Commit Graph

4592 Commits

Author SHA1 Message Date
b6760f9e44 Format all Java in WebRTC.
BUG=webrtc:6419
TBR=henrika@webrtc.org

Review-Url: https://codereview.webrtc.org/2377003002
Cr-Commit-Position: refs/heads/master@{#14432}
2016-09-29 11:12:51 +00:00
Per
a48ddb7636 Add VideoSendStream::Stats::prefered_media_bitrate_bps
This cl move calculation of stats for prefered_media_bitrate_bps from webrtcvideoengine2.GetStats to SendStatisticsProxy::OnEncoderReconfigured.
This aligns better with how other send stats are reported and is needed as a prerequisite for moving video encoder configuration due to video resolution change
from WebRtcVideoEngine2 to ViEEncoder.

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

Review URL: https://codereview.webrtc.org/2368223002 .

Cr-Commit-Position: refs/heads/master@{#14431}
2016-09-29 09:49:01 +00:00
fd0d426692 Fix race / crash in OnNetworkRouteChanged().
To achieve this some refactoring was done to make it possible to synchronize
access to the DelayBasedBwe in TransportFeedbackAdapter:
- The callback was removed from DelayBasedBwe, it now instead returns its
  result.
- TransportFeedbackAdapter was moved to modules/congestion_controller to avoid
  unnecessary dependencies.

Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds.

BUG=webrtc:6427, webrtc:6422

Review-Url: https://codereview.webrtc.org/2366333003
Cr-Commit-Position: refs/heads/master@{#14430}
2016-09-29 09:44:38 +00:00
eddb7571d8 Revert of Unify the macOS and iOS capturer implementations (patchset #4 id:60001 of https://codereview.webrtc.org/2309253005/ )
Reason for revert:
Breaks internal project

Original issue's description:
> Unify the macOS and iOS capturer implementations
>
> This removes the QTKit based capturer for mac, and removes the need
> to link against deprecated system libraries on macOS.
>
> BUG=webrtc:3968,webrtc:6275,webrtc:6333
>
> Committed: https://crrev.com/242d8bdddd77109781cbb70c59d161be7566ac98
> Cr-Commit-Position: refs/heads/master@{#14418}

TBR=magjed@webrtc.org,tkchin@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:3968,webrtc:6275,webrtc:6333

Review-Url: https://codereview.webrtc.org/2381853002
Cr-Commit-Position: refs/heads/master@{#14429}
2016-09-29 09:43:27 +00:00
77eab70470 Enable the -Wundef warning for clang
NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2358993004
Cr-Commit-Position: refs/heads/master@{#14425}
2016-09-29 00:42:08 +00:00
24c7c1238d Move FunctionView from AudioCodingModule to the rtc namespace
It's a very general type, and we're about to start needing it in other
places besides AudioCodingModule.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2380463003
Cr-Commit-Position: refs/heads/master@{#14423}
2016-09-28 18:57:17 +00:00
478681e1e6 Move the QP scaling thresholds to the relevant encoders.
Also provide a new set of thresholds for the VideoToolbox encoder. The new thresholds were experimentally determined to work well on the iPhone 6S, and also adequately on the iPhone 5S.

BUG=webrtc:5678

Review-Url: https://codereview.webrtc.org/2309743002
Cr-Commit-Position: refs/heads/master@{#14420}
2016-09-28 15:17:51 +00:00
e75f204b06 Expose Ivf logging through the native API
BUG=webrtc:6300

Review-Url: https://codereview.webrtc.org/2303273002
Cr-Commit-Position: refs/heads/master@{#14419}
2016-09-28 13:19:53 +00:00
242d8bdddd Unify the macOS and iOS capturer implementations
This removes the QTKit based capturer for mac, and removes the need
to link against deprecated system libraries on macOS.

BUG=webrtc:3968,webrtc:6275,webrtc:6333

Review-Url: https://codereview.webrtc.org/2309253005
Cr-Commit-Position: refs/heads/master@{#14418}
2016-09-28 12:51:44 +00:00
e5684c5387 Delete method webrtc::VideoFrame::allocated_size and enum PlaneType.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2380623002
Cr-Commit-Position: refs/heads/master@{#14416}
2016-09-28 10:14:15 +00:00
798896a4aa Replace RtcpReceiveTimeInfo with rtcp::ReceiveTimeInfo
structs are exactly the same but last one follow naming style.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2368983002
Cr-Commit-Position: refs/heads/master@{#14415}
2016-09-28 09:54:30 +00:00
e0b2f15417 Frame continuity is now tested as soon as a frame is inserted into the FrameBuffer.
Since we want to stop sending NACKS for frames not needed to keep the stream
decodable we must know which frames that are continuous or not.

BUG=webrtc:5514
R=danilchap@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2322263002 .

Cr-Commit-Position: refs/heads/master@{#14412}
2016-09-28 08:24:02 +00:00
89a3a1a363 Moved Gn target rtc_event_log to one directory above.
This is done to ensure GN targets are placed in the same directory as of the source files.

BUG=webrtc:6412
NOTRY=True

Review-Url: https://codereview.webrtc.org/2365383004
Cr-Commit-Position: refs/heads/master@{#14411}
2016-09-28 07:49:04 +00:00
e5e632f873 Hooking up target audio bitrate to audio network adaptor.
After the landing of BitrateController, it is time to hook up the network data (target_audio_bitrate_bps) required by BitrateController.

BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2364473005
Cr-Commit-Position: refs/heads/master@{#14406}
2016-09-27 19:54:27 +00:00
822a16f64c Reland of Unify rtcp packet setters (patchset #1 id:1 of https://codereview.webrtc.org/2372713005/ )
Reason for revert:
Fix backward compatibility support

Original issue's description:
> 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
>
> Committed: https://crrev.com/efc6e41866662e0922858fbce1d9ee3bdd0637ed
> Cr-Commit-Position: refs/heads/master@{#14400}

TBR=sprang@webrtc.org,stefan@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:5260

Review-Url: https://codereview.webrtc.org/2370313002
Cr-Commit-Position: refs/heads/master@{#14402}
2016-09-27 16:27:52 +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
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
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
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
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
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
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
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
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
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
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
d0ede4493e Adding FecController to audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2337103006
Cr-Commit-Position: refs/heads/master@{#14351}
2016-09-22 13:20:59 +00:00
799a9d017a Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ )
Reason for revert:
breaks downstream code

Original issue's description:
> Remove unnecessary interface TelephoneEventHandler.
>
> BUG=webrtc:2795
>
> Committed: https://crrev.com/2beb42983ca24e1326a9a7f2c06b3ad740eea2c3
> Cr-Commit-Position: refs/heads/master@{#14346}

TBR=henrik.lundin@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2362673002
Cr-Commit-Position: refs/heads/master@{#14348}
2016-09-22 10:36:34 +00:00
a70695a3e1 Moved Opus-specific payload splitting into AudioDecoderOpus.
The biggest change to NetEq is the move from a primary flag, to a
Priority with two separate levels: one set by RED splitting and one
set by the codec itself. This allows us to unambigously prioritize
"fallback" packets from these two sources. I've chosen what I believe
is the sensible ordering: packets that the codec prioritizes are
chosen first, regardless of if they are secondary RED packets or
not. So if we were to use Opus w/ FEC in RED, we'd only do Opus FEC
decoding if there was no RED packet that could cover the time slot.

With this change, PayloadSplitter now only deals with RED
packets. Maybe it should be renamed RedPayloadSplitter?

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2342443005
Cr-Commit-Position: refs/heads/master@{#14347}
2016-09-22 09:07:03 +00:00
2beb42983c Remove unnecessary interface TelephoneEventHandler.
BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2357583002
Cr-Commit-Position: refs/heads/master@{#14346}
2016-09-22 08:46:08 +00:00
bc77ed7657 Adding reordering logic in audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2349113002
Cr-Commit-Position: refs/heads/master@{#14344}
2016-09-22 07:45:23 +00:00
4aec1d4437 Relanding of "Adding BitrateController to audio network adaptor."
Adding BitrateController to audio network adaptor was first landed in https://codereview.webrtc.org/2334613002/ but reverted in https://codereview.webrtc.org/2352223002/ due to ODR violation.

This CL tries to use namespace trick to solve the ODR problem.

BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2353293002
Cr-Commit-Position: refs/heads/master@{#14343}
2016-09-22 06:01:34 +00:00