803 Commits

Author SHA1 Message Date
solenberg
566ef247b9 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1403363003

Cr-Commit-Position: refs/heads/master@{#10548}
2015-11-06 23:34:58 +00:00
Fredrik Solenberg
0ccae13556 Changed FakeVoiceEngine into a MockVoiceEngine.
BUG=webrtc:4690
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1402403008 .

Cr-Commit-Position: refs/heads/master@{#10491}
2015-11-03 09:15:59 +00:00
kwiberg
102c6a61bc Replace rtc:🦗:Settable with rtc::Maybe
The former is very similar to the latter, but less general (mostly in
naming).

This CL, which is the first to use Maybe at scale, also removes the implicit conversion from T to Maybe<T>, since it was agreed that the increased verbosity increased legibility.

Review URL: https://codereview.webrtc.org/1430433004

Cr-Commit-Position: refs/heads/master@{#10461}
2015-10-30 09:47:44 +00:00
Stefan Holmer
bbaf3633c5 Filter overlapping RTP header extensions.
This removes unnecessary RTP header extension overhead since only one of
these extensions is used at a time.

BUG=webrtc:4254
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1429753003 .

Cr-Commit-Position: refs/heads/master@{#10455}
2015-10-29 17:53:33 +00:00
Henrik Kjellander
98f53510b2 system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1413333002 .

Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
rlester
ec9d187f70 Added override keyword to overridden methods to stop compiler warnings.
BUG=

Review URL: https://codereview.webrtc.org/1417543002

Cr-Commit-Position: refs/heads/master@{#10433}
2015-10-27 21:22:21 +00:00
solenberg
85a0496b8c Implement AudioSendStream::GetStats().
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1414743004

Cr-Commit-Position: refs/heads/master@{#10424}
2015-10-27 10:35:30 +00:00
pbos
4cba4eba59 Disable denoising for VP9 by default.
BUG=webrtc:5108
R=marpan@webrtc.org

Review URL: https://codereview.webrtc.org/1418133012

Cr-Commit-Position: refs/heads/master@{#10413}
2015-10-26 18:18:24 +00:00
noahric
5d9b92b53d Update Bind to match its comments and always capture by value. Also update the generated count to 9 args.
The existing comment is wrong, and the test even ensures it: Bind will capture reference values by reference. That makes it hard to use with AsyncInvoker, because you can't safely Bind to a function that takes (const) reference params.

The new version of this code strips references in the bound object, so it captures by value, but can bind against functions that take const references, they'll just be references to the copy.

As the class comment implies, actual by-reference args should be passed as pointers or things that safely share (e.g. scoped_refptr) and not references directly. A new test case ensures the pointer reference works. The new code will also give a compiler error if you try to bind
to a non-const reference.

BUG=

Review URL: https://codereview.webrtc.org/1291543006

Cr-Commit-Position: refs/heads/master@{#10397}
2015-10-24 18:14:52 +00:00
tfarina
ff134ebd3d talk: Use NDEBUG macro.
NDEBUG is a standard macro with the semantic "Not Debug" for C89, C99, C++98,
  C++2003, C++2011, C++2014 standards. There are no _DEBUG macros in the
  standards.

_DEBUG is a macro Visual Studio defines when you specify the /MTd or /MDd
option.

http://stackoverflow.com/a/29253284/5237416

This should help fix the TODO in third_party/libjingle/libjingle.gyp

BUG=None
R=sergeyu@chromium.org

Review URL: https://codereview.webrtc.org/1419733004

Cr-Commit-Position: refs/heads/master@{#10377}
2015-10-22 23:15:25 +00:00
ivoc
797ef12324 Added StopAecDump function to PeerConnectionFactory.
The function to stop recording an AEC dump was missing from the PeerConnectionFactory interface (only a start function was provided). This CL adds the missing stop function.

BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1415733005

Cr-Commit-Position: refs/heads/master@{#10372}
2015-10-22 10:25:45 +00:00
Fredrik Solenberg
4f4ec0a927 Re-Land: Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
BUG=webrtc:4690

Committed: a457752f4a

Review URL: https://codereview.webrtc.org/1390753002 .

Cr-Commit-Position: refs/heads/master@{#10369}
2015-10-22 08:49:39 +00:00
solenberg
c96df779b0 - Introduce internal classes WebRtcAudio[Send|Receive]Stream in WebRtcVoiceMediaChannel.
- Remove WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
- Create webrtc::AudioSendStreams.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1415563003

Cr-Commit-Position: refs/heads/master@{#10361}
2015-10-21 20:02:00 +00:00
Peter Boström
dfa2815b4f Update receive report SSRCs on RemoveSendStream.
Prevents RTCP receiver reports, including PLIs with an old
receiver-report SSRC, from being dropped from the remote sender's
BundleFilter due to no longer being in use.

BUG=chromium:523928, webrtc:4883
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1404363003 .

Cr-Commit-Position: refs/heads/master@{#10359}
2015-10-21 15:21:21 +00:00
Peter Boström
b64a32bf25 Remove old VideoFrame::Reset.
Hopefully all external implementations are updated, I could build
Chromium locally with this patch. This Reset implementation causes (for
some mysterious reason) -WError=overloaded-virtual failures when trying
to build libjingle APKs.

R=guoweis@webrtc.org, magjed@webrtc.org, pthatcher@webrtc.org
BUG=webrtc:2365

Review URL: https://codereview.webrtc.org/1411253002 .

Cr-Commit-Position: refs/heads/master@{#10352}
2015-10-21 09:54:10 +00:00
tommi
e4f96501fc Remove system_wrappers/interface/trace_event.h
BUG=

Review URL: https://codereview.webrtc.org/1417773002

Cr-Commit-Position: refs/heads/master@{#10346}
2015-10-21 06:00:57 +00:00
solenberg
0a617e22a4 Remove the default send channel in WVoE.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1364643003

Cr-Commit-Position: refs/heads/master@{#10344}
2015-10-20 22:49:45 +00:00
solenberg
43e83d44f0 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
Reason for revert:
webrtc_perf_tests started failing on Win32 Release, Mac32 Release and Linux64 Release (all running large tests). These were not caught by try bots.

Original issue's description:
> Implement AudioReceiveStream::GetStats().
>
> R=tommi@webrtc.org
> TBR=hta@webrtc.org
> BUG=webrtc:4690
>
> Committed: a457752f4a

TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1411083006

Cr-Commit-Position: refs/heads/master@{#10340}
2015-10-20 13:41:06 +00:00
Fredrik Solenberg
a457752f4a Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
TBR=hta@webrtc.org
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1390753002 .

Cr-Commit-Position: refs/heads/master@{#10338}
2015-10-20 13:01:55 +00:00
Peter Boström
f56eca031c Remove dummyinstantiation.cc.
Prevented Android libjingle APK building since EnsureAPIMatch is defined
but not used.

BUG=webrtc:2365
R=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1410643005 .

Cr-Commit-Position: refs/heads/master@{#10323}
2015-10-19 12:47:20 +00:00
pbos
22993e1a0c Unify FrameType and VideoFrameType.
Prevents some heap allocation and frame-type conversion since interfaces
mismatch. Also it's less confusing to have one type for this.

BUG=webrtc:5042
R=magjed@webrtc.org, mflodman@webrtc.org, henrik.lundin@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1371043003

Cr-Commit-Position: refs/heads/master@{#10320}
2015-10-19 09:39:15 +00:00
pbos
be16f79818 Remove simulcast bitrate modes.
Instead always use the SBM_VERY_HIGH setting.

BUG=webrtc:4885
R=hta@webrtc.org, mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1407693005

Cr-Commit-Position: refs/heads/master@{#10305}
2015-10-16 19:49:47 +00:00
ivoc
112a3d81db Added functions on libjingle API to start and stop the recording of an RtcEventLog.
BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1374253002

Cr-Commit-Position: refs/heads/master@{#10297}
2015-10-16 09:22:23 +00:00
stefan
c1aeaf0dc3 Wire up packet_id / send time callbacks to webrtc via libjingle.
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1363573002

Cr-Commit-Position: refs/heads/master@{#10289}
2015-10-15 14:26:17 +00:00
solenberg
1ac561447e Remove default receive channel from WVoE; baby step 3.
Get rid of default receive channel.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1385893002

Cr-Commit-Position: refs/heads/master@{#10262}
2015-10-13 10:58:25 +00:00
solenberg
8fb30c328b Remove default receive channel from WVoE; baby step 2.
Rename voe_channel_ to default_send_channel_id_.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1388733002

Cr-Commit-Position: refs/heads/master@{#10261}
2015-10-13 10:07:07 +00:00
Magnus Jedvert
1b40a9a8af RefCountInterface: Make AddRef() and Release() const
This CL makes AddRef() and Release() const member methods and the refcount integer mutable. This is reasonable, because they only manage the lifetime of the object, and this is also how it's done in Chromium.

The purpose is to be able to capture a const pointer in a scoped_refptr, which is currenty impossible. The practial problem this CL solves is this:

void Foo::Bar() const {}

rtc::Callback0<void> Foo::MakeClosure() const {
  return rtc::Bind(&Foo::Bar, this);
}

We currently capture |this| as const Foo*. With this CL, |this| will be captured as scoped_refptr<const Foo>.

A test is also added in bind_unittest to check this behaviour.

BUG=webrtc:5065
R=perkj@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1403683004 .

Cr-Commit-Position: refs/heads/master@{#10253}
2015-10-12 13:50:50 +00:00
solenberg
d4cec0d8fa Remove MediaChannel::SetRemoteRenderer().
This is following discussion in: https://codereview.webrtc.org/1385893002/diff/60001/talk/media/webrtc/webrtcvoiceengine.cc#newcode2410

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1398823003

Cr-Commit-Position: refs/heads/master@{#10237}
2015-10-09 15:55:54 +00:00
solenberg
98c68865e7 - Remove AudioTrackRenderer.
- Remove AddChannel/RemoveChannel from AudioRenderer interface.

BUG=webrtc:4690

Committed: https://crrev.com/1c0bb386b67835feb5934f503dddfe0912bce3ac
Cr-Commit-Position: refs/heads/master@{#10226}

Review URL: https://codereview.webrtc.org/1399553003

Cr-Commit-Position: refs/heads/master@{#10235}
2015-10-09 10:27:42 +00:00
solenberg
4bac9c53da Change SetOutputScaling to set a single level, not left/right levels.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1397773002

Cr-Commit-Position: refs/heads/master@{#10234}
2015-10-09 09:32:58 +00:00
solenberg
0b67546d8c Remove default receive channel from WVoE; baby step 1.
Rx AGC config bits copied from https://codereview.webrtc.org/1315903004.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1388723002

Cr-Commit-Position: refs/heads/master@{#10233}
2015-10-09 08:37:17 +00:00
torbjorng
eefbc3bbd7 Revert of Remove AudioTrackRenderer (patchset #3 id:40001 of https://codereview.webrtc.org/1399553003/ )
Reason for revert:
Breaks Chrome since its build files were not updated prior to file removal.

Original issue's description:
> - Remove AudioTrackRenderer.
> - Remove AddChannel/RemoveChannel from AudioRenderer interface.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/1c0bb386b67835feb5934f503dddfe0912bce3ac
> Cr-Commit-Position: refs/heads/master@{#10226}

TBR=tommi@webrtc.org,solenberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1393343003

Cr-Commit-Position: refs/heads/master@{#10228}
2015-10-08 20:10:40 +00:00
solenberg
1c0bb386b6 - Remove AudioTrackRenderer.
- Remove AddChannel/RemoveChannel from AudioRenderer interface.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1399553003

Cr-Commit-Position: refs/heads/master@{#10226}
2015-10-08 19:49:53 +00:00
Peter Boström
0c4e06b4c6 Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.

BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1362503003 .

Cr-Commit-Position: refs/heads/master@{#10196}
2015-10-07 10:23:32 +00:00
solenberg
d97ec30ce4 Remove default receive channel from WVoE; baby step 0.
Cleanup + add thread checker DCHECKs to various method in WebRtcVoiceEngine/MediaChannel.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1386653002

Cr-Commit-Position: refs/heads/master@{#10194}
2015-10-07 08:40:38 +00:00
stefan
1d8a506405 Add a PacketOptions struct to webrtc::Transport.
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.

BUG=4173

Review URL: https://codereview.webrtc.org/1376673004

Cr-Commit-Position: refs/heads/master@{#10144}
2015-10-02 10:39:40 +00:00
pbos
da903eaabb Unify newapi::RtcpMode and RTCPMethod.
BUG=webrtc:1695
R=solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1373903003

Cr-Commit-Position: refs/heads/master@{#10143}
2015-10-02 09:37:18 +00:00
peah
5aaa9b4fe4 Removed unused API functions in AudioProcessing and AudioProcessingModule
BUG=

Review URL: https://codereview.webrtc.org/1379123002

Cr-Commit-Position: refs/heads/master@{#10138}
2015-10-02 06:58:21 +00:00
solenberg
5b14b42e93 Remove unused SignalMediaError and infrastructure.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1362913004

Cr-Commit-Position: refs/heads/master@{#10133}
2015-10-01 11:10:40 +00:00
magjed
b09b660c53 Remove cricket::VideoFrame::Set/GetElapsedTime()
This CL is a baby step towards consolidating the timestamps in cricket::VideoFrame and webrtc::VideoFrame, so that we can unify the frame classes in the future.

The elapsed time functionality is not really used. If a video sink wants to know the elapsed time since the first frame they can store the first timestamp themselves and calculate the time delta to later frames. This is already done in all video sinks that need the elapsed time. Having redundant timestamps in the frame classes is confusing and error prone.

TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1324263004

Cr-Commit-Position: refs/heads/master@{#10131}
2015-10-01 10:03:05 +00:00
solenberg
dfc8f4ff87 Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1378513003

Cr-Commit-Position: refs/heads/master@{#10130}
2015-10-01 09:32:41 +00:00
solenberg
63b345441a Simplify handling of options in WebRtcVoiceMediaEngine.
Also removes unnecessary typedef ChannelList.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1364753002

Cr-Commit-Position: refs/heads/master@{#10107}
2015-09-29 13:06:36 +00:00
pbos
2d566686a2 Unify Transport and newapi::Transport interfaces.
BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1369263002

Cr-Commit-Position: refs/heads/master@{#10096}
2015-09-28 16:59:36 +00:00
Peter Boström
34fbfff068 Remove VideoMediaChannel::SetRender().
Was a no-op in current implementation.

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1334793003 .

Cr-Commit-Position: refs/heads/master@{#10059}
2015-09-24 17:20:36 +00:00
solenberg
4a3ccad29e Remove SetAudioDelayOffset() and friends.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1364093002

Cr-Commit-Position: refs/heads/master@{#10047}
2015-09-24 10:53:14 +00:00
solenberg
61e933eac7 Remove ChannelManager::GetCapabilities()
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1364083002

Cr-Commit-Position: refs/heads/master@{#10045}
2015-09-24 08:45:41 +00:00
deadbeef
cbecd358e0 Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ )
Reason for revert:
This CL just landed: https://codereview.chromium.org/1323243006/

Which fixes the FYI bots for the original CL, and breaks them for this revert.

Original issue's description:
> Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
>
> Reason for revert:
> This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
>
> Original issue's description:
> > TransportController refactoring.
> >
> > Getting rid of TransportProxy, and in its place adding a
> > TransportController class which will facilitate access to and manage
> > the lifetimes of Transports. These Transports will now be accessed
> > solely from the worker thread, simplifying their implementation.
> >
> > This refactoring also pulls Transport-related code out of BaseSession.
> > Which means that BaseChannels will now rely on the TransportController
> > interface to create channels, rather than BaseSession.
> >
> > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> > Cr-Commit-Position: refs/heads/master@{#10022}
>
> TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c
> Cr-Commit-Position: refs/heads/master@{#10024}

TBR=pthatcher@webrtc.org,torbjorng@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1361773005

Cr-Commit-Position: refs/heads/master@{#10036}
2015-09-23 18:50:31 +00:00
Fredrik Solenberg
d0b5b091e4 Add myself as OWNER of webrtc/voice_engine and talk/media/webrtc.
BUG=webrtc:4690
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1359983003 .

Cr-Commit-Position: refs/heads/master@{#10035}
2015-09-23 15:15:29 +00:00
henrika
c14f5ff60f Improving support for Android Audio Effects in WebRTC.
Now also supports AGC and NS effects and adds the possibility
to override default settings.

R=magjed@webrtc.org, pbos@webrtc.org, sophiechang@chromium.org
TBR=perkj
BUG=NONE

Review URL: https://codereview.webrtc.org/1344563002 .

Cr-Commit-Position: refs/heads/master@{#10030}
2015-09-23 12:09:40 +00:00
Peter Boström
d5c75b1a0b Reduce LS_INFO spam from voice_engine/.
Removes ShouldIgnoreTrace from WebRtcVoiceEngine and removes the spammy
log instances instead. Also removes trace-style logging from getters
(::GetLocalSSRC() for instance would print what SSRC it got, spamming
the log).

BUG=
R=henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1347353004 .

Cr-Commit-Position: refs/heads/master@{#10028}
2015-09-23 11:24:43 +00:00