Commit Graph

145 Commits

Author SHA1 Message Date
adc1e9bf94 Remove old videosourceinterface files.
I have one downstream CL that needs to land first before landing this.

Bug: webrtc:6828
Change-Id: Ib6f3ae78f83775278e4c2e4d34a93fe3748fb851
Reviewed-on: https://webrtc-review.googlesource.com/38340
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21567}
2018-01-11 08:23:33 +00:00
002f921c5d Inline default constructors for MediaChannel structs
Bug: None
Change-Id: I72b534c49d3f26e988d1c92aae09435a9483a930
Reviewed-on: https://webrtc-review.googlesource.com/37143
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21540}
2018-01-10 01:31:40 +00:00
dc8b5ab350 Remove dead code for media channel errors
Bug: None
Change-Id: Ifb8f2cd42a5e24ce8386eff97435890766bbd5fc
Reviewed-on: https://webrtc-review.googlesource.com/37142
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21507}
2018-01-06 00:25:29 +00:00
9e19403d10 Move videosourceinterface to api.
VideoSourceInterface is clearly an integral part of
mediastreaminterface.h already, so moving that interface to api makes
sense. This also resolves a circular dependency in call/.

Bug: webrc:6828
Change-Id: Ic1862f118363b0b55a235a9c0c35d9adc647184c
Reviewed-on: https://webrtc-review.googlesource.com/37500
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21498}
2018-01-05 09:14:19 +00:00
be214a26f8 Move videosinkinterface.h to common_video to solve a circular dep.
I updated some dependency enforcement rules to allow examples and pc
to depend on common_video. I reckoned depending on common_video is
not controversial when they already dependend on media/base, which
is a lower-level abstraction.

Bug: webrtc:6828
Change-Id: I77dbeb10187b4e70dda1d873a29994fa76070758
Reviewed-on: https://webrtc-review.googlesource.com/34187
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21495}
2018-01-04 13:19:49 +00:00
7622048be2 Add an AudioOptions field to force software echo cancellation on iOS.
This is a temporary hack for the iPad Pro 12.9" gen2, which has
non-functional echo cancellation.

Bug: webrtc:8682
Change-Id: I646deeeb4723c4accac6f364c5c76a015791e202
Reviewed-on: https://webrtc-review.googlesource.com/35680
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21417}
2017-12-22 00:04:43 +00:00
4ab68eec96 Move sessiondescription.h/cc from p2p/base to pc/
SDP is a detail of PeerConnection and is not used by anything in p2p, so
it belongs in the pc/ directory. This also allows
MediaContentDescription to be inlined in the future.

Bug: webrtc:8620
Change-Id: I38b65ede9942e29eb15035ab29f2be988da1e5ce
Reviewed-on: https://webrtc-review.googlesource.com/33781
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21376}
2017-12-20 00:21:52 +00:00
3e113438b1 Fix circular dependencies in webrtc_common.
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.

I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.

Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:33:26 +00:00
0a37547033 Add optional stereo codec to SDP negotiation
- Defines stereo codec case, similar to RTX, that adds stereo codec to the SDP
negotiation. The underlying codec's payload type is similarly defined by "apt".
- If this negotiation is successful, codec name is included in sdp line via
"acn".
- Adds codec setting initializers for these specific stereo cases.
- Introduces new Stereo*Factory classes as optional convenience wrappers that
inserts stereo codec to the existing set of supported codecs on demand.

This CL is the step 5 for adding alpha channel support over the wire in webrtc.
Design Doc: https://goo.gl/sFeSUT

Bug: webrtc:7671
Change-Id: Ie12c56c8fcf7934e216135d73af33adec5248f76
Reviewed-on: https://webrtc-review.googlesource.com/22901
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21210}
2017-12-11 16:30:06 +00:00
3a233744eb Reland "Remove the aec_quality_min metric."
This is a reland of 99b1bd1553d442ef7d27755567594ac7e65c53b7
Original change's description:
> Remove the aec_quality_min metric.
> 
> Removing this unused metric.
> 
> Bug: webrtc:8563
> Change-Id: I47446d6aaf5dcc3a8ea57f9248576d68bbe2a304
> Reviewed-on: https://webrtc-review.googlesource.com/30720
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21158}

Bug: webrtc:8563
Change-Id: I622df96528cd6e54e252b22a315840e12d521c7f
Reviewed-on: https://webrtc-review.googlesource.com/31780
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21201}
2017-12-11 10:36:55 +00:00
9a44f96ea7 Delete rtc_base/window.h.
Bug: webrtc:6424
Change-Id: Iaed83b07dd469a9990f48fe41fcdff5e7493eb31
Reviewed-on: https://webrtc-review.googlesource.com/31480
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21194}
2017-12-11 07:59:35 +00:00
a3fad93d87 Revert "Remove the aec_quality_min metric."
This reverts commit 99b1bd1553d442ef7d27755567594ac7e65c53b7.

Reason for revert: breaks downstream projects.

Original change's description:
> Remove the aec_quality_min metric.
> 
> Removing this unused metric.
> 
> Bug: webrtc:8563
> Change-Id: I47446d6aaf5dcc3a8ea57f9248576d68bbe2a304
> Reviewed-on: https://webrtc-review.googlesource.com/30720
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21158}

TBR=solenberg@webrtc.org,gustaf@webrtc.org

Change-Id: I90f16915d517123e4bfba39db64424cdcc4ef03f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8563
Reviewed-on: https://webrtc-review.googlesource.com/31360
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21160}
2017-12-08 11:45:50 +00:00
99b1bd1553 Remove the aec_quality_min metric.
Removing this unused metric.

Bug: webrtc:8563
Change-Id: I47446d6aaf5dcc3a8ea57f9248576d68bbe2a304
Reviewed-on: https://webrtc-review.googlesource.com/30720
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21158}
2017-12-08 10:48:49 +00:00
606a5971e3 Remove adjust_agc_delta from WebRtcVoiceEngine
The setting is no longer used anywhere.

Bug: None
Change-Id: Id4143ca0a565472a4f08905c06f5d3f7d5dfb756
Reviewed-on: https://webrtc-review.googlesource.com/31100
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21151}
2017-12-07 23:06:19 +00:00
4e76ebb372 Rename FOURCC -> CRICKET_FOURCC.
This macro collides with a libjingle macro with the same name.

Bug: webrtc:8589
Change-Id: Icdae45c3550d09a591ee02eaf37e11a44d4230d5
Reviewed-on: https://webrtc-review.googlesource.com/27982
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21029}
2017-12-04 10:31:28 +00:00
aba85d1f53 Resolve circular dependency in rtc_media_base.
This one was pretty straightforward fortunately.

Bug: webrtc:6828
Change-Id: Ie7b5e71f1298c409dbca2c74eaa09c0986e41d8f
Reviewed-on: https://webrtc-review.googlesource.com/25821
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20914}
2017-11-28 15:28:58 +00:00
56d460902e Use the new AudioProcessing statistics everywhere.
The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM.

Bug: webrtc:8563, b/67926135
Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24
Reviewed-on: https://webrtc-review.googlesource.com/25621
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20877}
2017-11-24 18:17:39 +00:00
523589dbd8 Create common helper method for comparing video formats
Unfortunately, H264 makes it non-trivial to compare video formats for
equality. For every video format besides H264 it's enough to look at the
name, but for H264, we need to dig into the parameters. This logic is
currently in several places, and this CL unifies it to one place.

Bug: webrtc:7925
Change-Id: I83a516b108d6b4d6792fd0bf1d24296916d9f7fa
Reviewed-on: https://webrtc-review.googlesource.com/25120
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20853}
2017-11-23 15:41:38 +00:00
c61ce0d0cd Fixing some clang-tidy findings.
Bug: None
Change-Id: I949c1ff35284ce79c99e8f76148f63b8bba965a9
Reviewed-on: https://webrtc-review.googlesource.com/24041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20818}
2017-11-21 16:43:07 +00:00
7880758b90 Optional: Use nullopt and implicit construction in /media
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

TBR=pthatcher@webrtc.org

Bug: None
Change-Id: I6dd8677a65f897877fc848aefa7ab37d844e70ed
Reviewed-on: https://webrtc-review.googlesource.com/23573
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20816}
2017-11-21 14:46:37 +00:00
f2d7beb1d4 Created the DtlsSrtpTransport.
The DtlsSrtpTransport is designed to take DTLS responsibilities from BaseChannel.
DtlsSrtpTransport is responsible for exporting keys from DtlsTransport
and setting up the wrapped SrtpTransport.

The DtlsSrtpTransport is not hooked up to BaseChannel yet in this CL.

Bug: webrtc:7013
Change-Id: I318c00dadf9b1e033ec842de6e1536e9227ab713
Reviewed-on: https://webrtc-review.googlesource.com/6700
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20804}
2017-11-20 23:18:22 +00:00
c3ed630560 Add stats googHasEnteredLowResolution.
Indicates if the forced sw fallback has had an effect (or would have had an effect if it had been
enabled).


Bug: webrtc:6634
Change-Id: I574b9001a2fae650fb894a1caa0d0f84257658e3
Reviewed-on: https://webrtc-review.googlesource.com/23300
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20729}
2017-11-17 13:02:07 +00:00
c97cf03ede Removes unused sample-rate APIs from the ADM.
The following four methods are removed:

SetRecordingSampleRate(const uint32_t samplesPerSec)
RecordingSampleRate(uint32_t* samplesPerSec) const
SetPlayoutSampleRate(const uint32_t samplesPerSec)
PlayoutSampleRate(uint32_t* samplesPerSec) const

Bug: webrtc:7306
Change-Id: I2c3c2e7bd3fb1264da197699fd5de15ab6c35c1b
Reviewed-on: https://webrtc-review.googlesource.com/22001
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20703}
2017-11-16 08:59:53 +00:00
7aee3d538c Fix ortc_api circular deps.
This will help keep ortc dependencies clean in the future, since
gn --check forces us to depend on components from which we include
headers.

cryptoparams.h moves into api/, but ortc appears to think it
should be there anyway. We could consider moving it into the ortc/ api,
but it doesn't appear to be specific to ortc.

Bug: webrtc:6828
Change-Id: Iddae438d10b5e84b2fbc52565364319e20f90613
Reviewed-on: https://webrtc-review.googlesource.com/22660
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20686}
2017-11-15 13:31:51 +00:00
5f5918f4ef Merge MediaChannel's OnTransportOverheadChanged and OnNetworkRouteChanged callbacks.
OnTransportOverChanged is merged into OnNetworkRouteChanged in MediaChannel
because the transport overhead will be added to rtc::NetworkRoute structure.

This CL depends on https://webrtc-review.googlesource.com/c/src/+/13520

Bug: None
Change-Id: I6ed6583f6c91db4ce61a89406de39774239f3a04
Reviewed-on: https://webrtc-review.googlesource.com/15200
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20678}
2017-11-14 20:42:36 +00:00
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros.
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00
e78bcb97c3 Enable cpplint in media/
Bug: webrtc:5584
Change-Id: I2fd1395d35596d9002e19cc90fcda3a5d4cde9e7
Reviewed-on: https://webrtc-review.googlesource.com/16564
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20504}
2017-10-31 17:46:42 +00:00
d79314f9f9 Reland "Add fine grained dropped video frames counters on sending side"
Add fine grained dropped video frames counters on sending side

4 new counters added to SendStatisticsProxy and reported to UMA and logs.

Bug: webrtc:8355
Change-Id: I1f9bdfea9cbf17cf38b3cb2f55d406ffdb06614f
Reviewed-on: https://webrtc-review.googlesource.com/14580
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20421}
2017-10-25 09:32:15 +00:00
c4faa9c4e1 Remove QUIC transport/data channel
Originally, the idea was to implement QUIC data channels as a
PeerConnection API. Now, the effort has shifted to implementing it as a
part of ORTC which will live in Chromium. Since this code has not been
maintained and is not currently being used, remove it to reduce
maintenance overhead while a copy will be retained in the Git history.

Bug: webrtc:8385
Change-Id: I2719c007a0de0118b67d41a425f900b66c52f65a
Reviewed-on: https://webrtc-review.googlesource.com/14100
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20412}
2017-10-24 16:14:18 +00:00
a0b66c7566 Enable cpu adaptation for screenshare in chrome as field trial
Bug: webrtc:8433
Change-Id: I1632cb0191b06e99d090a090bad9db20d7a81349
Reviewed-on: https://webrtc-review.googlesource.com/14980
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20411}
2017-10-24 15:07:38 +00:00
1c1a6815ae Revert "Add fine grained dropped video frames counters on sending side"
This reverts commit 4b1a363e4c238f2e1ec2d8a9ce1f819f59d710ce.

Reason for revert: Breaks dependent android projects.

Original change's description:
> Add fine grained dropped video frames counters on sending side
> 
> 4 new counters added to SendStatisticsProxy and reported to UMA and logs.
> 
> Bug: webrtc:8355
> Change-Id: Idf9b8dfc295c92821e058a97cb3894dc6a446082
> Reviewed-on: https://webrtc-review.googlesource.com/12260
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20347}

TBR=deadbeef@webrtc.org,ilnik@webrtc.org,sprang@webrtc.org,stefan@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8355
Change-Id: I59b02f4eb77abad7ff1fbcbfa61844918c95d723
Reviewed-on: https://webrtc-review.googlesource.com/14500
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20378}
2017-10-21 09:23:54 +00:00
8d3444df2d Reland "Rewrite WebRtcSession media tests as PeerConnection tests"
This is a reland of 3df5dcac9b339ba4d3f4969602f094c2c8035b51
Original change's description:
> Rewrite WebRtcSession media tests as PeerConnection tests
> 
> Bug: webrtc:8222
> Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad
> Reviewed-on: https://webrtc-review.googlesource.com/6640
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20364}

Bug: webrtc:8222
Change-Id: I0a5398170d469eb9223bc781bfb417a85a72a2d2
Reviewed-on: https://webrtc-review.googlesource.com/14380
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20377}
2017-10-21 01:38:14 +00:00
f2662f08e5 Revert "Rewrite WebRtcSession media tests as PeerConnection tests"
This reverts commit 3df5dcac9b339ba4d3f4969602f094c2c8035b51.

Reason for revert: suspected of breaking chromium.webrtc.fyi:
WebRtcBrowserTest.NegotiateUnsupportedVideoCodec
WebRtcBrowserTest.NegotiateNonCryptoCall

android https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/builds/25506
linux https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Linux%20Tester/builds/38809
mac
https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Mac%20Tester/builds/44120
windows
https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win10%20Tester/builds/9236

Original change's description:
> Rewrite WebRtcSession media tests as PeerConnection tests
> 
> Bug: webrtc:8222
> Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad
> Reviewed-on: https://webrtc-review.googlesource.com/6640
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20364}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org

Change-Id: Iaacc950d050ba2835d262908658dc045f234ef5b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8222
Reviewed-on: https://webrtc-review.googlesource.com/14160
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20375}
2017-10-20 13:00:45 +00:00
3df5dcac9b Rewrite WebRtcSession media tests as PeerConnection tests
Bug: webrtc:8222
Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad
Reviewed-on: https://webrtc-review.googlesource.com/6640
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20364}
2017-10-20 01:52:23 +00:00
4b1a363e4c Add fine grained dropped video frames counters on sending side
4 new counters added to SendStatisticsProxy and reported to UMA and logs.

Bug: webrtc:8355
Change-Id: Idf9b8dfc295c92821e058a97cb3894dc6a446082
Reviewed-on: https://webrtc-review.googlesource.com/12260
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20347}
2017-10-19 10:37:12 +00:00
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
032f410ae0 Delete unneeded includes of pathutils.h
Bug: webrtc:6424
Change-Id: I73b2bc747c67d2fe2ad888dde9c2815a6d9aceaa
Reviewed-on: https://webrtc-review.googlesource.com/4760
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20059}
2017-10-02 07:05:19 +00:00
024d8970a7 Add conversion from webrtc::SdpVideoFormat to cricket::VideoCodec
We will have to convert from webrtc::SdpVideoFormat to
cricket::VideoCodec in a couple of places until
cricket::WebRtcVideoEncoderFactory is gone. It will be convenient to
have the conversion logic in a common place.

Bug: webrtc:7925
Change-Id: Ie5e88599f28aeea647e936300c04f9071daffd53
Reviewed-on: https://webrtc-review.googlesource.com/4840
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20048}
2017-09-29 14:45:57 +00:00
d8970dbd42 Delete unneeded includes of fileutils.h
It is now used only by FileRotatingStream.

Bug: webrtc:6424
Change-Id: I216b20baadae836d24c39899efe4cb45c2935f41
Reviewed-on: https://webrtc-review.googlesource.com/4720
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20040}
2017-09-29 12:39:09 +00:00
244ad80444 Clean up some bad constructs in media/
We currently suppress warnings for bad constructs in media/. Still, the
warnings are causing problems when trying to include header files from
this directory. This CL cleans up some of the bad constructs.

Bug: None
Change-Id: I808ad39eb23870d20fa5bb05429b50c9078543ae
Reviewed-on: https://webrtc-review.googlesource.com/4541
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20039}
2017-09-29 12:22:57 +00:00
02e7a1981a Remove unnecessary video factory references in PeerConnectionFactory
The video codec factories should be owned by the video engine instead
of by the PeerConnectionFactory.

Bug: None
Change-Id: If63d47cef565138d51377af3fc9ea973950c9390
Reviewed-on: https://webrtc-review.googlesource.com/1601
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20002}
2017-09-27 14:41:46 +00:00
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
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
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
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