Commit Graph

193 Commits

Author SHA1 Message Date
14d5dbe5b3 Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
The breaking tests in Chromium have been temporarily disabled, they will be fixed and reenabled soon.

Original CLs: https://codereview.webrtc.org/1748403002/, https://codereview.webrtc.org/2107253002/ and https://codereview.webrtc.org/2106103003/.

TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org,tkchin@webrtc.org
BUG=webrtc:4741, webrtc:5603, chromium:609749

Review-Url: https://codereview.webrtc.org/2110113003
Cr-Commit-Position: refs/heads/master@{#13379}
2016-07-04 14:07:03 +00:00
9e03c3b372 Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.

Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}

TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741,webrtc:5603,chromium:609749

Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
2016-06-30 07:59:49 +00:00
1895526c61 Move RtcEventLog object from inside VoiceEngine to Call.
In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.

BUG=webrtc:4741,webrtc:5603,chromium:609749
R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1748403002 .

Cr-Commit-Position: refs/heads/master@{#13321}
2016-06-29 11:57:01 +00:00
685440be2d Make mixing test die earlier on failure instead of spamming errors
This test currently takes 288 seconds to fail if output values are
wrong; there's no point to print the failure hundreds of times.
This change will exit the test early.

R=henrika@webrtc.org
BUG=623538
NOTRY=true

Review-Url: https://codereview.webrtc.org/2097363002
Cr-Commit-Position: refs/heads/master@{#13295}
2016-06-27 14:18:41 +00:00
fc3a8ee47b Delete unused code.
* Unused audio_coding and video_coding test code.
* Obsolete voice_engine android test app.
* Left-over placeholder files for remoteaudiotrack and
  portallocatorfactory.

In addition, change modules.gyp dependency from rtc_base to
rtc_base_approved.

BUG=
R=henrik.lundin@webrtc.org, henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/2065353002 .

Cr-Commit-Position: refs/heads/master@{#13166}
2016-06-16 13:51:40 +00:00
60c4e0ae8f Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2018553002/ )
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.

TBR=kjellander@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/2014973002 .

Cr-Commit-Position: refs/heads/master@{#12925}
2016-05-26 19:35:43 +00:00
b4ff7a7a8b Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2014183003/ )
Reason for revert:
Alas, the bot magically fails again.  Looks like indeed it's a Clang bug :(

https://build.chromium.org/p/client.webrtc/builders/Win64%20Debug%20%28Clang%29/builds/1687

Original issue's description:
> Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2006313009/ )
>
> Reason for revert:
> Relanding.  The errors look like legit disk space problems so I'm going to watch the bots this time.
>
> Original issue's description:
> > Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2009253004/ )
> >
> > Reason for revert:
> > Fails unexpectedly on multiple commit bots:
> > https://build.chromium.org/p/client.webrtc/builders/Win32%20Debug%20%28Clang%29/builds/1748
> > https://build.chromium.org/p/client.webrtc/builders/Win64%20Debug%20%28Clang%29/builds/1683
> >
> > I filed https://bugs.chromium.org/p/chromium/issues/detail?id=614967 to track the problem. I'll reland if it doesn't solve the problem.
> >
> > Original issue's description:
> > > Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2006243002/ )
> > >
> > > 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
> > > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > > BUG=chromium:613482
> > >
> > > Committed: https://crrev.com/ba189cc4f4f6fe311a815646059e8011ffa53894
> > > Cr-Commit-Position: refs/heads/master@{#12907}
> >
> > TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=chromium:613482
> >
> > Committed: https://crrev.com/6895d8ca788df29f4d3a4b97fe891f0fb3a6dbec
> > Cr-Commit-Position: refs/heads/master@{#12909}
>
> TBR=henrik.lundin@webrtc.org,kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=chromium:613482
>
> Committed: https://crrev.com/b7318f1293453d8d319e27473c5bfb72292ebfd2
> Cr-Commit-Position: refs/heads/master@{#12920}

TBR=henrik.lundin@webrtc.org,kjellander@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/2018553002
Cr-Commit-Position: refs/heads/master@{#12921}
2016-05-26 15:27:43 +00:00
b7318f1293 Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2006313009/ )
Reason for revert:
Relanding.  The errors look like legit disk space problems so I'm going to watch the bots this time.

Original issue's description:
> Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2009253004/ )
>
> Reason for revert:
> Fails unexpectedly on multiple commit bots:
> https://build.chromium.org/p/client.webrtc/builders/Win32%20Debug%20%28Clang%29/builds/1748
> https://build.chromium.org/p/client.webrtc/builders/Win64%20Debug%20%28Clang%29/builds/1683
>
> I filed https://bugs.chromium.org/p/chromium/issues/detail?id=614967 to track the problem. I'll reland if it doesn't solve the problem.
>
> Original issue's description:
> > Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2006243002/ )
> >
> > 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
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=chromium:613482
> >
> > Committed: https://crrev.com/ba189cc4f4f6fe311a815646059e8011ffa53894
> > Cr-Commit-Position: refs/heads/master@{#12907}
>
> TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=chromium:613482
>
> Committed: https://crrev.com/6895d8ca788df29f4d3a4b97fe891f0fb3a6dbec
> Cr-Commit-Position: refs/heads/master@{#12909}

TBR=henrik.lundin@webrtc.org,kjellander@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/2014183003
Cr-Commit-Position: refs/heads/master@{#12920}
2016-05-26 15:17:29 +00:00
6895d8ca78 Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2009253004/ )
Reason for revert:
Fails unexpectedly on multiple commit bots:
https://build.chromium.org/p/client.webrtc/builders/Win32%20Debug%20%28Clang%29/builds/1748
https://build.chromium.org/p/client.webrtc/builders/Win64%20Debug%20%28Clang%29/builds/1683

I filed https://bugs.chromium.org/p/chromium/issues/detail?id=614967 to track the problem. I'll reland if it doesn't solve the problem.

Original issue's description:
> Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2006243002/ )
>
> 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
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:613482
>
> Committed: https://crrev.com/ba189cc4f4f6fe311a815646059e8011ffa53894
> Cr-Commit-Position: refs/heads/master@{#12907}

TBR=henrik.lundin@webrtc.org,tommi@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/2006313009
Cr-Commit-Position: refs/heads/master@{#12909}
2016-05-26 09:46:59 +00:00
ba189cc4f4 Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2006243002/ )
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
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:613482

Review-Url: https://codereview.webrtc.org/2009253004
Cr-Commit-Position: refs/heads/master@{#12907}
2016-05-26 09:13:14 +00:00
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
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
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
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
d8de1154c9 Remove mutable from rtc::CriticalSections.
A couple of mutables were added after last removal of mutables, so
removing those. rtc::CriticalSection is const-lockable.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1652983002

Cr-Commit-Position: refs/heads/master@{#11447}
2016-02-01 17:00:59 +00:00
31fc21f454 Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/
Also remove mischievous tab character!
This is a part of getting rid of CriticalSectionWrapper and makes the code slightly simpler.

BUG=

Review URL: https://codereview.webrtc.org/1607353002

Cr-Commit-Position: refs/heads/master@{#11346}
2016-01-21 18:37:44 +00:00
6955870806 Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1316523002 .

Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-13 00:26:55 +00:00
e2976c87f7 Remove DISABLED_ON_ macros.
Macro incorrectly displays DISABLED_ON_ANDROID in test names for
parameterized tests under --gtest_list_tests, causing tests to be
disabled on all platforms since they contain the DISABLED_ prefix rather
than their expanded variants.

This expands the macro variants to inline if they're disabled or not,
and removes building some tests under configurations where they should
fail, instead of building them but disabling them by default.

The change also removes gtest_disable.h as an unused include from many
other files.

BUG=webrtc:5387, webrtc:5400
R=kjellander@webrtc.org, phoglund@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1547343002 .

Cr-Commit-Position: refs/heads/master@{#11150}
2016-01-04 21:44:16 +00:00
44307630d3 AudioCodingModuleImpl: Stop failing artificially for non-Opus encoders
All encoders already handle the "Opus-specific" requests sanely (by
failing nicely), so we don't need extra checks to protect them.

BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1527453005

Cr-Commit-Position: refs/heads/master@{#11051}
2015-12-16 14:24:09 +00:00
84e78f9102 Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/.
Created a simple unit test for the new random number generator. (It mostly tests
that the generated numbers are consistent with the intended distribution, e.g. uniform.
It is not a comprehensive test of the quality of the random numbers.)

Several assertions in OveruseDetectorTest seem to depend on the exact sequence of random numbers. I updated those numbers to work with the new PRNG.

Compute the standard deviation of the expected result in TestReorderFilter instead of passing an uncertainty parameter.

BUG=webrtc:5177

Review URL: https://codereview.webrtc.org/1457023002

Cr-Commit-Position: refs/heads/master@{#10965}
2015-12-10 09:51:02 +00:00
8c38e8b9b9 Clean up PlatformThread.
* Move PlatformThread to rtc::.
* Remove ::CreateThread factory method.
* Make non-scoped_ptr from a lot of invocations.
* Make Start/Stop void.
* Remove rtc::Thread priorities, which were unused and would collide.
* Add ::IsRunning() to PlatformThread.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1476453002 .

Cr-Commit-Position: refs/heads/master@{#10812}
2015-11-26 16:45:57 +00:00
12411ef40e Move ThreadWrapper to ProcessThread in base.
Also removes all virtual methods. Permits using a thread from
rtc_base_approved (namely event tracing).

BUG=webrtc:5158
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1469013002

Cr-Commit-Position: refs/heads/master@{#10760}
2015-11-23 22:48:01 +00:00
1323fc39ba Remove webrtc/test/channel_transport/include
Move the header file into webrtc/test/channel_transport instead.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel -m tryserver.webrtc --bot=ios_rel
R=henrika@webrtc.org, henrikg@webrtc.org

Review URL: https://codereview.webrtc.org/1431983006 .

Cr-Commit-Position: refs/heads/master@{#10595}
2015-11-11 09:34:35 +00:00
3cea256806 Reland "Prevent Opus DTX from generating intermittent noise during silence"
The original CL is reviewed at
https://codereview.webrtc.org/1415173005/

A silly mistake was made at the last patch set, and the CL was reverted. This CL is to fix and reland it.

BUG=

Review URL: https://codereview.webrtc.org/1422213003

Cr-Commit-Position: refs/heads/master@{#10574}
2015-11-10 11:49:32 +00:00
b4a753fdb5 Revert of Prevent Opus DTX from generating intermittent noise during silence (patchset #10 id:250001 of https://codereview.webrtc.org/1415173005/ )
Reason for revert:
Breaks voe_auto_test on all three "large tests bots".
https://build.chromium.org/p/client.webrtc/builders/Win32%20Release%20%5Blarge%20tests%5D/builds/5630/steps/voe_auto_test/logs/stdio
https://build.chromium.org/p/client.webrtc/builders/Mac32%20Release%20%5Blarge%20tests%5D/builds/5599/steps/voe_auto_test/logs/stdio
https://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/5645/steps/voe_auto_test/logs/stdio

Notice these bots are no longer a part of the default trybot set, so they have to be run manually when working with code that affects their tests (they were removed as they queued up all the time in the CQ, and usually don't catch breakages).

Original issue's description:
> Prevent Opus DTX from generating intermittent noise during silence.
>
> Opus may have an internal error that causes this. Here we make a workaround by adding some small disturbance to the input signals to break a long sequence of zeros.
>
> BUG=webrtc:5127
>
> Committed: https://crrev.com/f475add57eada116bc960fe2935876ec8c077977
> Cr-Commit-Position: refs/heads/master@{#10565}

TBR=tina.legrand@webrtc.org,kwiberg@webrtc.org,solenberg@webrtc.org,minyue@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5127

Review URL: https://codereview.webrtc.org/1428613004

Cr-Commit-Position: refs/heads/master@{#10567}
2015-11-09 21:27:11 +00:00
f475add57e Prevent Opus DTX from generating intermittent noise during silence.
Opus may have an internal error that causes this. Here we make a workaround by adding some small disturbance to the input signals to break a long sequence of zeros.

BUG=webrtc:5127

Review URL: https://codereview.webrtc.org/1415173005

Cr-Commit-Position: refs/heads/master@{#10565}
2015-11-09 18:08:20 +00:00
e56c763691 Removing trace checks in VoETestManager.
Trace checks in the ctor and dtor of VoETestManager are removed, since they can fail if there are more than one VoETestManager (or VoE) used in a test.

BUG=

Review URL: https://codereview.webrtc.org/1407883007

Cr-Commit-Position: refs/heads/master@{#10507}
2015-11-04 15:11:52 +00:00
ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00
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
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
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
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
5c389d3e09 Split webrtc/video into webrtc/{audio,call,video}.
Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.

BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1227923005 .

Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 11:58:39 +00:00
ac547a6538 Remove channel ids from various interfaces.
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.

IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately

BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1335353005 .

Cr-Commit-Position: refs/heads/master@{#9978}
2015-09-17 21:06:02 +00:00
f2bfc2b8ef Remove some dead code.
WebRtcPassthroughRender has been dead since webrtcvideoengine.cc was
removed, FakeExternalTransport has probably been unused for a long time.

BUG=webrtc:1695
R=henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1343393003 .

Cr-Commit-Position: refs/heads/master@{#9968}
2015-09-17 11:04:21 +00:00
91d6edef35 Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1335923002

Cr-Commit-Position: refs/heads/master@{#9964}
2015-09-17 07:24:51 +00:00
b04965ccf8 Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call.
An option was added to voe_cmd_test to make a RtcEventLog dump.

BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1267683002

Cr-Commit-Position: refs/heads/master@{#9901}
2015-09-09 07:09:49 +00:00
dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00
03bb7c7bfa Add LoudestFilter in ConferenceTransport
BUG=

Review URL: https://codereview.webrtc.org/1236793003

Cr-Commit-Position: refs/heads/master@{#9712}
2015-08-14 14:34:05 +00:00
728d9037c0 Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)

BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1172163004

Cr-Commit-Position: refs/heads/master@{#9420}
2015-06-11 21:31:48 +00:00
23c2e55479 Remove remaining .mk files.
These files are not supported, kept up to date or likely to build
anymore.

BUG=
R=glaznev@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46489004

Cr-Commit-Position: refs/heads/master@{#9303}
2015-05-28 09:05:11 +00:00
afef4bfd1c Reland "Adding a test framework for conference mode application in VoE."
"Adding a test framework for conference mode application in VoE." was wrongly committed and therefore was temporarily reverted.

This is to reland.

The CL is indifferent from its original version
https://review.webrtc.org/46249004/

TBR=phoglund@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/50109005

Cr-Commit-Position: refs/heads/master@{#9290}
2015-05-26 22:21:25 +00:00
a4b7e5e35a Revert "Adding a test framework for conference mode application in VoE."
This reverts commit fc052055e939fa93d3ab92914e0dc8ed5e5d1d90.
since it was not committed correctly.

I committed it from a wrong machine, which did not have the correct patch.

BUG=
TBR=phoglund@webrtc.org,

Review URL: https://webrtc-codereview.appspot.com/56469005

Cr-Commit-Position: refs/heads/master@{#9289}
2015-05-26 21:21:55 +00:00
fc052055e9 Adding a test framework for conference mode application in VoE.
BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46249004

Cr-Commit-Position: refs/heads/master@{#9286}
2015-05-26 19:00:50 +00:00
f09e09c7ee VoE: Remove unused interfaces
BUG=4690

I have removed methods in VoE interfaces that were marked to be removed. I have removed them also in fake and mock implementations. I have also updated the callers in various ways:
1. Project win_test had some calls to the removed methods, but it turned out that the project is not used anymore, so I removed it entirely.
2. There were some calls to removed methods in jni methods. I have removed couple of jni methods as now they seem to do nothing.
3. With the remaining callers I just removed the calls to removed methods.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53519004

Cr-Commit-Position: refs/heads/master@{#9281}
2015-05-26 08:25:00 +00:00
c3f4dbc40b Remove rtp_rtcp/ dump functionality.
Removes RTP dumping from VideoEngine and VoiceEngine.

BUG=1695
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47179004

Cr-Commit-Position: refs/heads/master@{#9234}
2015-05-20 12:10:56 +00:00
300eeb68f5 Remove VideoEngine interfaces.
Removes ViE interfaces, _impl.cc files, managers (such as
ViEChannelManager and ViEInputManager) as well as ViESharedData.

Interfaces necessary to implement observers have been moved to a
corresponding header (such as vie_channel.h).

BUG=1695, 4491
R=mflodman@webrtc.org, solenberg@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55379004

Cr-Commit-Position: refs/heads/master@{#9179}
2015-05-12 14:51:08 +00:00
79c143312b Delete VoiceChannelTransport before deleting Channel in voe_cmd_test
Current voe_cmd_test shows following error when quitting:
DeRegisterExternalTransport() failed to locate channel.

This is to fix it.

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45349004

Cr-Commit-Position: refs/heads/master@{#9129}
2015-05-04 09:21:00 +00:00
adf89b7e33 Added SetBitRate function to VoE API to allow changing the audio bitrate.
If the requested bitrate is not valid for the codec, the codec will decide on
an appropriate value.
Updated VoE command line tool to use new SetBitRate function.
Includes unittests for SetBitRate function.

BUG=
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kwiberg@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50789004

Cr-Commit-Position: refs/heads/master@{#9115}
2015-04-29 14:03:45 +00:00