Commit Graph

604 Commits

Author SHA1 Message Date
fb98b9edb4 Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #6 id:100001 of https://codereview.webrtc.org/2007563002/ )
Reason for revert:
Reverting temporarily.  Need to fix tests downstream that pass invalid arguments.

Original issue's description:
> Adding a some checks and switching out a few assert for RTC_[D]CHECK.
> These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled.  I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.
>
> BUG=chromium:613482
> NOTRY=true
> (using notry due to offline android_arm64_rel bot)
>
> Committed: https://crrev.com/d36df89d40bde3c62ee5cbff841933e50b3c007b
> Cr-Commit-Position: refs/heads/master@{#12870}

TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:613482

Review-Url: https://codereview.webrtc.org/2006243002
Cr-Commit-Position: refs/heads/master@{#12874}
2016-05-24 13:44:36 +00:00
d36df89d40 Adding a some checks and switching out a few assert for RTC_[D]CHECK.
These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled.  I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.

BUG=chromium:613482
NOTRY=true
(using notry due to offline android_arm64_rel bot)

Review-Url: https://codereview.webrtc.org/2007563002
Cr-Commit-Position: refs/heads/master@{#12870}
2016-05-24 12:49:10 +00:00
a89ab965f2 Enable muted state by default in VoE
This change turns muted state on by default in VoiceEngine, but not
for NetEq or AudioCodingModule when used stand-alone.

The expected effect is that voice channels that have not received any
packets for some time should reduce their CPU usage. This should have
a noticeable effect on endpoints with many incoming streams, but where
only a few have packets incoming at any given time (i.e., where an
intermediate server filters out the majority of the streams).

BUG=webrtc:5606
NOTRY=True

Review-Url: https://codereview.webrtc.org/1987143003
Cr-Commit-Position: refs/heads/master@{#12797}
2016-05-18 15:52:52 +00:00
42dda50860 Propagate muted info from VoE Channel to AudioConferenceMixer
Required updating of a few related classes and tests.

BUG=webrtc:5609
NOTRY=True

Review-Url: https://codereview.webrtc.org/1986093002
Cr-Commit-Position: refs/heads/master@{#12794}
2016-05-18 12:36:07 +00:00
d4ccb00b9e Propagate muted parameter to VoE::Channel
Deleted the temporary ACM method without the muted parameter, and had
to modify several tests for this. The muted parameter is not yet propagated to the AudioConferenceMixer; this is the next step.

BUG=webrtc:5609
TBR=perkj@webrtc.org

Review-Url: https://codereview.webrtc.org/1985743002
Cr-Commit-Position: refs/heads/master@{#12779}
2016-05-17 19:22:03 +00:00
cd6ae6652f Removing some old code which looked like it had to do with NACK handling but in reality did nothing.
BUG=webrtc:5762, webrtc:4690
R=stefan@webrtc.org
TBR=mflodman

Review URL: https://codereview.webrtc.org/1946183002 .

Cr-Commit-Position: refs/heads/master@{#12682}
2016-05-11 11:05:13 +00:00
d28db7fd65 Delete all use of tick_util.h.
Depends on Chrome cl https://codereview.chromium.org/1888003002/, which was landed some time ago.

BUG=webrtc:5740
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1888593004 .

Cr-Commit-Position: refs/heads/master@{#12674}
2016-05-10 14:31:58 +00:00
82d7862fe7 Change default timestamp to 64 bits in all webrtc directories.
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1835053002 .

Cr-Commit-Position: refs/heads/master@{#12646}
2016-05-06 18:29:27 +00:00
4adbbcfe7a Move ADM Create() method to public interface.
ADMs were previously created by CreateAudioDeviceModule which was
removed in previous refactoring without a replacement added.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1944883002 .

Cr-Commit-Position: refs/heads/master@{#12613}
2016-05-03 19:51:31 +00:00
3d7db263b9 Switch voice transport to use Call and Stream instead of VoENetwork.
VoENetwork is kept for now, but is not really used anylonger.

webrtcvoiceengine is changed to have the same behavior for unsignaled
ssrc as video has, which is reflected by disabling one test case and
this will be discussed and followed up.

BUG=webrtc:5079

TBR=tommi

Review-Url: https://codereview.webrtc.org/1909333002
Cr-Commit-Position: refs/heads/master@{#12555}
2016-04-29 07:57:21 +00:00
1c7fdd86eb Remove calls to ScopedToUnique and UniqueToScoped
They're just no-ops now, and will soon go away.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1914153002

Cr-Commit-Position: refs/heads/master@{#12510}
2016-04-26 15:18:13 +00:00
4311ba59d8 Refactored CL for moving the output to a separate thread.
The logging thread is always active. The main thread uses SwapQueues to pass events to the logging thread. The logging thread moves the events to either a RingBuffer history in memory, or to a string which is written to disc.

RtcEventLogImpl constructor takes a clock for easier testing.

BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1687703002

Cr-Commit-Position: refs/heads/master@{#12476}
2016-04-22 19:40:46 +00:00
e532aec252 Add isolate files for Android tests
BUG=chromium:583318
TESTED=Passing runs with:
GYP_DEFINES='test_isolation_mode=prepare OS=android' webrtc/build/gyp_webrtc
ninja -C out/Release
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1882963003

Cr-Commit-Position: refs/heads/master@{#12397}
2016-04-18 03:08:28 +00:00
d53a3f9758 Early initialize recording on the ADM from WebRtcVoiceMediaChannel.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1827263002

Cr-Commit-Position: refs/heads/master@{#12369}
2016-04-14 20:56:45 +00:00
c8d071e4e0 Switch to using new ACM methods for encoder management
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1677013002

Cr-Commit-Position: refs/heads/master@{#12267}
2016-04-06 19:22:45 +00:00
96bd50262a VoE: Handle empty playout timestamp differently
With this change, the VoE Channel will handle the case of an empty
playout timestamp (from audio_coding_->PlayoutTimestamp())
differently. The purpose of the change is to prepare for an upcoming
change in NetEq where empty values will be returned more often (i.e.,
not only before the first packet is received).

BUG=webrtc:5669

Review URL: https://codereview.webrtc.org/1857183002

Cr-Commit-Position: refs/heads/master@{#12261}
2016-04-06 11:14:03 +00:00
9a410dd082 Change NetEq::GetPlayoutTimestamp to return an rtc::Optional<uint32_t>
This is in preparation for changes to when the playout timestamp is
valid.

BUG=webrtc:5669

Review URL: https://codereview.webrtc.org/1853183002

Cr-Commit-Position: refs/heads/master@{#12256}
2016-04-06 08:39:30 +00:00
ff97631e3c - Add temporary VoEBase::audio_device_module() method.
- Remove WVoE::SetAudioDeviceModule() - the ADM is now supplied in ctor.
- Remove WVoE::Init() and WVoE::Terminate().
- Remove MediaEngineInterface::Terminate().

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1830213002

Cr-Commit-Position: refs/heads/master@{#12173}
2016-03-31 06:28:56 +00:00
1d0313916b Reland https://codereview.webrtc.org/1802993002/
Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.

BUG=webrtc:4690

Committed: https://crrev.com/69a81999ace08e40e2b2ec526b0e111aa11b9538
Cr-Commit-Position: refs/heads/master@{#12015}

Review URL: https://codereview.webrtc.org/1840893004

Cr-Commit-Position: refs/heads/master@{#12157}
2016-03-30 09:42:37 +00:00
1c2af8e319 Avoid clicks when muting/unmuting a voe::Channel.
Muting/unmuting is triggered in the PeerConnection API by calling setEnable() on an audio track.

BUG=webrtc:5671

Review URL: https://codereview.webrtc.org/1810413002

Cr-Commit-Position: refs/heads/master@{#12121}
2016-03-24 17:36:06 +00:00
1d1944187f Replace RefCountImpl with rtc::RefCountedObject.
Removes code duplication and use of the dangerous public destructor in
RefCountImpl.

Also making wider use of scoped_refptr and fixing various leaks in the
process.

BUG=webrtc:5229
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1477013005 .

Cr-Commit-Position: refs/heads/master@{#12075}
2016-03-21 15:44:41 +00:00
b031955770 Deprecate AudioProcessing::AnalyzeReverseStream(AudioFrame) API
Review URL: https://codereview.webrtc.org/1783693005

Cr-Commit-Position: refs/heads/master@{#12045}
2016-03-18 03:39:57 +00:00
da116c4c37 Use ProcessReverseStream in VoiceEngines OutputMixer
Review URL: https://codereview.webrtc.org/1776363002

Cr-Commit-Position: refs/heads/master@{#12044}
2016-03-17 23:43:35 +00:00
94a23f04af Reland "Add check_deps rules in DEPS files."
Relanding https://codereview.webrtc.org/1796413002/
without the change to the openmax_dl include path
(which broke downstream code).

TBR=tommi@webrtc.org
BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc

Review URL: https://codereview.webrtc.org/1804333002 .

Cr-Commit-Position: refs/heads/master@{#12031}
2016-03-17 11:05:50 +00:00
b69395b374 Revert of Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code. (patchset #2 id:20001 of https://codereview.webrtc.org/1802993002/ )
Reason for revert:
Revert because it breaks downstream code.

Original issue's description:
> Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/69a81999ace08e40e2b2ec526b0e111aa11b9538
> Cr-Commit-Position: refs/heads/master@{#12015}

TBR=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/1812453002

Cr-Commit-Position: refs/heads/master@{#12016}
2016-03-16 14:05:21 +00:00
69a81999ac Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1802993002

Cr-Commit-Position: refs/heads/master@{#12015}
2016-03-16 12:59:04 +00:00
56cf60e717 Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
Reason for revert:
The openmax_dl include change breaks downstream projects.

Original issue's description:
> Add check_deps rules in DEPS files.
>
> Add fine-grained check_deps rules for all of WebRTC.
> This will help both maintaining sane dependencies and provides a way
> to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
>
> Example:
> buildtools/checkdeps/graphdeps.py --root=. --format=png \
> --out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
> --excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
>
> will produce a neat webrtc.png image showcasing the dependencies
> (according to the DEPS file) for the bitrate_controller module.
> Some dependencies are filtered out for readability.
>
> BUG=webrtc:5623
> TESTED=Passing runs using:
> buildtools/checkdeps/checkdeps.py --root=. talk
> buildtools/checkdeps/checkdeps.py --root=. webrtc
>
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/086f851b7b9b4bcbd4fe507c3bf83b760bd7f4d9
> Cr-Commit-Position: refs/heads/master@{#12008}

TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5623

Review URL: https://codereview.webrtc.org/1808573002

Cr-Commit-Position: refs/heads/master@{#12009}
2016-03-16 00:41:04 +00:00
086f851b7b Add check_deps rules in DEPS files.
Add fine-grained check_deps rules for all of WebRTC.
This will help both maintaining sane dependencies and provides a way
to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.

Example:
buildtools/checkdeps/graphdeps.py --root=. --format=png \
--out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
--excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'

will produce a neat webrtc.png image showcasing the dependencies
(according to the DEPS file) for the bitrate_controller module.
Some dependencies are filtered out for readability.

BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc

R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1796413002 .

Cr-Commit-Position: refs/heads/master@{#12008}
2016-03-16 00:22:53 +00:00
776593b139 Reland: Drop the 16kHz sample rate restriction on AECM and zero out higher bands
Landed originally here: https://codereview.webrtc.org/1774553002/
Revertede here: https://codereview.webrtc.org/1781893002/

TBR=solenberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1777093004

Cr-Commit-Position: refs/heads/master@{#12005}
2016-03-15 21:05:05 +00:00
6021fe2b1e Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1803923003

Cr-Commit-Position: refs/heads/master@{#12003}
2016-03-15 18:41:58 +00:00
e50872be13 Remove unused method OutputMixer::PlayDtmfTone() and infrastructure.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1796183002

Cr-Commit-Position: refs/heads/master@{#11990}
2016-03-14 22:32:53 +00:00
1122dc0d9b Relanding https://codereview.webrtc.org/1715883002/ in pieces.
- Remove unused callback OnPlayTelephoneEvent from voe::Channel.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1804523002

Cr-Commit-Position: refs/heads/master@{#11984}
2016-03-14 18:52:33 +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
b2a24ecf44 Relanding https://codereview.webrtc.org/1715883002/ in pieces.
- Clean up unused methods in voe::Channel following removal of VoEDtmf APIs.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1785643006

Cr-Commit-Position: refs/heads/master@{#11976}
2016-03-14 10:25:17 +00:00
b25345ee3f Replace scoped_ptr with unique_ptr in webrtc/call/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1789903003

Cr-Commit-Position: refs/heads/master@{#11970}
2016-03-12 14:10:53 +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
dfc2870380 Revert of Drop the 16kHz sample rate restriction on AECM and zero out higher bands (patchset #3 id:40001 of https://codereview.webrtc.org/1774553002/ )
Reason for revert:
Breaks Android it looks like.
See your own try jobs and
https://build.chromium.org/p/client.webrtc/builders/Android32%20Tests%20%28L%...

Original issue's description:
> Drop the 16kHz sample rate restriction on AECM and zero out higher bands
>
> The restriction has been removed completely and AECM now supports any
> number of higher bands. But this has been achieved by always zeroing out the
> higher bands, instead of applying a constant gain which is the average over half
> of the lower band (like it is done for the AEC), because that would be
> non-trivial to implement and we don't want to spend too much time on AECM, since
> we want to get rid of it in the long term anyway.
>
> R=peah@webrtc.org, solenberg@webrtc.org, tina.legrand@webrtc.org
>
> Committed: https://crrev.com/f687d53aabee0523ce6e9e0636163af8df120e41
> Cr-Commit-Position: refs/heads/master@{#11931}

TBR=peah@webrtc.org,turaj@webrtc.org,tina.legrand@webrtc.org,solenberg@webrtc.org,aluebs@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1781893002

Cr-Commit-Position: refs/heads/master@{#11932}
2016-03-10 00:23:32 +00:00
f687d53aab Drop the 16kHz sample rate restriction on AECM and zero out higher bands
The restriction has been removed completely and AECM now supports any
number of higher bands. But this has been achieved by always zeroing out the
higher bands, instead of applying a constant gain which is the average over half
of the lower band (like it is done for the AEC), because that would be
non-trivial to implement and we don't want to spend too much time on AECM, since
we want to get rid of it in the long term anyway.

R=peah@webrtc.org, solenberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1774553002 .

Cr-Commit-Position: refs/heads/master@{#11931}
2016-03-09 15:38:09 +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
622d8950f5 Remove the VoEDtmf interface.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1723153002

Cr-Commit-Position: refs/heads/master@{#11906}
2016-03-08 12:11:00 +00:00
0e73934694 Remove webrtc/test/webrtc_test_common.gyp
Move the "webrtc_test_common" target to test.gyp and rename
it to "test_common".

Move all tests in "webrtc_test_common_unittests" (which
wasn't run on the bots) into "test_support_unittests".

NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1754593002

Cr-Commit-Position: refs/heads/master@{#11848}
2016-03-02 18:46:25 +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
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
3dd5d1d84a Remove PacketRouter sender distinction.
Instead relies on SetSendingMediaStatus() to filter out receiving RTP
modules. This status is now set in VoiceEngine's SetSend() for senders
along with SetSendingStatus().

BUG=
R=solenberg@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1705763002 .

Cr-Commit-Position: refs/heads/master@{#11768}
2016-02-25 15:56:58 +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
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
b7f89d6e66 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/
Also introduce a pair of scoped_ptr <-> unique_ptr conversion
functions. By using them judiciously, we can keep these CL:s small and
avoid having to convert enormous amounts of code at once.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1702983002

Cr-Commit-Position: refs/heads/master@{#11658}
2016-02-17 18:04:26 +00:00