1125 Commits

Author SHA1 Message Date
elad.alon
5bbf43f9d4 Move delay_based_bwe_ into CongestionController
BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2725823002
Cr-Commit-Position: refs/heads/master@{#17146}
2017-03-09 14:40:08 +00:00
nisse
c69385de8b Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
Use of FlexFEC is known when streams are created in
WebRtcVideoChannel2, so this replaces the code in Call to infer
FlexFEC config of video streams from the configuration of the FlexFEC
stream(s). This also allows us to switch to a more logical creation
order, where media streams are created before the FlexFEC stream.

This is done in preparation for a larger refactoring of the RTP
demuxing done in Call.

BUG=None

Review-Url: https://codereview.webrtc.org/2712683002
Cr-Commit-Position: refs/heads/master@{#17143}
2017-03-09 14:13:20 +00:00
ilnik
3ae7c259ef Disable failing 15 thumbernails full-stack test on windows
BUG=webrtc:7301

Review-Url: https://codereview.webrtc.org/2737703005
Cr-Commit-Position: refs/heads/master@{#17105}
2017-03-08 09:17:35 +00:00
ilnik
f89a738626 Disable failing fullstack test with 15 thumbnail streams
BUG=webrtc:7301

Review-Url: https://codereview.webrtc.org/2739613003
Cr-Commit-Position: refs/heads/master@{#17095}
2017-03-07 14:15:27 +00:00
ilnik
566c43b525 Cleaning up full-stack simulcast tests and making them more realistic.
BUG=none

Review-Url: https://codereview.webrtc.org/2734753003
Cr-Commit-Position: refs/heads/master@{#17093}
2017-03-07 12:42:54 +00:00
ilnik
a014cc5eb1 Reland of "Added large room scenario to full-stack tests"
Added thumbnail streams functionality to video quality test.

Changed simulcast full-stack tests to be 30fps instead of 50 to
better reflect real usecases (expect all kind of perf metrics to
improve).

BUG=webrtc:7095, webrtc:7301

Review-Url: https://codereview.webrtc.org/2733943003
Cr-Commit-Position: refs/heads/master@{#17092}
2017-03-07 12:21:04 +00:00
ilnik
bfb124596e Revert of Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v… (patchset #8 id:140001 of https://codereview.webrtc.org/2730073002/ )
Reason for revert:
webrtc_perf_tests crashes on android and windows due to too large test.

Original issue's description:
> Added large room scenario to full-stack tests. Added thumbnail streams functionality to video quality test.
>
> Changed simulcast full-stack tests to be 30fps instead of 50 to better reflect real usecases (expect all kind of perf metrics to improve).
>
> BUG=webrtc:7095
>
> Review-Url: https://codereview.webrtc.org/2730073002
> Cr-Commit-Position: refs/heads/master@{#17068}
> Committed: d8bd1b1d82

TBR=sprang@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2734753004
Cr-Commit-Position: refs/heads/master@{#17071}
2017-03-06 15:35:13 +00:00
ilnik
d8bd1b1d82 Added large room scenario to full-stack tests. Added thumbnail streams functionality to video quality test.
Changed simulcast full-stack tests to be 30fps instead of 50 to better reflect real usecases (expect all kind of perf metrics to improve).

BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2730073002
Cr-Commit-Position: refs/heads/master@{#17068}
2017-03-06 14:10:28 +00:00
tommi
db23ea69b6 Add performance tracing for PlatformThread and parts of the video code.
BUG=webrtc:7219

Review-Url: https://codereview.webrtc.org/2729783004
Cr-Commit-Position: refs/heads/master@{#17009}
2017-03-03 15:21:18 +00:00
danilchap
4336d73bd3 Reland of Fix flaky EndToEndTest.TransportSeqNumOnAudioAndVideo (patchset #1 id:1 of https://codereview.webrtc.org/2730893002/ )
Reason for revert:
Fixing race and relanding

Original issue's description:
> Revert of Fix flaky EndToEndTest.TransportSeqNumOnAudioAndVideo (patchset #2 id:20001 of https://codereview.webrtc.org/2728103002/ )
>
> Reason for revert:
> Fails on tsan
> http://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/10235
>
> Original issue's description:
> > Fix flaky EndToEndTest.TransportSeqNumOnAudioAndVideo
> > Add extra checks to it to simplify diagnostic should it fail again.
> >
> > BUG=webrtc:7292
> >
> > Review-Url: https://codereview.webrtc.org/2728103002
> > Cr-Commit-Position: refs/heads/master@{#16999}
> > Committed: bcb6004a9d
>
> TBR=asapersson@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7292
>
> Review-Url: https://codereview.webrtc.org/2730893002
> Cr-Commit-Position: refs/heads/master@{#17000}
> Committed: 9a59473f2c

TBR=asapersson@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7292

Review-Url: https://codereview.webrtc.org/2730673004
Cr-Commit-Position: refs/heads/master@{#17006}
2017-03-03 14:21:54 +00:00
danilchap
9a59473f2c Revert of Fix flaky EndToEndTest.TransportSeqNumOnAudioAndVideo (patchset #2 id:20001 of https://codereview.webrtc.org/2728103002/ )
Reason for revert:
Fails on tsan
http://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/10235

Original issue's description:
> Fix flaky EndToEndTest.TransportSeqNumOnAudioAndVideo
> Add extra checks to it to simplify diagnostic should it fail again.
>
> BUG=webrtc:7292
>
> Review-Url: https://codereview.webrtc.org/2728103002
> Cr-Commit-Position: refs/heads/master@{#16999}
> Committed: bcb6004a9d

TBR=asapersson@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7292

Review-Url: https://codereview.webrtc.org/2730893002
Cr-Commit-Position: refs/heads/master@{#17000}
2017-03-03 13:04:49 +00:00
danilchap
bcb6004a9d Fix flaky EndToEndTest.TransportSeqNumOnAudioAndVideo
Add extra checks to it to simplify diagnostic should it fail again.

BUG=webrtc:7292

Review-Url: https://codereview.webrtc.org/2728103002
Cr-Commit-Position: refs/heads/master@{#16999}
2017-03-03 12:38:11 +00:00
tommi
dea489f33e Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule.
This makes a few things a lot clearer when looking at perf trace data:

* What module instances (where they were created) are called
* On what thread
* How frequently
* For how long

ProcessThread will be replaced by TaskQueue moving forward and this is a step towards understanding the behavior of the affected code.

BUG=webrtc:7219

Review-Url: https://codereview.webrtc.org/2729053002
Cr-Commit-Position: refs/heads/master@{#16998}
2017-03-03 11:20:24 +00:00
kthelgason
2fc52544b0 Fix flaky ViEEncoder tests.
This CL fixes two issues. The first is a tsan complaint about a data
race. The test had a fix to make it deterministic but the race still
existed, so this adds locks around the critical section to appease
the sanitizer.

The second, more annoying issue, is that occasionally the test would
start before the encoder had been configured, as this happens
asynchronously on a task queue. This would cause frames to be queued
up and dropped, but not where we expected them to be dropped.
This was causing some tests to fail very occasionally. I've added
a synchronisation mechanism by posting a barrier task on the queue
and not proceeding with the tests until it finishes.

BUG=webrtc:7217, webrtc:7232, webrtc:7260

Review-Url: https://codereview.webrtc.org/2722183004
Cr-Commit-Position: refs/heads/master@{#16995}
2017-03-03 08:24:41 +00:00
asapersson
a563c21e6a Increase kMinRequiredSamples (5 -> 200) for updating histogram stats from OnFrameBufferTimingsUpdated callback.
Callback used to be reported periodically (each 1000ms) but is now reported per frame.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2729903002
Cr-Commit-Position: refs/heads/master@{#16977}
2017-03-02 16:25:46 +00:00
ilnik
9fd9f6c15f Fixed VP8 simulcast full-stack-tests to not decode non-selected streams.
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2730433005
Cr-Commit-Position: refs/heads/master@{#16976}
2017-03-02 16:10:10 +00:00
danilchap
9f5b6224e8 Disabled flaky VideoSendStreamTest.DoesUtilizeUlpfecForVp*WithNackEnabled
BUG=webrtc:7285
TBR=sprang@webrtc.org
NOTRY=true

Review-Url: https://codereview.webrtc.org/2727263002
Cr-Commit-Position: refs/heads/master@{#16971}
2017-03-02 14:22:21 +00:00
ilnik
68af10de7f Revert of fixed VP8 simulcast to not decode non-selected streams (patchset #5 id:80001 of https://codereview.webrtc.org/2728553003/ )
Reason for revert:
Causes regression in VP8 simulcast metrics (receive time, encoded frame size, etc) as two excluded streams' decoders request keyframes periodically, which affects metrics of a selected stream.

Original issue's description:
> In full-stack tests: fixed VP8 simulcast to not decode non-selected streams.
>
> BUG=webrtc:7095
>
> Review-Url: https://codereview.webrtc.org/2728553003
> Cr-Commit-Position: refs/heads/master@{#16948}
> Committed: 8dccd67520

TBR=sprang@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2729623005
Cr-Commit-Position: refs/heads/master@{#16967}
2017-03-02 12:59:33 +00:00
ilnik
8dccd67520 In full-stack tests: fixed VP8 simulcast to not decode non-selected streams.
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2728553003
Cr-Commit-Position: refs/heads/master@{#16948}
2017-03-01 15:10:30 +00:00
ilnik
daa574d1db Adding memory usage metric to full-stack video tests (only for WIN until we find more stable method to measure memory usage)
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2719013004
Cr-Commit-Position: refs/heads/master@{#16945}
2017-03-01 14:46:05 +00:00
perkj
c5726c1579 Cleanup video test target dependencies in video_test_common.
The purpose is to fix (Asan discovered) duble definitions in upstream project.

BUG=none

Review-Url: https://codereview.webrtc.org/2721303002
Cr-Commit-Position: refs/heads/master@{#16939}
2017-03-01 11:37:08 +00:00
sprang
f24a06406a Recreate WebrtcVideoSendStream if screen content setting is changed.
This avoids the situation where an encoder, not supporting certain
screen content settings, is created for a config where screencast is
off, and later ReconfigureEncoder() is called updating the configuration
but not the encoder instance, causing an inconsistency in the encoder's
InitEncode() call.

TBR=pthatcher@webrtc.org
BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2710493008
Cr-Commit-Position: refs/heads/master@{#16921}
2017-02-28 21:23:26 +00:00
danilchap
b050cf9903 Disable ViEEncoderTest.InitialFrameDropOffWhenEncoderDisabledScaling on tsan
BUG=webrtc:7260
TBR=magjed@webrtc.org
NOTRY=true

Review-Url: https://codereview.webrtc.org/2722863002
Cr-Commit-Position: refs/heads/master@{#16919}
2017-02-28 18:12:29 +00:00
sprang
c1b57a15bf Test field trial group with startswith rather than equals.
BUG=webrtc:7266

Review-Url: https://codereview.webrtc.org/2717973005
Cr-Commit-Position: refs/heads/master@{#16915}
2017-02-28 16:50:47 +00:00
sprang
f8ee65ead6 Fix rare race in ViEEncoderTest.UMACpuLimitedResolutionInPercent
The test aims to synchronize by waiting for the last sent frame, using
sink_.WaitForEncodedFrame(). Then it stops the vie_encoder instance and
finally reset the SendStatsProxy to trigger UMA stats to be registered.

The problem is that the callback that triggers WaitForEncodedFrame() to
release will call FrameSent() on the overuse detecter just after that,
and so it may try to update the stats proxy while it is being deleted.

To prevent this, simply destroy the vie_encoder instance too, so that
we know that the task queue has been stopped.

BUG=None

Review-Url: https://codereview.webrtc.org/2720183004
Cr-Commit-Position: refs/heads/master@{#16914}
2017-02-28 16:49:33 +00:00
philipel
2dfea3e548 Avoid busy looping as the VideoReceiveStream is shut down.
BUG=webrtc:7269

Review-Url: https://codereview.webrtc.org/2720963003
Cr-Commit-Position: refs/heads/master@{#16910}
2017-02-28 15:19:43 +00:00
johan
f2183ffbf6 Create unit test for VideoReceiveStream.
- Create an unit test skeleton for VideoReceiveStream.
- Add an actual test case for creating a frame from H264 Sprop Parameter
  Sets and an Rtp H264 IDR Nalu.

BUG=webrtc:5948

Review-Url: https://codereview.webrtc.org/2721653002
Cr-Commit-Position: refs/heads/master@{#16892}
2017-02-28 09:33:09 +00:00
danilchap
f8475433be Disable flaky ViEEncoderTest.NrOfDroppedFramesLimited
BUG=webrtc:7232
NOTRY=true
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/2720003002
Cr-Commit-Position: refs/heads/master@{#16876}
2017-02-27 17:21:14 +00:00
perkj
a8ba195db5 Replace test::FrameGenerator::ChromaGenerator with new FrameGenerator::SquareGenerator The problem with the ChromaGenerator is that the VP8 encoder produce a too low bitrate for each frame. The squaregenerator make the VP8 encoder produce about 600kbit/s at VGA.
SquareGenerator is a FrameGenerator that draws 10 randomly sized and colored
squares. Between each new generated frame, the squares are moved slightly
towards the lower right corner.

BUG=webrtc:7192

Review-Url: https://codereview.webrtc.org/2705973002
Cr-Commit-Position: refs/heads/master@{#16870}
2017-02-27 14:52:10 +00:00
sprang
21253fc2c8 Fix race condition in VideoSendStreamTest.ChangingTransportOverhead
BUG=webrtc:7190

Review-Url: https://codereview.webrtc.org/2708573003
Cr-Commit-Position: refs/heads/master@{#16859}
2017-02-27 11:35:47 +00:00
tommi
f284b7ff5f Remove |running_| state from NackModule to avoid running a tight loop.
Also reducing locking in NackModule (and by extension RtpStreamReceiver)

BUG=webrtc:7246

Review-Url: https://codereview.webrtc.org/2720603002
Cr-Commit-Position: refs/heads/master@{#16856}
2017-02-27 09:59:36 +00:00
kthelgason
fa5fdce100 Reland of Set scaling limit at 320 * 180 for all implementations. (patchset #1 id:1 of https://codereview.webrtc.org/2711913007/ )
Reason for revert:
Reland after fixing broken perf tests.

Original issue's description:
> Revert of Set scaling limit at 320 * 180 for all implementations. (patchset #2 id:20001 of https://codereview.webrtc.org/2709153002/ )
>
> Reason for revert:
> Looks like webrtc_perf_test started failing on linux, mac and windows after this cl landed.
>
> Example failure:
>
> https://build.chromium.org/p/client.webrtc.perf/builders/Linux%20Trusty/builds/1386/steps/webrtc_perf_tests/logs/stdio
>
> [ RUN      ] CallPerfTest.ReceivesCpuOveruseAndUnderuse
> ../../webrtc/call/call_perf_tests.cc:522: Failure
> Value of: Wait()
>   Actual: false
> Expected: true
> Timed out before receiving an overuse callback.
> [  FAILED  ] CallPerfTest.ReceivesCpuOveruseAndUnderuse (120056 ms)
>
>
> Original issue's description:
> > Set scaling limit at 320 * 180 for all implementations.
> >
> > The MediaCodec decoder on android has trouble decoding video at
> > so low resolutions. We set the limit a bit higher for all implementations
> > pending a robust software fallback implementation for MediaCodec.
> >
> > BUG=webrtc:7206
> >
> > Review-Url: https://codereview.webrtc.org/2709153002
> > Cr-Commit-Position: refs/heads/master@{#16798}
> > Committed: 560ddb7321
>
> TBR=magjed@webrtc.org,sprang@webrtc.org,kthelgason@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7206
>
> Review-Url: https://codereview.webrtc.org/2711913007
> Cr-Commit-Position: refs/heads/master@{#16839}
> Committed: 37510bf094

TBR=magjed@webrtc.org,sprang@webrtc.org,tommi@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7206

Review-Url: https://codereview.webrtc.org/2718013002
Cr-Commit-Position: refs/heads/master@{#16853}
2017-02-27 08:15:31 +00:00
tommi
37510bf094 Revert of Set scaling limit at 320 * 180 for all implementations. (patchset #2 id:20001 of https://codereview.webrtc.org/2709153002/ )
Reason for revert:
Looks like webrtc_perf_test started failing on linux, mac and windows after this cl landed.

Example failure:

https://build.chromium.org/p/client.webrtc.perf/builders/Linux%20Trusty/builds/1386/steps/webrtc_perf_tests/logs/stdio

[ RUN      ] CallPerfTest.ReceivesCpuOveruseAndUnderuse
../../webrtc/call/call_perf_tests.cc:522: Failure
Value of: Wait()
  Actual: false
Expected: true
Timed out before receiving an overuse callback.
[  FAILED  ] CallPerfTest.ReceivesCpuOveruseAndUnderuse (120056 ms)

Original issue's description:
> Set scaling limit at 320 * 180 for all implementations.
>
> The MediaCodec decoder on android has trouble decoding video at
> so low resolutions. We set the limit a bit higher for all implementations
> pending a robust software fallback implementation for MediaCodec.
>
> BUG=webrtc:7206
>
> Review-Url: https://codereview.webrtc.org/2709153002
> Cr-Commit-Position: refs/heads/master@{#16798}
> Committed: 560ddb7321

TBR=magjed@webrtc.org,sprang@webrtc.org,kthelgason@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7206

Review-Url: https://codereview.webrtc.org/2711913007
Cr-Commit-Position: refs/heads/master@{#16839}
2017-02-25 19:30:55 +00:00
ilnik
1e7732c3d9 Fixed Full stack tests to correctly process selected TL and SL while
calculating frame sizes. Added actual_bitrate metric which also accounts
for TL and SL info. Metric encoded_frame_size calculation is cleaned up. Perf alerts should be ignored.

BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2709483009
Cr-Commit-Position: refs/heads/master@{#16800}
2017-02-23 13:07:56 +00:00
kthelgason
560ddb7321 Set scaling limit at 320 * 180 for all implementations.
The MediaCodec decoder on android has trouble decoding video at
so low resolutions. We set the limit a bit higher for all implementations
pending a robust software fallback implementation for MediaCodec.

BUG=webrtc:7206

Review-Url: https://codereview.webrtc.org/2709153002
Cr-Commit-Position: refs/heads/master@{#16798}
2017-02-23 11:22:02 +00:00
ilnik
df92c5cb8c Adding cpu measurments to video_quality_tests
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2711493002
Cr-Commit-Position: refs/heads/master@{#16791}
2017-02-23 10:08:44 +00:00
tommi
0f8b403eb5 Introduce a new constructor to PlatformThread.
The new constructor introduces two new changes:

* Support specifying thread priority at construction time.
  - Moving forward, the SetPriority() method will be removed.
* New thread function type.
  - The new type has 'void' as a return type and a polling loop
    inside PlatformThread, is not used.

The old function type is still supported until all places have been moved over.

In this CL, the first steps towards deprecating the old mechanism are taken
by moving parts of the code that were simple to move, over to the new callback
type.

BUG=webrtc:7187

Review-Url: https://codereview.webrtc.org/2708723003
Cr-Commit-Position: refs/heads/master@{#16779}
2017-02-22 19:22:05 +00:00
philipel
a45102f7b4 Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ )
Reason for revert:
Fix here: https://codereview.chromium.org/2708593003

Original issue's description:
> Revert Make the new jitter buffer the default jitter buffer.
>
> Speculative revert of https://codereview.chromium.org/2656983002/ to see if it fixes a downstream bug.
>
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2682073003
> Cr-Commit-Position: refs/heads/master@{#16492}
> Committed: e525d6aba6

TBR=nisse@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2704183002
Cr-Commit-Position: refs/heads/master@{#16772}
2017-02-22 13:30:39 +00:00
Tommi
5dd5f5a319 RembWithSendSideBwe: Rename |event_| to |stop_event_| and set it when the test ends.
BUG=webrtc:7200
R=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2706223002 .
Cr-Commit-Position: refs/heads/master@{#16744}
2017-02-21 13:22:59 +00:00
perkj
070ba85f5b Replace DCHECK with ASSERT_TRUE in vie_encoder_unittest.cc
BUG=none
TBR=sprang@webrtc.org

Review-Url: https://codereview.webrtc.org/2699593007
Cr-Commit-Position: refs/heads/master@{#16656}
2017-02-16 23:46:27 +00:00
ilnik
2a8c2f589a Added Vp9 simulcast tests.
For them implemeted upscaling in libyuv metrics calculation.
Updated maximum number of SL in vp9 encoder to 3.
Refactored names of some fields in Video_quality_check analyzer.

BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2681683003
Cr-Commit-Position: refs/heads/master@{#16625}
2017-02-15 10:23:28 +00:00
ilnik
9ae0d76b92 Added WebRTC-QuickPerfTest field trial. If enabled only 1 frame will be sent.
BUG=webrtc:7101

Review-Url: https://codereview.webrtc.org/2690903004
Cr-Commit-Position: refs/heads/master@{#16622}
2017-02-15 08:53:12 +00:00
kthelgason
b83797bd7a Fix flaky ViEEncoder unit test.
The flaky test was introduced in ad9010c9836, and is essentially a race
where the ViE Encoder has already configured the quality scaler on the
encoder thread before we've updated the ScalingSettings. This CL adds
a forced reconfiguration of the quality scaler to avoid this issue.

BUG=None
TBR=sprang@webrtc.org

Review-Url: https://codereview.webrtc.org/2695873004
Cr-Commit-Position: refs/heads/master@{#16612}
2017-02-14 19:57:25 +00:00
kthelgason
ad9010c983 Make sure initial framedrop is off where quality scaling is off.
BUG=chromium:689972,chromium:689915

Review-Url: https://codereview.webrtc.org/2684683004
Cr-Commit-Position: refs/heads/master@{#16602}
2017-02-14 08:46:51 +00:00
nisse
38cc1d6b31 Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.
This avoids redoing RTP header parsing already done in Call, for video.

The next step is to convert other types of receive streams, i.e.,
audio and flexfec, to use a compatible OnRtpPacket method. We can then
introduce a shared base interface, and simplify media-independent
receive processing in Call.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2681673004
Cr-Commit-Position: refs/heads/master@{#16583}
2017-02-13 13:59:46 +00:00
sprang
552c7c70b0 Make SendStatisticsProxy paramter mandatory in ViEEncoder ctor.
The only implementation which used a nullptr was a mock used in tests,
so add a dummy instance there instead.
Remove tests for stats_proxy_ in vie_encoder and just dcheck in the
constructor instead.

BUG=None

Review-Url: https://codereview.webrtc.org/2695643002
Cr-Commit-Position: refs/heads/master@{#16577}
2017-02-13 12:41:45 +00:00
sprang
be03724ae1 Fix nits in vie_encoder
Updated comment.
Don't call AdaptUp/AdaptDown in tests without first emitting a frame.
Handle frame received precondition in AdaptUp/AdaptDown with DCHECK
instead of return.

BUG=webrtc:4172, webrtc:6850

Review-Url: https://codereview.webrtc.org/2690023002
Cr-Commit-Position: refs/heads/master@{#16572}
2017-02-13 10:38:17 +00:00
ilnik
46a0021e4e Retransmitted packets are now counted in receive time
BUG=chromium:690358

Review-Url: https://codereview.webrtc.org/2683423002
Cr-Commit-Position: refs/heads/master@{#16536}
2017-02-10 17:16:05 +00:00
sprang
84a3759825 Change rtc::VideoSinkWants to have target and a max pixel count
The current method with max_pixel_count and max_pixel_count_step_up,
where only one should be used at a time and this first signaling an
inclusive upper bound and other other an exclusive lower bound, makes
for a lot of confusion.

I've updated this to have a desired target and a maximum instead. The
source should select a resolution as close to the target as possible,
but no higher than the maximum.

I intend to also add similar frame rate settings in an upcoming cl.

BUG=webrtc:4172,webrtc:6850

Review-Url: https://codereview.webrtc.org/2672793002
Cr-Commit-Position: refs/heads/master@{#16533}
2017-02-10 15:04:27 +00:00
ilnik
1e1c84db10 Fixing typo
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2686033004
Cr-Commit-Position: refs/heads/master@{#16518}
2017-02-09 16:32:53 +00:00