Commit Graph

13633 Commits

Author SHA1 Message Date
41514718a8 Add usage description strings to Info.plist
These are required for apps that use the camera and microphone on iOS 10

BUG=webrtc:6428
R=magjed@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/2372203002 .

Cr-Commit-Position: refs/heads/master@{#14401}
2016-09-27 16:02:57 +00:00
efc6e41866 Revert of Unify rtcp packet setters (patchset #8 id:130001 of https://codereview.webrtc.org/2348623003/ )
Reason for revert:
Breaks compilation of internal downstream project.

Original issue's description:
> Unify rtcp packet setters
> Renamed setters in rtcp classes
> from WithField to SetField
> from WithItem to AddItem or SetItems
> from From to SetSenderSsrc
> from To to SetMediaSsrc
> Some redundant or unsued setters removed.
> Pass-by-const& replaced with pass-by-value when appropriate.
>
> BUG=webrtc:5260
>
> Committed: https://crrev.com/20e77c7b8a9f19942ef3c3c4f1fa3888b2cd54ea
> Cr-Commit-Position: refs/heads/master@{#14393}

TBR=sprang@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2372713005
Cr-Commit-Position: refs/heads/master@{#14400}
2016-09-27 15:39:39 +00:00
9532124659 RTCPReceiver store cname as std::string.
simplifying cname management.

Remove RTCPUtility::RTCPCnameInformation
since it was last use of the structure.

BUG=webrtc:5565
NOTRY=true

Review-Url: https://codereview.webrtc.org/2354333004
Cr-Commit-Position: refs/heads/master@{#14399}
2016-09-27 14:05:39 +00:00
f1363fdf57 Adds support for AVAudioSessionSilenceSecondaryAudioHintNotification on iOS
BUG=b/30944297
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2366753005
Cr-Commit-Position: refs/heads/master@{#14398}
2016-09-27 13:06:48 +00:00
46a8d18efa ACM: Removed the code for InitialDelayManager
It looks to have been unused since the landing of
https://codereview.webrtc.org/1419573013

BUG=webrtc:3520

Review-Url: https://codereview.webrtc.org/2363993002
Cr-Commit-Position: refs/heads/master@{#14397}
2016-09-27 12:43:37 +00:00
29a44e351e This is a resubmission of https://codereview.webrtc.org/2047513002/
Original description:
Add proper lifetime of encoder-specific settings.

Permits passing VideoEncoderConfig between threads and not worry about
the lifetime of an underlying void pointer. Also adds type safety to
unpacking of codec-specific settings.

These settings are not yet propagating to VideoEncoder interfaces, but
the aim is to get rid of webrtc::VideoCodec for VideoEncoder.

BUG=webrtc:3424
R=perkj@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/2347843002
Cr-Commit-Position: refs/heads/master@{#14396}
2016-09-27 10:52:05 +00:00
5f8ebaeffd Add limitations of number of frames that can be created in I420BufferPool::CreateBuffer.
If more than 60 frames are created and not returned, the implementation will crash.

I420BufferPool are currently used by the VP8 decoder, Quality scaler and VideoFrameFactory.

BUG=b/31390397
NOTRY=true  // Because of failing gclient runhooks on some bots

Review-Url: https://codereview.webrtc.org/2370653003
Cr-Commit-Position: refs/heads/master@{#14395}
2016-09-27 10:47:48 +00:00
c8299f9f87 Posting Opus's set-force-channels functionality to WebRTC.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2352713005
Cr-Commit-Position: refs/heads/master@{#14394}
2016-09-27 09:08:54 +00:00
20e77c7b8a Unify rtcp packet setters
Renamed setters in rtcp classes
from WithField to SetField
from WithItem to AddItem or SetItems
from From to SetSenderSsrc
from To to SetMediaSsrc
Some redundant or unsued setters removed.
Pass-by-const& replaced with pass-by-value when appropriate.

BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2348623003
Cr-Commit-Position: refs/heads/master@{#14393}
2016-09-27 08:37:51 +00:00
4ecd9700ee GN: Fix incorrect include_dir for video_coding on iOS
When rtc_build_libyuv=false an incorrect code path
is surfaced in GN.

BUG=webrtc:6412
NOTRY=True
TESTED=gn gen out/foo --args='rtc_build_libyuv=false target_os="ios"'

Review-Url: https://codereview.webrtc.org/2375603002
Cr-Commit-Position: refs/heads/master@{#14392}
2016-09-27 08:11:24 +00:00
c1815cf084 Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
Reason for revert:
Internal project is updated.

Original issue's description:
> Revert of Rename AppRTCDemo on Android and iOS to AppRTCMobile (patchset #2 id:20001 of https://codereview.webrtc.org/2343403002/ )
>
> Reason for revert:
> Breaks internal project.
>
> Original issue's description:
> > Rename AppRTCDemo on Android and iOS to AppRTCMobile
> >
> > The purpose is to make it clearer it is a mobile application.
> >
> > BUG=webrtc:6359
> > NOPRESUBMIT=true
> >
> > Committed: https://crrev.com/d3af58bdab5b25acd62cd816363becc7003d3e5a
> > Cr-Commit-Position: refs/heads/master@{#14356}
>
> TBR=sakal@webrtc.org,kthelgason@webrtc.org,tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6359
>
> Committed: https://crrev.com/87ef6f750126f9f17f4714d696a8e77a2dd0a3f1
> Cr-Commit-Position: refs/heads/master@{#14358}

TBR=sakal@webrtc.org,kthelgason@webrtc.org,tommi@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6359

Review URL: https://codereview.webrtc.org/2373443005 .

Cr-Commit-Position: refs/heads/master@{#14391}
2016-09-27 08:10:52 +00:00
0a52c7003d THis CL enables possibility to select full-duplex OpenSL ES audio in AppRTCDemo, i.e., it adds support for OpenSL ES for input as well. The user must explicitly select this new mode in the debug UI hence it is not the default selection. There is no separate UI for input and output; instead both are enabled/disabled by the same switch.
BUG=webrtc:5925
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2366383002 .

Cr-Commit-Position: refs/heads/master@{#14390}
2016-09-27 07:35:37 +00:00
64ec8f826f Reland of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #1 id:1 of https://codereview.webrtc.org/2354223002/ )
Reason for revert:
Downstream application now fixed.

Original issue's description:
> Revert of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #14 id:260001 of https://codereview.webrtc.org/2278883002/ )
>
> Reason for revert:
> Broke downstream application.
>
> Original issue's description:
> > Move MutableDataY{,U,V} methods to I420Buffer only.
> >
> > Deleted from the VideoFrameBuffer base class.
> >
> > BUG=webrtc:5921
> >
> > Committed: https://crrev.com/5539ef6c03c273f39fadae41ace47fdc11ac6d60
> > Cr-Commit-Position: refs/heads/master@{#14317}
>
> TBR=perkj@webrtc.org,magjed@webrtc.org,pthatcher@webrtc.org,honghaiz@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:5921
>
> Committed: https://crrev.com/776870a2599b8f43ad56987f9031690e3ccecde8
> Cr-Commit-Position: refs/heads/master@{#14325}

TBR=perkj@webrtc.org,magjed@webrtc.org,pthatcher@webrtc.org,honghaiz@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5921

Review-Url: https://codereview.webrtc.org/2372483002
Cr-Commit-Position: refs/heads/master@{#14389}
2016-09-27 07:17:40 +00:00
c637389949 Delete unused file mock_audio_vector.h.
BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2367323002
Cr-Commit-Position: refs/heads/master@{#14388}
2016-09-27 06:29:57 +00:00
de2920cb46 Delete unused file sessionid.h.
BUG=None.

Review-Url: https://codereview.webrtc.org/2370723002
Cr-Commit-Position: refs/heads/master@{#14387}
2016-09-27 06:28:51 +00:00
89175a606e Trust that calls to RemoteEstimatorProxy::Process are done at the right frequency.
BUG=None

Review-Url: https://codereview.webrtc.org/2365293002
Cr-Commit-Position: refs/heads/master@{#14386}
2016-09-26 18:56:03 +00:00
25337bb5ab Android: Update clang-format to follow Google style guide
BUG=webrtc:6419
NOTRY=True

Review-Url: https://codereview.webrtc.org/2368963002
Cr-Commit-Position: refs/heads/master@{#14385}
2016-09-26 18:47:53 +00:00
fd8e33d3ad Removing a useless ctor in AudioNetworkAdaptorImpl.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2367743002
Cr-Commit-Position: refs/heads/master@{#14384}
2016-09-26 18:46:45 +00:00
8af4fd0128 Disabled flaky VideoSendStreamTest.ChangingNetworkRoute
BUG=webrtc:6422
NOTRY=True
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2372553002
Cr-Commit-Position: refs/heads/master@{#14383}
2016-09-26 18:45:44 +00:00
98088dcccc header_usage.sh script: Exclude matches in gyp and gn files.
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2363423002
Cr-Commit-Position: refs/heads/master@{#14382}
2016-09-26 18:44:36 +00:00
660312b0c6 Enable //build/config/clang:extra_warnings for rtc_media
The code compiles fine with this config removed.

BUG=webrtc:6323
NOTRY=True
NOTREECHECKS=True

Review-Url: https://codereview.webrtc.org/2369893002
Cr-Commit-Position: refs/heads/master@{#14381}
2016-09-26 13:11:57 +00:00
464382da71 Remove duplicated entry for bwe_simulations.cc
Since modules_unittests already depends on
remote_bitrate_estimator:bwe_simulator and the bwe_simulations.cc
source was added to that target in https://codereview.webrtc.org/2296253002
there's no point having it added here.

BUG=webrtc:6323
NOTRY=True
NOPRESUBMIT=True
NOTREECHECKS=True

Review-Url: https://codereview.webrtc.org/2368933002
Cr-Commit-Position: refs/heads/master@{#14380}
2016-09-26 10:00:09 +00:00
3901128075 Remove unnecessary jsoncpp includes.
These are already added in third_party/jsoncpp/BUILD.gn so they
shouldn't be needed.

BUG=webrtc:6412
NOTRY=True
NOTREECHECKS=True

Review-Url: https://codereview.webrtc.org/2368543002
Cr-Commit-Position: refs/heads/master@{#14379}
2016-09-26 09:52:40 +00:00
20684110fd r14326 added '-Wno-unused-result' to 'WARNING_CFLAGS!' which removes the
flag. This change moves it to WARNING_CFLAGS, and makes it work for both
iOS and mac.

BUG=webrtc:6396
NOTRY=True
NOPRESUBMIT=True
NOTREECHECKS=True

Review-Url: https://codereview.webrtc.org/2369803002
Cr-Commit-Position: refs/heads/master@{#14378}
2016-09-26 09:41:40 +00:00
c59bf0415a Remove differ from ScreenCapturer implementations
We can use ScreenCapturerDifferWrapper if needed, otherwise ScreenCapturer does
not need to calculate updated region itself, setting to entire screen is enough.

BUG=633802

Review-Url: https://codereview.webrtc.org/2348803003
Cr-Commit-Position: refs/heads/master@{#14377}
2016-09-24 00:54:40 +00:00
d3d230f788 - Make RtpSenderAudio not inherit from DtmfQueue.
- Remove unused method DtmfQueue::ResetDTMF()

BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2365873002
Cr-Commit-Position: refs/heads/master@{#14376}
2016-09-23 20:10:50 +00:00
92ea601e90 Move class RTCPHelp::RTCPPacketInformation into RTCPReceiver
Use it by pointer instead of by reference.
Renamed PacketInformation members to follow style,
Unused members removed.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2366563002
Cr-Commit-Position: refs/heads/master@{#14375}
2016-09-23 17:36:12 +00:00
dda366611e Fixes minor issue in AudioDeviceTest.RunPlayoutAndRecordingInFullDuplex for iOS.
Followup on https://codereview.webrtc.org/2349263004/

BUG=NONE
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2362263002
Cr-Commit-Position: refs/heads/master@{#14374}
2016-09-23 15:42:49 +00:00
Per
f8c5f2b485 Fix vie_encoder_unittest.cc.
This was broken in https://codereview.webrtc.org/2338133003/ Let ViEEncoder tell VideoSendStream about reconfigurations when I manually landed that cl without rebasing.
Shame on me.

BUG=webrtc:5687, webrtc:6371
TBR=mflodman@webrtc.org
NOTREECHECKS=true

Review URL: https://codereview.webrtc.org/2359153004 .

Cr-Commit-Position: refs/heads/master@{#14373}
2016-09-23 14:25:10 +00:00
44428a8aa6 iOS: Always build H264 HW encoder/decoder
This CL removes the use_objc_h264 flag. This means that the VideoToolbox
H264 encoder and decoder will always be built.

BUG=webrtc:4081
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2366443003
Cr-Commit-Position: refs/heads/master@{#14372}
2016-09-23 14:01:44 +00:00
Per
512ecb3206 Let ViEEncoder tell VideoSendStream about reconfigurations.
This cl change so that all encoder configuration changes are reported to VideoSendStream through the ViEEncoder.
Also, the PayLoadRouter is changed to never stop sending on a an ssrc due to the encoder video frame size changes. Instead, the number of sending streams is only decided by the number of sending ssrc.

This cl is a preparation for moving encoder reconfiguration due to input video frame size changes from WebRtcVideoSendStream to ViEEncoder.

BUG=webrtc:5687, webrtc:6371
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/2338133003 .

Cr-Commit-Position: refs/heads/master@{#14371}
2016-09-23 13:52:20 +00:00
2a27b0a478 Delete unused class FakeScreenCapturerFactory.
Last use of the header file probably deleted in cl
https://codereview.webrtc.org/1903393004

BUG=None

Review-Url: https://codereview.webrtc.org/2353043003
Cr-Commit-Position: refs/heads/master@{#14370}
2016-09-23 11:42:49 +00:00
347ec5c72e Change thread check to race check. Also, add comment to explain implementation of RaceChecker.
BUG=webrtc:6345

Review-Url: https://codereview.webrtc.org/2350663002
Cr-Commit-Position: refs/heads/master@{#14369}
2016-09-23 11:21:55 +00:00
f1b08da5b4 Stopped using the NetEqDecoder enum internally in NetEq.
NetEqDecoder is still used in the external interfaces, but this change
opens up the ability to use SdpAudioFormats directly, once appropriate
interfaces have been added.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2355503002
Cr-Commit-Position: refs/heads/master@{#14368}
2016-09-23 09:19:49 +00:00
1490f7aa55 Add histogram for end-to-end delay:
"WebRTC.Video.EndToEndDelayInMs"

Make capture time in local timebase available for decoded VP9 video frames (propagate ntp_time_ms from EncodedImage to decoded VideoFrame).

BUG=webrtc:6409

Review-Url: https://codereview.webrtc.org/1905563002
Cr-Commit-Position: refs/heads/master@{#14367}
2016-09-23 09:09:59 +00:00
6d4c8c307e Renaming a proto target in GYP for audio network adaptor.
It was incorrectly named for GYP in https://codereview.webrtc.org/2365723002
This makes the target name be the same for GN and GYP.

BUG=webrtc:6303
NOTRY=True

Review-Url: https://codereview.webrtc.org/2366883002
Cr-Commit-Position: refs/heads/master@{#14366}
2016-09-23 08:42:22 +00:00
e87d6734ea Return texture frame when dropping frames in CameraCapturer.
BUG=webrtc:6411,b/31686979

Review-Url: https://codereview.webrtc.org/2363043002
Cr-Commit-Position: refs/heads/master@{#14365}
2016-09-23 08:27:38 +00:00
b62dbbe985 GN: Change rtc_source_set targets --> rtc_static_library
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).

After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()

See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.

NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.

BUG=webrtc:6410, chromium:630755

Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 07:38:58 +00:00
25f6a39181 Relanding of "Adding debug dump to audio network adaptor."
The original CL was https://codereview.webrtc.org/2356763002

but got reverted https://codereview.webrtc.org/2362003002/.

The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests.

BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2365723002
Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-23 05:23:28 +00:00
161b3907ab Revert of Adding debug dump to audio network adaptor. (patchset #5 id:140001 of https://codereview.webrtc.org/2356763002/ )
Reason for revert:
Chromium bot fails

Original issue's description:
> Adding debug dump to audio network adaptor.
>
> BUG=webrtc:6303
>
> Committed: https://crrev.com/7e4f8928062afc8d571bb69f3223711701cbaad6
> Cr-Commit-Position: refs/heads/master@{#14361}

TBR=michaelt@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2362003002
Cr-Commit-Position: refs/heads/master@{#14362}
2016-09-22 21:17:01 +00:00
7e4f892806 Adding debug dump to audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2356763002
Cr-Commit-Position: refs/heads/master@{#14361}
2016-09-22 20:39:18 +00:00
a78213e4d5 Add tools/determinism to setup_links.
BUG=chromium:583318
NOTRY=True

Review-Url: https://codereview.webrtc.org/2360853003
Cr-Commit-Position: refs/heads/master@{#14360}
2016-09-22 17:46:19 +00:00
893a7eeecb Support more QCOM specific color formats for Android HW decoder.
BUG=b/31483393

Review-Url: https://codereview.webrtc.org/2349843002
Cr-Commit-Position: refs/heads/master@{#14359}
2016-09-22 17:44:34 +00:00
87ef6f7501 Revert of Rename AppRTCDemo on Android and iOS to AppRTCMobile (patchset #2 id:20001 of https://codereview.webrtc.org/2343403002/ )
Reason for revert:
Breaks internal project.

Original issue's description:
> Rename AppRTCDemo on Android and iOS to AppRTCMobile
>
> The purpose is to make it clearer it is a mobile application.
>
> BUG=webrtc:6359
> NOPRESUBMIT=true
>
> Committed: https://crrev.com/d3af58bdab5b25acd62cd816363becc7003d3e5a
> Cr-Commit-Position: refs/heads/master@{#14356}

TBR=sakal@webrtc.org,kthelgason@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6359

Review-Url: https://codereview.webrtc.org/2358133003
Cr-Commit-Position: refs/heads/master@{#14358}
2016-09-22 17:15:43 +00:00
3e02430587 Fix a stun attribute leak.
In https://cs.chromium.org/chromium/src/third_party/webrtc/p2p/base/stun.cc?rcl=1474384719&l=352,
if read returned false, the created attr would not be released.

BUG=chromium:648064
R=skvlad@webrtc.org

Review URL: https://codereview.webrtc.org/2357733002 .

Cr-Commit-Position: refs/heads/master@{#14357}
2016-09-22 16:52:30 +00:00
d3af58bdab Rename AppRTCDemo on Android and iOS to AppRTCMobile
The purpose is to make it clearer it is a mobile application.

BUG=webrtc:6359
NOPRESUBMIT=true

Review-Url: https://codereview.webrtc.org/2343403002
Cr-Commit-Position: refs/heads/master@{#14356}
2016-09-22 16:12:38 +00:00
051d151569 Adds audio session status to logs for each valid audio route change on iOS
BUG=b/30944297
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2355393005
Cr-Commit-Position: refs/heads/master@{#14355}
2016-09-22 15:48:10 +00:00
c37e9835a7 Add custom info.plist to modules_unittests
This is to fix an issue introduced with iOS 10 where all applications that access the microphone have to include a string in the Info.plist file explaining why they need it.

BUG=webrtc:6403

Review-Url: https://codereview.webrtc.org/2359863003
Cr-Commit-Position: refs/heads/master@{#14354}
2016-09-22 15:00:57 +00:00
f292e31511 Relax too strict DCHECKs while parsing rtcp reports
BUG=chromium:649129

Review-Url: https://codereview.webrtc.org/2361493004
Cr-Commit-Position: refs/heads/master@{#14353}
2016-09-22 14:24:38 +00:00
aac9d6fb25 Added graph for plotting the audio level from an Rtc event log.
This uses the audio level values that are stored in the RTP header extension.

BUG=webrtc:4741

Review-Url: https://codereview.webrtc.org/2346363003
Cr-Commit-Position: refs/heads/master@{#14352}
2016-09-22 14:01:55 +00:00