Commit-Position: refs/heads/master@{#15325}
2016-11-30 12:47:47 +00:00
nisse
0245da0fa0 Move ownership of PacketRouter from CongestionController to Call.
And delete the method CongestionController::packet_router.

BUG=None

Review-Url: https://codereview.webrtc.org/2516983004
Cr-Commit-Position: refs/heads/master@{#15323}
2016-11-30 11:35:28 +00:00
hbos
706a45e68e Added missing include to fix waterfall compile error.
Bots failue caused by https://codereview.webrtc.org/2517243005/

NOTRY=True
TBR=stefan@webrtc.org
BUG=webrtc:6638

Review-Url: https://codereview.webrtc.org/2544473002
Cr-Commit-Position: refs/heads/master@{#15318}
2016-11-30 09:53:19 +00:00
asapersson
0c43f779f8 Update video histograms that do not have a minimum lifetime limit before being recorded.
Updated histograms:
"WebRTC.Video.ReceivedPacketsLostInPercent" (two RTCP RR previously needed)
"WebRTC.Video.ReceivedFecPacketsInPercent" (one received packet previously needed)
"WebRTC.Video.RecoveredMediaPacketsInPercentOfFec" (one received FEC packet previously needed)

Prevents logging stats if call was shortly in use.

BUG=b/32659204

Review-Url: https://codereview.webrtc.org/2536653002
Cr-Commit-Position: refs/heads/master@{#15315}
2016-11-30 09:42:32 +00:00
philipel
759e0b7241 Fix memory leak in video_coding::PacketBuffer::InsertPacket.
BUG=webrtc:6788

Review-Url: https://codereview.webrtc.org/2535203002
Cr-Commit-Position: refs/heads/master@{#15314}
2016-11-30 09:32:11 +00:00
philipel
be74270ebe Calculate JitterBufferDelayInMs in the new jitter buffer.
JitterBufferDelayInMs is used for the WebRTC-NewVideoJitterBuffer finch
experiment, and therefore needs to be calculated.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2534093003
Cr-Commit-Position: refs/heads/master@{#15313}
2016-11-30 09:31:45 +00:00
michaelt
a33287761a Remove overhead from video bitrate.
BUG=webrtc:6638

Review-Url: https://codereview.webrtc.org/2517243005
Cr-Commit-Position: refs/heads/master@{#15303}
2016-11-29 17:25:10 +00:00
kthelgason
876222f77d Move usage of QualityScaler to ViEEncoder.
This brings QualityScaler much more in line with OveruseFrameDetector.
The two classes are conceptually similar, and should be used in the
same way. The biggest changes in this CL are:
- Quality scaling is now only done in ViEEncoder and not in each
  encoder implementation separately.
- QualityScaler now checks the average QP asynchronously, instead of
  having to be polled on each frame.
- QualityScaler is no longer responsible for actually scaling the frames,
  but has a callback to ViEEncoder that it uses to express it's desire
  for lower resolution.

BUG=webrtc:6495

Review-Url: https://codereview.webrtc.org/2398963003
Cr-Commit-Position: refs/heads/master@{#15286}
2016-11-29 09:44:22 +00:00
asapersson
320e45ad87 Use RateCounter for input/sent fps stats. Reports average of periodically computed stats over a call.
Intervals when video is paused is no longer included in the stats:
"WebRTC.Video.InputFramesPerSecond"
"WebRTC.Video.SentFramesPerSecond"

BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2536743002
Cr-Commit-Position: refs/heads/master@{#15285}
2016-11-29 09:40:46 +00:00
kwiberg
352444fcac RTC_[D]CHECK_op: Remove superfluous casts
There's no longer any need to make the two arguments have the same
signedness, so we can remove a bunch of superfluous (and sometimes
dangerous) casts.

It turned out I also had to fix the safe_cmp functions to properly handle
enums that are implicitly convertible to integers.

NOPRESUBMIT=true
BUG=webrtc:6645

Review-Url: https://codereview.webrtc.org/2534683002
Cr-Commit-Position: refs/heads/master@{#15281}
2016-11-28 23:59:03 +00:00