22 Commits

Author SHA1 Message Date
Mirko Bonadei
245660a33d Fix Gn untracked headers in webrtc/call.
This CL is the same CL we had at https://codereview.webrtc.org/3014543002/.
Since we cannot land it with Rietveld anymore let's move the discussion
to Gerrit.

BUG=webrtc:7641
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I5662bec318544b07f476c12ecada997d726e7361
Reviewed-on: https://webrtc-review.googlesource.com/7981
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20224}
2017-10-10 15:13:48 +00:00
Edward Lemur
88b23f6662 Fix flag name in low_bandwidth_audio_test.py
TBR=kjellander@webrtc.org, oprypin@webrtc.org

No-Try: true
Bug: chromium:755660
Change-Id: I6bf5dda5374cae16da54aa10e77b136c638e1975
Reviewed-on: https://webrtc-review.googlesource.com/6442
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20148}
2017-10-04 17:26:14 +00:00
Edward Lemur
7e3b5697d9 Ignore swarming arguments in low_bandwidth_audio_test.py
Needed because swarming adds --isolated-script-test-output and --isolated-script-test-perf-output

See for example:
https://chromium-swarm.appspot.com/task?id=39006c763bebf710

No-Try: true
Bug: chromium:755660
Change-Id: Iff9fb3441200f760c511a67211fbc4a1272717b4
Reviewed-on: https://webrtc-review.googlesource.com/6362
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20146}
2017-10-04 16:19:44 +00:00
Edward Lemur
b0250f0504 Reland "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script."
This is a reland of f4898a650954691d79bbc146d5b454fb5e67ec47
Original change's description:
> Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script.
> 
> They should've been downloaded already.
> 
> NOTRY=True
> 
> Bug: chromium:755660
> Change-Id: I8ecb355f06026a38bd9377633e2be6c55d7c6452
> Reviewed-on: https://webrtc-review.googlesource.com/5620
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20109}

No-Try: true
Bug: chromium:755660
Change-Id: I391130545eee5d4928101f87ac4a4e0945d665a1
Reviewed-on: https://webrtc-review.googlesource.com/6380
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20140}
2017-10-04 13:03:24 +00:00
Danil Chapovalov
90e1f539a5 Fix potentional race in AudioSendStream constructor
RegisterPacketFeedbackObserver signals congestion controller object is
ready to process incoming packet, thus call it as last statement in the constructor

Bug: webrtc:8325
Change-Id: I31d8ab04c568e639db12c97b649c2d50a489ce24
Reviewed-on: https://webrtc-review.googlesource.com/5860
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20121}
2017-10-03 16:15:33 +00:00
Niels Möller
c3fa8e1ce7 New method RtpReceiver::GetLatestTimestamps.
The two timestamps, rtp time and corresponding system time, are always
used together, for audio/video sync. The new method reads both
timestamps, without releasing a lock in between. Ensures that the
caller gets values corresponding to the same packet.

Bug: webrtc:7135
Change-Id: I25bdcbe9ad620016bfad39841b339c266efade14
Reviewed-on: https://webrtc-review.googlesource.com/4062
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20120}
2017-10-03 16:14:29 +00:00
Edward Lemur
45a0b36d3f Revert "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script."
Reason for revert: Breaks windows bot.
https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/3804

TBR=kjellander@webrtc.org,phoglund@webrtc.org,ehmaldonado@webrtc.org,oprypin@webrtc.org

Change-Id: I0f2221b66c4f7dcf0a6f03004e5acc24c77ba8b0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:755660
Reviewed-on: https://webrtc-review.googlesource.com/6001
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20117}
2017-10-03 14:00:15 +00:00
Edward Lemur
f4898a6509 Reland "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script."
This reverts commit bb1222f3adacec0a096451a79c391663e9f22335.

Reason for revert: Fixing

Original change's description:
> Revert "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script."
> 
> This reverts commit 20196984558b890ee7648d5cc778bc055fdeb5b2.
> 
> Reason for revert: Fails on Mac
> https://build.chromium.org/p/client.webrtc.perf/builders/Mac%2010.11/builds/4070
> 
> Original change's description:
> > Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script.
> > 
> > They should've been downloaded already.
> > 
> > NOTRY=True
> > 
> > Bug: chromium:755660
> > Change-Id: I8ecb355f06026a38bd9377633e2be6c55d7c6452
> > Reviewed-on: https://webrtc-review.googlesource.com/5620
> > Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20109}
> 
> TBR=kjellander@webrtc.org,phoglund@webrtc.org,ehmaldonado@webrtc.org,oprypin@webrtc.org
> 
> Change-Id: I0cfc1d0b398587a023af528536e7d995c6de1413
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:755660
> Reviewed-on: https://webrtc-review.googlesource.com/5940
> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20112}

No-Try: true
Bug: chromium:755660
Change-Id: Idbf30003c78c4093afcaf99dc0c5bb3468cdcf6d
Reviewed-on: https://webrtc-review.googlesource.com/5941
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20113}
2017-10-03 12:49:51 +00:00
Edward Lemur
bb1222f3ad Revert "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script."
This reverts commit 20196984558b890ee7648d5cc778bc055fdeb5b2.

Reason for revert: Fails on Mac
https://build.chromium.org/p/client.webrtc.perf/builders/Mac%2010.11/builds/4070

Original change's description:
> Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script.
> 
> They should've been downloaded already.
> 
> NOTRY=True
> 
> Bug: chromium:755660
> Change-Id: I8ecb355f06026a38bd9377633e2be6c55d7c6452
> Reviewed-on: https://webrtc-review.googlesource.com/5620
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20109}

TBR=kjellander@webrtc.org,phoglund@webrtc.org,ehmaldonado@webrtc.org,oprypin@webrtc.org

Change-Id: I0cfc1d0b398587a023af528536e7d995c6de1413
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:755660
Reviewed-on: https://webrtc-review.googlesource.com/5940
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20112}
2017-10-03 12:33:52 +00:00
Edward Lemur
2019698455 Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script.
They should've been downloaded already.

NOTRY=True

Bug: chromium:755660
Change-Id: I8ecb355f06026a38bd9377633e2be6c55d7c6452
Reviewed-on: https://webrtc-review.googlesource.com/5620
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20109}
2017-10-03 11:25:10 +00:00
Edward Lemur
2011075a58 MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py.
NOTRY=True

Bug: chromium:755660
Change-Id: I92de99cd1e3dd206f6cd366dbfd1c8c211d37cc7
Reviewed-on: https://webrtc-review.googlesource.com/4420
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20089}
2017-10-02 16:57:09 +00:00
Gustaf Ullberg
b0a0207838 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio
Description of this stat can be found here:
https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-jitterbufferdelay

Bug: webrtc:8281
Change-Id: Ib2e8174f3449e68ad419ae2d58d5565fc9854023
Reviewed-on: https://webrtc-review.googlesource.com/3381
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20069}
2017-10-02 10:47:00 +00:00
solenberg
1c239d476e Remove voe::Statistics.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3020473002
Cr-Commit-Position: refs/heads/master@{#20042}
2017-09-29 13:00:28 +00:00
solenberg
fc3a2e3393 Remove the VoiceEngineObserver callback interface.
BUG=webrtc:4690
TBR=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3019513002
Cr-Commit-Position: refs/heads/master@{#19976}
2017-09-26 16:35:01 +00:00
solenberg
2397b9a114 Remove voe::OutputMixer and AudioConferenceMixer.
This code path is not used anymore.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3015553002
Cr-Commit-Position: refs/heads/master@{#19929}
2017-09-22 13:48:10 +00:00
solenberg
4652e86c0c Disable flaky AudioStats.NoLoss test.
BUG=none

Review-Url: https://codereview.webrtc.org/3013783002
Cr-Commit-Position: refs/heads/master@{#19928}
2017-09-22 13:07:56 +00:00
Gustaf Ullberg
9a2e906b0c Added RTCMediaStreamTrackStats.concealmentEvents
The number of concealment events. This counter increases every time a concealed sample is
synthesized after a non-concealed sample. That is, multiple consecutive concealed samples
will increase the concealedSamples count multiple times but is a single concealment event.

Bug: webrtc:8246
Change-Id: I7ef404edab765218b1f11e3128072c5391e83049
Reviewed-on: https://webrtc-review.googlesource.com/1221
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19881}
2017-09-18 08:58:06 +00:00
solenberg
18f5427e4c Remove voe_auto_test and add new tests to cover the missing cases.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3007383002
Cr-Commit-Position: refs/heads/master@{#19865}
2017-09-15 16:56:08 +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
Henrik Kjellander
5a6aa4f05d Fix path to root in low_bandwidth_audio_test.py
BUG=chromium:611808
TBR=solenberg@webrtc.org
NOTRY=True
NOPRESUBMIT=True
NOTREECHECKS=True

Change-Id: Iba2b0851ee99916b9809231b4b27046315fd8565
Reviewed-on: https://webrtc-review.googlesource.com/1569
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19849}
2017-09-15 08:29:11 +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