Commit Graph

292 Commits

Author SHA1 Message Date
8ad582d83f Remove DeviceManager and DeviceInfo.
BUG=webrtc:5615, webrtc:5620

Review URL: https://codereview.webrtc.org/1715883002

Cr-Commit-Position: refs/heads/master@{#12020}
2016-03-16 16:35:04 +00:00
05103314e5 Drop VideoOptions from VideoSendParameters.
BUG=webrtc:5438

Review URL: https://codereview.webrtc.org/1695663003

Cr-Commit-Position: refs/heads/master@{#12011}
2016-03-16 09:22:57 +00:00
fb647a67be Initialize/configure video encoders asychronously.
Greatly speeds up setRemoteDescription() by moving encoder initialization
off the main worker thread, which is free to move onto gathering ICE
candidates and other tasks while InitEncode() is performed. It also
un-blocks PeerConnection GetStats() which is no longer blocked on
encoder initialization.

BUG=webrtc:5410
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1757313002 .

Cr-Commit-Position: refs/heads/master@{#11983}
2016-03-14 15:59:03 +00:00
31642aa8f9 Relanding https://codereview.webrtc.org/1715883002/ in pieces.
- Change argument type to int for SetSendTelephoneEventPayloadType()

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1798903002

Cr-Commit-Position: refs/heads/master@{#11980}
2016-03-14 15:00:40 +00:00
d44c077dce Revert of Safe numeric library: base/numerics (copied from Chromium) (patchset #11 id:250001 of https://codereview.webrtc.org/1753293002/ )
Reason for revert:
Looks like the Chrome iOS build is broken because of these two changes.  So I'm going to have to revert.  Here's the error:

https://build.chromium.org/p/tryserver.chromium.mac/builders/ios_rel_device_ninja/builds/185624/steps/compile/logs/stdio

FAILED: rm -f arch/libsafe_numerics.arm64.a && ./gyp-mac-tool filter-libtool libtool  -static -o arch/libsafe_numerics.arm64.a
error: /Applications/Xcode7.0.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/libtool: no files specified
Usage: /Applications/Xcode7.0.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/libtool -static [-] file [...] [-filelist listfile[,dirname]] [-arch_only arch] [-sacLT] [-no_warning_for_no_symbols]
Usage: /Applications/Xcode7.0.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/libtool -dynamic [-] file [...] [-filelist listfile[,dirname]] [-arch_only arch] [-o output] [-install_name name] [-compatibility_version #] [-current_version #] [-seg1addr 0x#] [-segs_read_only_addr 0x#] [-segs_read_write_addr 0x#] [-seg_addr_table <filename>] [-seg_addr_table_filename <file_system_path>] [-all_load] [-noall_load]
FAILED: rm -f arch/libsafe_numerics.armv7.a && ./gyp-mac-tool filter-libtool libtool  -static -o arch/libsafe_numerics.armv7.a
error: /Applications/Xcode7.0.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/libtool: no files specified
Usage: /Applications/Xcode7.0.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/libtool -static [-] file [...] [-filelist listfile[,dirname]] [-arch_only arch] [-sacLT] [-no_warning_for_no_symbols]
Usage: /Applications/Xcode7.0.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/libtool -dynamic [-] file [...] [-filelist listfile[,dirname]] [-arch_only arch] [-o output] [-install_name name] [-compatibility_version #] [-current_version #] [-seg1addr 0x#] [-segs_read_only_addr 0x#] [-segs_read_write_addr 0x#] [-seg_addr_table <filename>] [-seg_addr_table_filename <file_system_path>] [-all_load] [-noall_load]
ninja: build stopped: subcommand failed.

Original issue's description:
> Safe numeric library added: base/numerics (copied from Chromium)
>
> This copies the contents (unittest excluded) of base/numerics in
> chromium to base/numerics in webrtc. Files added:
> - safe_conversions.h
> - safe_conversions_impl.h
> - safe_math.h
> - safe_math_impl.h
>
> A really old version of safe_conversions[_impl].h previously existed in
> base/, this has been deleted and sources using it have been updated
> to include the new base/numerics/safe_converions.h.
>
> This CL also adds a DEPS file to webrtc/base.
>
> NOPRESUBMIT=True
> BUG=webrtc:5548, webrtc:5623
>
> Committed: https://crrev.com/de1c81b2d2196be611674aa6019b9db3a9329042
> Cr-Commit-Position: refs/heads/master@{#11907}

TBR=kjellander@webrtc.org,kwiberg@webrtc.org,tina.legrand@webrtc.org,hbos@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5548, webrtc:5623

Review URL: https://codereview.webrtc.org/1792613002 .

Cr-Commit-Position: refs/heads/master@{#11965}
2016-03-12 01:12:40 +00:00
e7ba08695c Reconfigure video encoders even when not sending.
Permits sending faster when having an attached track before actually
sending since the configured stream is ready to encode as soon as a call
is accepted.

BUG=webrtc:5410
R=deadbeef@webrtc.org

Review URL: https://codereview.webrtc.org/1790703002 .

Cr-Commit-Position: refs/heads/master@{#11963}
2016-03-11 23:02:37 +00:00
8842c3e41b Relanding https://codereview.webrtc.org/1715883002/ in pieces.
- Use better types in AudioSendStream::SendTelephoneEvent() and related methods.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1782053002

Cr-Commit-Position: refs/heads/master@{#11953}
2016-03-11 11:06:48 +00:00
f0dcfe2c81 Change VideoRtpReceiver to create remote VideoTrack and VideoTrackSource.
This enabled us to be able to remove VideoTrack::GetSink and RemoteVideoCapturer.

Since video frames from the decoder is delivered on a media engine internal thread, VideoBroadCaster must be made thread safe.

BUG=webrtc:5426
R=deadbeef@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1765423005 .

Cr-Commit-Position: refs/heads/master@{#11944}
2016-03-10 17:32:08 +00:00
3ecb5c8698 Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ )
Reason for revert:
Breaks Chromium FYI bots for Android. E.g. https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/4486/steps/content_browsertests/logs/stdio

Original issue's description:
> - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs.
> - Use better types in AudioSendStream::SendTelephoneEvent() and related methods.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/8886c816582a7c6190c5429222cb8096fca302a6
> Cr-Commit-Position: refs/heads/master@{#11927}

TBR=tina.legrand@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1776243003

Cr-Commit-Position: refs/heads/master@{#11930}
2016-03-09 15:32:05 +00:00
8886c81658 - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs.
- Use better types in AudioSendStream::SendTelephoneEvent() and related methods.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1722253002

Cr-Commit-Position: refs/heads/master@{#11927}
2016-03-09 11:32:53 +00:00
ca8b404e86 Add tracing to interesting media-related methods.
Accounts for a lot of worker-thread blocking by voice-related code or
initializing SRTP.

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1780543003 .

Cr-Commit-Position: refs/heads/master@{#11920}
2016-03-08 22:24:21 +00:00
1a018dcda3 Prevent a voice channel from sending data before a source is set.
At the top level, setting a track on an RtpSender is equivalent to
setting a source (previously called a renderer)
on a voice send stream. An RtpSender without a track
is not supposed to send data (not even muted data), so a send stream without
a source shouldn't send data.

Also replacing SendFlags with a boolean and implementing "Start"
and "Stop" methods on AudioSendStream, which was planned anyway
and simplifies this CL.

R=pthatcher@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1741933002 .

Cr-Commit-Position: refs/heads/master@{#11918}
2016-03-08 20:37:48 +00:00
72e29d2cbb On WVoMC::SetSendParameters(), figure out send codec settings ONCE, not for each send stream.
This CL is a first step to moving codec configuration into AudioSendStream.

BUG=webrtc:4690
TBR=ossu@webrtc.org

Review URL: https://codereview.webrtc.org/1765873002

Cr-Commit-Position: refs/heads/master@{#11911}
2016-03-08 14:35:22 +00:00
de1c81b2d2 Safe numeric library added: base/numerics (copied from Chromium)
This copies the contents (unittest excluded) of base/numerics in
chromium to base/numerics in webrtc. Files added:
- safe_conversions.h
- safe_conversions_impl.h
- safe_math.h
- safe_math_impl.h

A really old version of safe_conversions[_impl].h previously existed in
base/, this has been deleted and sources using it have been updated
to include the new base/numerics/safe_converions.h.

This CL also adds a DEPS file to webrtc/base.

NOPRESUBMIT=True
BUG=webrtc:5548, webrtc:5623

Review URL: https://codereview.webrtc.org/1753293002

Cr-Commit-Position: refs/heads/master@{#11907}
2016-03-08 12:46:07 +00:00
905f8e7e9d Make ReconfigureVideoEncoder void.
Also moves and simplifies SetSendCodec from VideoSendStream to mostly
inside ViEEncoder. This is necessary for making
ReconfigureVideoEncoder asynchronous as we don't post any result back.

BUG=webrtc:5494
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1754283002 .

Cr-Commit-Position: refs/heads/master@{#11847}
2016-03-02 16:00:07 +00:00
f475277547 Rename constants files in webrtc/{media,p2p}
Multiple sources with the same names forces ugly GYP hacks in
Chromium's libjingle.gyp. Rename the sources in WebRTC to
enable cleaning this up in Chromium.

To summarize:
webrtc/media/base/constants.{cc,h} -> mediaconstants.{cc,h}
webrtc/p2p/base/constants.{cc,h} -> p2pconstants.{cc,h}

This CL will require coordinating landing a roll in Chromium.

BUG=webrtc:4256
NOTRY=True

Review URL: https://codereview.webrtc.org/1750593002

Cr-Commit-Position: refs/heads/master@{#11842}
2016-03-02 13:42:35 +00:00
91e1c15f8e Make sure rotation is not applied by the capturer if the CVO exenstion is set before the send stream is created.
BUG=webrtc:5621

TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1757853002

Cr-Commit-Position: refs/heads/master@{#11841}
2016-03-02 13:34:08 +00:00
60653ba3cc New flag is_screencast in VideoOptions.
This cl copies the value of cricket::VideoCapturer::IsScreencast into
a flag in VideoOptions. It is passed on via the chain

VideortpSender::SetVideoSend
WebRtcVideoChannel2::SetVideoSend
WebRtcVideoChannel2::SetOptions
WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions

Where it's used, in
WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame, we can look it up
in parameters_, instead of calling capturer_->IsScreencast().

Doesn't touch screencast logic related to cpu adaptation, since that
code is in flux in a different cl.

Also drop the is_screencast flag from the Dimensions struct, and drop separate options argument from ConfigureVideoEncoderSettings and SetCodecAndOptions, instead always using the options recorded in VideoSendStreamParameters::options.

In the tests, changed FakeVideoCapturer::is_screencast to be a construction time flag. Generally, unittests of screencast have to both use a capturer configured for screencast, and set the screencast flag using SetSendParameters. Since the automatic connection via VideoSource and VideoRtpSender isn't involved in the unit tests.

Note that using SetSendParameters to set the screencast flag doesn't make sense, since it's not per-stream. SetVideoSend would be more appropriate. That should be fixed if/when we drop VideoOptions from SetSendParameters.

BUG=webrtc:5426
R=pbos@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1711763003 .

Cr-Commit-Position: refs/heads/master@{#11837}
2016-03-02 10:41:49 +00:00
0db023a70b Move suspend_below_min_bitrate from VideoOptions to MediaConfig.
Rename SetCodecAndOptions to SetCodec, it no longer sets or uses the
VideoOptions. In MediaConfig, collect the video-related flags into a
struct.

As a followup, it should be possible to delete VideoOptions from
VideoSendParameters and VideoSendStreamParameters.

TBR=pthatcher@webrtc.org
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1745003002

Cr-Commit-Position: refs/heads/master@{#11828}
2016-03-01 12:30:07 +00:00
739fcb989d Cleanup of webrtc::VideoFrame.
Delete EqualsFrame method, used only by tests. Delete one of the
CreateFrame methods. Drop return value for CreateEmptyFrame, CreateFrame
and CopyFrame.

BUG=webrtc:5426

Committed: https://crrev.com/208019637bfed975f8f13b16d40b90e200763cd6
Cr-Commit-Position: refs/heads/master@{#11783}

R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1679323002 .

Cr-Commit-Position: refs/heads/master@{#11811}
2016-02-29 12:11:57 +00:00
Per
fb45d170c0 Reland Remove unused cricket::VideoCapturer methods. Originally reviewed and landed as patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/)
I readded virtual bool Pause(bool paused) for now with a dummy implementation since Chrome remoting override this method.

Original cl description:

Removed unused cricket::VideoCapturer methods:

void UpdateAspectRatio(int ratio_w, int ratio_h);
void ClearAspectRatio();
bool Pause(bool paused);
Restart(const VideoFormat& capture_format);
MuteToBlackThenPause(bool muted);
IsMuted() const
set_square_pixel_aspect_ratio
bool square_pixel_aspect_ratio()

This cl also remove the use of messages and posting of state change.
Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
It does not add restrictions on what thread frames are delivered on though.

There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.

BUG=webrtc:5426
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1744153002 .

Cr-Commit-Position: refs/heads/master@{#11809}
2016-02-29 11:07:45 +00:00
2d5f0913f2 Move direct use of VideoCapturer::VideoAdapter to VideoSinkWants.
The purose of this cl is to remove dependency on cricket::VideoCapturer from WebRtcVideoChannel2.
This cl change CPU adaptation to use a new VideoSinkWants.Resolution

Cl is WIP and uploaded to start the discussion.

Tested on a N5 with hw acceleration turned off.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1695263002

Cr-Commit-Position: refs/heads/master@{#11804}
2016-02-29 08:04:50 +00:00
7ffeab525c Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
This is a reland of https://codereview.webrtc.org/1737593002/ minus
the added missing headers in webrtc/{BUILD.gn,webrtc.gyp} and
webrtc/common.gyp that breaks GN in Chromium since it's using
the --check flag (which we should support).

BUG=webrtc:4243, webrtc:5589
TESTED=Tried generating GN files with --check in a Chromium checkout with this patch applied, successfully.
TBR=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1740873003 .

Cr-Commit-Position: refs/heads/master@{#11794}
2016-02-26 21:46:22 +00:00
54ebfca934 Revert of Cleanup of webrtc::VideoFrame. (patchset #6 id:100001 of https://codereview.webrtc.org/1679323002/ )
Reason for revert:
Breaks downstream compilation. Please make non-breaking API changes for the reland or coordinate fixing downstream code quickly with the sheriff.

Original issue's description:
> Cleanup of webrtc::VideoFrame.
>
> Delete EqualsFrame method, used only by tests. Delete one of the
> CreateFrame methods. Drop return value for CreateEmptyFrame, CreateFrame
> and CopyFrame.
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/208019637bfed975f8f13b16d40b90e200763cd6
> Cr-Commit-Position: refs/heads/master@{#11783}

TBR=pbos@webrtc.org,perkj@webrtc.org,pthatcher@webrtc.org,mflodman@webrtc.org,marpan@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1743613002

Cr-Commit-Position: refs/heads/master@{#11789}
2016-02-26 15:38:57 +00:00
208019637b Cleanup of webrtc::VideoFrame.
Delete EqualsFrame method, used only by tests. Delete one of the
CreateFrame methods. Drop return value for CreateEmptyFrame, CreateFrame
and CopyFrame.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1679323002

Cr-Commit-Position: refs/heads/master@{#11783}
2016-02-26 14:40:47 +00:00
686a8efad9 Replace scoped_ptr with unique_ptr in webrtc/media/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1728503002

Cr-Commit-Position: refs/heads/master@{#11779}
2016-02-26 11:00:39 +00:00
74622e0613 Revert of Removed unused cricket::VideoCapturer methods (patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/ )
Reason for revert:
Breaks remoting::protocol::WebrtcVideoCapturerAdapter::Pause'

See https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/3689/steps/compile/logs/stdio

Original issue's description:
> Removed unused cricket::VideoCapturer methods:
>
> void UpdateAspectRatio(int ratio_w, int ratio_h);
> void ClearAspectRatio();
> ool Pause(bool paused);
> Restart(const VideoFormat& capture_format);
> MuteToBlackThenPause(bool muted);
> IsMuted() const
> set_square_pixel_aspect_ratio
> bool square_pixel_aspect_ratio()
>
> This cl also remove the use of messages and posting of state change.
> Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
> It does not add restrictions on what thread frames are delivered on though.
>
> There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/e9c0cdff2dad2553b6ff6820c0c7429cb2854861
> Cr-Commit-Position: refs/heads/master@{#11773}

TBR=magjed@webrtc.org,pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1740963002

Cr-Commit-Position: refs/heads/master@{#11777}
2016-02-26 10:54:43 +00:00
e9c0cdff2d Removed unused cricket::VideoCapturer methods:
void UpdateAspectRatio(int ratio_w, int ratio_h);
void ClearAspectRatio();
ool Pause(bool paused);
Restart(const VideoFormat& capture_format);
MuteToBlackThenPause(bool muted);
IsMuted() const
set_square_pixel_aspect_ratio
bool square_pixel_aspect_ratio()

This cl also remove the use of messages and posting of state change.
Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
It does not add restrictions on what thread frames are delivered on though.

There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1733673002

Cr-Commit-Position: refs/heads/master@{#11773}
2016-02-26 07:36:22 +00:00
7324eb9e62 Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
Reason for revert:
Breaks GN in chromium.

Original issue's description:
> Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.
>
> webrtc/audio/audio_sink.h is used by voice engine, but webrtc/audio is
> depending on voice engine, resulting in a cyclic dependency (which we
> don't detect since we have that check turned off, see webrtc:4243).
>
> BUG=webrtc:4243, webrtc:5589
> R=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org
> TBR=tommi@webrtc.org
>
> Committed: https://crrev.com/99b345c4e50c59a776c56949c17da3f50992f1a2
> Cr-Commit-Position: refs/heads/master@{#11766}

TBR=solenberg@webrtc.org,pbos@webrtc.org,perkj@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:4243, webrtc:5589

Review URL: https://codereview.webrtc.org/1739783002

Cr-Commit-Position: refs/heads/master@{#11769}
2016-02-25 16:37:02 +00:00
99b345c4e5 Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.
webrtc/audio/audio_sink.h is used by voice engine, but webrtc/audio is
depending on voice engine, resulting in a cyclic dependency (which we
don't detect since we have that check turned off, see webrtc:4243).

BUG=webrtc:4243, webrtc:5589
R=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1737593002 .

Cr-Commit-Position: refs/heads/master@{#11766}
2016-02-25 14:12:48 +00:00
a26ac925f7 Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ )
Reason for revert:
Revert breaks other uses, a fix will be rolled into Chromium instead.

Original issue's description:
> Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ )
>
> Reason for revert:
> Breaks Chromium.
>
> Original issue's description:
> > Remove ignored return code from modules.
> >
> > ModuleProcessImpl doesn't act on return codes and having them around is
> > confusing (it's unclear what an error return code here would do even).
> >
> > BUG=
> > R=tommi@webrtc.org
> >
> > Committed: f14c47a58c
>
> TBR=tommi@webrtc.org,pbos@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/da33a8a2a22f6d19ba2a8cce963beafbdbaa8fd8
> Cr-Commit-Position: refs/heads/master@{#11761}

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

Review URL: https://codereview.webrtc.org/1737013002

Cr-Commit-Position: refs/heads/master@{#11762}
2016-02-25 12:50:09 +00:00
da33a8a2a2 Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ )
Reason for revert:
Breaks Chromium.

Original issue's description:
> Remove ignored return code from modules.
>
> ModuleProcessImpl doesn't act on return codes and having them around is
> confusing (it's unclear what an error return code here would do even).
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: f14c47a58c

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

Review URL: https://codereview.webrtc.org/1736663004

Cr-Commit-Position: refs/heads/master@{#11761}
2016-02-25 12:34:12 +00:00
65c8fd78c6 Remove the 'audioDebugRecording' media constraint and the aec_dump AudioOptions flag.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1565133002

Cr-Commit-Position: refs/heads/master@{#11753}
2016-02-24 22:43:18 +00:00
f14c47a58c Remove ignored return code from modules.
ModuleProcessImpl doesn't act on return codes and having them around is
confusing (it's unclear what an error return code here would do even).

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1703833002 .

Cr-Commit-Position: refs/heads/master@{#11747}
2016-02-24 15:51:23 +00:00
fffa42b57e Replace scoped_ptr with unique_ptr in webrtc/audio/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1706183002

Cr-Commit-Position: refs/heads/master@{#11723}
2016-02-23 18:46:39 +00:00
a293ef0618 Apply VideoOptions per stream.
BUG=webrtc:5438

Review URL: https://codereview.webrtc.org/1608793004

Cr-Commit-Position: refs/heads/master@{#11656}
2016-02-17 15:24:57 +00:00
4b4dc86c61 Remove conference_mode flag from AudioOptions and VideoOptions.
For audio, the flag is apparently unused. For video, the flag is moved to
VideoSendParameters, with the intention to keep only per-stream flags in
VideoOptions. The flag is used for the webrtcvideoengine2 logic commented like

  // Conference mode screencast uses 2 temporal layers split at 100kbit.

  // For screenshare in conference mode, tl0 and tl1 bitrates are piggybacked
  // on the VideoCodec struct as target and max bitrates, respectively.
  // See eg. webrtc::VP8EncoderImpl::SetRates().

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1697163002

Cr-Commit-Position: refs/heads/master@{#11651}
2016-02-17 13:25:40 +00:00
Per
a50924115e This reland https://codereview.webrtc.org/1655793003/ with the change that cricket::VideoCapturer::SignalVideoFrame is added back and used for frame forwarding. It is used in Chrome remoting.
BUG=webrtc:5426
R=pthatcher@webrtc.org
TBR=pthatcher@webrtc.org for moved channelmanager....

Review URL: https://codereview.webrtc.org/1689923002 .

Cr-Commit-Position: refs/heads/master@{#11597}
2016-02-12 12:31:08 +00:00
51542be8ce Introduce struct MediaConfig, with construction-time settings.
Pass it to MediaController constructor and down to WebRtcVideoEngine2
and WebRtcVoiceEngine.

Follows discussion on https://codereview.webrtc.org/1646253004/

TBR=pthatcher@webrtc.org
BUG=webrtc:5438

Review URL: https://codereview.webrtc.org/1670153003

Cr-Commit-Position: refs/heads/master@{#11595}
2016-02-12 10:27:12 +00:00
d73c99cf22 Initial cleanup of cricket::VideoFrame.
Deleted GetRotation (old alias for GetVideoRotation).
Deleted CopyToBuffer.
Deleted Sizeof.
Deleted Write.
Demote CopyToPlanes to protected status.

BUG=webrtc:5426

Committed: https://crrev.com/4d575b0d4276422bdf7b595d92c57c4f0f8ce0e9
Cr-Commit-Position: refs/heads/master@{#11564}

Review URL: https://codereview.webrtc.org/1688643003

Cr-Commit-Position: refs/heads/master@{#11594}
2016-02-12 09:01:15 +00:00
9b8df25c73 Move talk/session/media -> webrtc/pc
The libjingle_p2p target is renamed to rtc_pc.
The libjingle_p2p_unittest test will be renamed in a
separate follow-up CL, to make it possible to run all
trybots successfully for this CL.

BUG=webrtc:5419
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1691463002 .

Cr-Commit-Position: refs/heads/master@{#11592}
2016-02-12 05:48:10 +00:00
5ad129741c Rename webrtc/media/webrtc -> webrtc/media/engine
BUG=webrtc:5420
NOTRY=True
R=pthatcher@google.com, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1684163002 .

Cr-Commit-Position: refs/heads/master@{#11591}
2016-02-12 05:39:50 +00:00