3285 Commits

Author SHA1 Message Date
Alejandro Luebs
4458d09ee4 Drop support for playing output through aplay in intelligibility_proc
It was hardly used, making the code more complex than needed and caused problems on iOS because it uses system.

BUG=webrtc:5549
R=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1708353002 .

Cr-Commit-Position: refs/heads/master@{#11677}
2016-02-19 03:16:17 +00:00
Zeke Chin
b3fb71c101 Add RTCAudioSession proxy class.
BUG=
R=haysc@webrtc.org, henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1709853002 .

Cr-Commit-Position: refs/heads/master@{#11676}
2016-02-18 23:44:17 +00:00
kjellander
9ac4df1ba6 iOS: Enable modules_unittests and common_audio_unittests
Instead of excluding the whole test binaries, only exclude the parts that cause the
compilation to fail for modules_unittests and common_audio_unittests.

BUG=webrtc:4752, webrtc:4755, webrtc:5544
TESTED=Successful build with:
GYP_DEFINES='OS=ios target_arch=x64' webrtc/build/gyp_webrtc
ninja -C out/Debug-iphonesimulator modules_unittests common_audio_unittests
NOTRY=True

Review URL: https://codereview.webrtc.org/1698033002

Cr-Commit-Position: refs/heads/master@{#11675}
2016-02-18 21:15:17 +00:00
kjellander@webrtc.org
66a99283be Roll chromium_revision 1d144ca..fa5d546 (375480:376142)
* Disable iOS warnings triggered by moving from ios_deployment_target 7.0 to 9.0
(see 1d144ca..fa5d546/build/common.gypi)
* Fix errors that will fail when MSVS 2015 is rolled in (coming soon).
* Start using sysroot for building on Linux since http://crbug.com/561584 has been fixed.

Change log: 1d144ca..fa5d546
Full diff: 1d144ca..fa5d546

Changed dependencies:
* src/third_party/libyuv: 903c91c..20343f4
* src/tools/gyp: 2f9ffdc..ed163ce
DEPS diff: 1d144ca..fa5d546/DEPS

No update to Clang.

TBR=
BUG=webrtc:5549
NOTRY=True

Review URL: https://codereview.webrtc.org/1713493002 .

Cr-Commit-Position: refs/heads/master@{#11673}
2016-02-18 19:30:25 +00:00
aleungbroadsoft
0e2e50ca1c Always append the BYE packet type at the end
When composing a RTCP packet, if there is a BYE
to be appended, preserve it and append it at the
end after all other packet types are added.

BUG=webrtc:5498
NOTRY=true

Review URL: https://codereview.webrtc.org/1674963004

Cr-Commit-Position: refs/heads/master@{#11672}
2016-02-18 16:33:33 +00:00
Stefan Holmer
3ee73a59ad Make RemoteBitrateEstimator::GetStats() virtual.
Should have been added in 59c634b605.

R=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1713463002 .

Cr-Commit-Position: refs/heads/master@{#11667}
2016-02-18 10:42:40 +00:00
Stefan Holmer
59c634b605 Re-add RemoteBitrateEstimator::GetStats.
Will be kept around until implementations have been updated.

This fixes build issues in dependent code caused by removing GetStats in 62a5ccdb53

R=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1709673003 .

Cr-Commit-Position: refs/heads/master@{#11664}
2016-02-18 09:14:55 +00:00
Alejandro Luebs
32348192cc Fix and simplify the power estimation in the IntelligibilityEnhancer
R=henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://codereview.webrtc.org/1685703004 .

Cr-Commit-Position: refs/heads/master@{#11663}
2016-02-18 04:04:25 +00:00
kwiberg
dabf07f477 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/vad/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1699003002

Cr-Commit-Position: refs/heads/master@{#11657}
2016-02-17 15:59:51 +00:00
Stefan Holmer
789ba92e14 Simplify CongestionController.
- Removes the dependency on CallStats.
- Implements Module interface so that we don't have to register
  each internal component to the process thread separately.

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

Review URL: https://codereview.webrtc.org/1704983002 .

Cr-Commit-Position: refs/heads/master@{#11655}
2016-02-17 14:52:25 +00:00
kwiberg
62eaacf5ee Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/test/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1694423002

Cr-Commit-Position: refs/heads/master@{#11653}
2016-02-17 14:39:13 +00:00
kjellander
28c99bc44a iOS: Include legacy objc API in all.gyp + fix H264 libyuv dependency
The legacy objc API is not included in the GYP generation if include_tests=0.
This causes problems downstream in some cases, so it's changed in this CL.

The libyuv dependency needs to be possible to disable using the build_libyuv
GYP variable.

NOTRY=True

Review URL: https://codereview.webrtc.org/1705733002

Cr-Commit-Position: refs/heads/master@{#11652}
2016-02-17 13:38:35 +00:00
danilchap
69e59e619a [rtp_rtcp] rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer
rtcp::RawPacket is rtc::Buffer, it had no extra functionality.
rtc::Buffer is a movable class - no point to wrap it into rtc::scoped_ptr
change is large, but straightforward:
  rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer
  ->Buffer() replaced with .data()
  ->Length() replaced with .size()

BUG=webrtc:5260

Review URL: https://codereview.webrtc.org/1696203002

Cr-Commit-Position: refs/heads/master@{#11649}
2016-02-17 11:11:50 +00:00
peah
a332e2d3af Added boilerplate code for being able to test the upcoming
AEC functionality.

BUG=webrtc:5201

Review URL: https://codereview.webrtc.org/1700703005

Cr-Commit-Position: refs/heads/master@{#11647}
2016-02-17 09:11:24 +00:00
kjellander
0206000a66 iOS: Add resource files for tests and implement OutputPath
With this change the following tests have been successfully
passing in the iOS Simulator for iPhone 5 and iOS 9:
* audio_decoder_unittests
* common_video_unittests
* modules_tests
* rtc_api_objc_tests
* rtc_pc_unittests
* system_wrappers_unittests
* voice_engine_unittests

The modules_unittests and common_audio_unittests are
handled in https://codereview.webrtc.org/1698033002/

BUG=webrtc:4755
NOTRY=True

Review URL: https://codereview.webrtc.org/1694353003

Cr-Commit-Position: refs/heads/master@{#11646}
2016-02-17 06:06:17 +00:00
kwiberg
85d8bb025a Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/transient/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1698843003

Cr-Commit-Position: refs/heads/master@{#11645}
2016-02-17 04:39:43 +00:00
henrik.lundin
2c38c20e7b Fix out-of-buffer write in iLBC
In some cases, the decoder can write outside of an allocated array. See
the new comment in the code for more details.

BUG=chromium:568885, webrtc:5305

Review URL: https://codereview.webrtc.org/1704463002

Cr-Commit-Position: refs/heads/master@{#11641}
2016-02-16 18:01:57 +00:00
Stefan Holmer
44c65e9eed Enable adaptive threshold experiment by default.
BUG=webrtc:4711
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1682403002 .

Cr-Commit-Position: refs/heads/master@{#11640}
2016-02-16 17:22:30 +00:00
Peter Boström
9d0c43242b Remove video-codec max bitrate from TMMBN.
TMMBN was capped by configured max bitrate for no apparent reason.
Removing this to not require payload-type reconfiguration on new
video-codec settings. Actual removal of payload-type reconfiguration
will happen in a pending CL.

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

Review URL: https://codereview.webrtc.org/1702043002 .

Cr-Commit-Position: refs/heads/master@{#11639}
2016-02-16 16:59:36 +00:00
henrik.lundin
ee31f0a7d5 Fix out-of-buffer read in iLBC
In some cases, the decoder can read outside of an allocated array. See
the new comment in the code for more details.

BUG=chromium:568889, webrtc:5305

Review URL: https://codereview.webrtc.org/1700973002

Cr-Commit-Position: refs/heads/master@{#11637}
2016-02-16 16:42:15 +00:00
Stefan Holmer
62a5ccdb53 Update bitrate only when we have incoming packet.
Also cleans up some unused code and makes sure the min bitrate of the BWE can't be set to anything lower than 10 kbps.

BUG=webrtc:5474
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1699903003 .

Cr-Commit-Position: refs/heads/master@{#11636}
2016-02-16 16:07:31 +00:00
peah
58cf5f14ec Changed order of events when synthesizing a call.
This is needed when synthesizing a call based on
48 kHz audio files as otherwise an error is
generated about the wrong sample rate is generated.
That error is in turned caused by the sample rate
being changed from the default 16 kHz
at the first Capture API call event.

BUG=

Review URL: https://codereview.webrtc.org/1698243003

Cr-Commit-Position: refs/heads/master@{#11635}
2016-02-16 15:26:25 +00:00
Peter Boström
0453ef857f Prevent busy-looping PacedSender on small packets.
Skip accounting for small packets and suspend the prober if no
large-enough packets have been sent for some time. This especially seems
to have triggered in audio-only calls where all packets are too small,
making TimeUntilNextProbe return 0 forever, causing the module process
thread to wake up forever.

BUG=webrtc:5506
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1688703002 .

Cr-Commit-Position: refs/heads/master@{#11634}
2016-02-16 15:23:16 +00:00
kwiberg
16c5a96630 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1694073002

Cr-Commit-Position: refs/heads/master@{#11622}
2016-02-15 10:27:34 +00:00
kwiberg
3747838afb Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/test/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1695763004

Cr-Commit-Position: refs/heads/master@{#11618}
2016-02-15 04:41:04 +00:00
kwiberg
2d0c33277c Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/neteq/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1697823002

Cr-Commit-Position: refs/heads/master@{#11616}
2016-02-14 17:28:39 +00:00
kwiberg
91d9756bc6 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/codecs/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1696853004

Cr-Commit-Position: refs/heads/master@{#11613}
2016-02-14 09:10:08 +00:00
peah
be61562bfb Moved the GainControlForNewAGC class to be a separate file.
Apart from being motivated in order to make the source files shorter
this is needed when separating the APM submodules structs into
separate files.

BUG=

Review URL: https://codereview.webrtc.org/1678813002

Cr-Commit-Position: refs/heads/master@{#11611}
2016-02-14 00:40:53 +00:00
henrika
e78765bd4b Removes Nexus 5 from AEC and NS blacklists
BUG=b/27086464
R=tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1695713002 .

Cr-Commit-Position: refs/heads/master@{#11605}
2016-02-12 15:33:44 +00:00
Peter Boström
b72dada927 Remove Reset from conditionally-compiled decoders.
Reset is no longer called but removal of them was missed in a previous
commit.

BUG=webrtc:5475
TBR=hbos@webrtc.org,tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1690193003 .

Cr-Commit-Position: refs/heads/master@{#11603}
2016-02-12 14:57:03 +00:00
aluebs
73e2373af2 Fix negative shift exponent in WPDTree
BUG=webrtc:5510

Review URL: https://codereview.webrtc.org/1685993003

Cr-Commit-Position: refs/heads/master@{#11585}
2016-02-11 19:03:04 +00:00
Alejandro Luebs
66d2481106 Fix division by zero errors in IntelligibilityEnhancer
BUG=webrtc:5509
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1686333002 .

Cr-Commit-Position: refs/heads/master@{#11584}
2016-02-11 18:37:23 +00:00
danilchap
9f35d55c58 Added accessor and Parse function.
Create function merged into one.

BUG=webrtc:5260

Review URL: https://codereview.webrtc.org/1439553003

Cr-Commit-Position: refs/heads/master@{#11581}
2016-02-11 16:19:06 +00:00
henrik.lundin
07c51e3f79 Fix two UBSan warnings in NetEq
Both were related to very large jumps in RTP timestamps.

BUG=webrtc:5488

Review URL: https://codereview.webrtc.org/1685103002

Cr-Commit-Position: refs/heads/master@{#11569}
2016-02-11 11:35:51 +00:00
Alejandro Luebs
42d8aa77d8 Fix IntelligibilityEnhancerTest.TestRenderUpdate test
TBR=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1685293002 .

Cr-Commit-Position: refs/heads/master@{#11561}
2016-02-11 00:10:37 +00:00
aluebs
c466badd86 Using the NS noise estimate for the IE
Review URL: https://codereview.webrtc.org/1672343002

Cr-Commit-Position: refs/heads/master@{#11559}
2016-02-10 20:03:05 +00:00
jbauch
541f1869ca Cleanup temporary files created by tests.
This CL removes some temporary files created by OptionsFileTest and
TransientFileUtilsTest.

BUG=

Review URL: https://codereview.webrtc.org/1688553002

Cr-Commit-Position: refs/heads/master@{#11554}
2016-02-10 17:10:00 +00:00
peah
1147b75b52 Moved buffering of farend into the EchoSubtraction method.
This makes sense since the buffered data is only used by
the echo subtraction method. Furthermore, it simplifies the
upcoming modifications to the echo subtraction method since
the way the buffering is done can then be specific for the
echo subtraction implementation used.

The change is bitexact and this was verified using a fairly
extensive bitexactness suite.

BUG=

Review URL: https://codereview.webrtc.org/1639773002

Cr-Commit-Position: refs/heads/master@{#11547}
2016-02-10 11:55:38 +00:00
henrik.lundin
6608d9a1aa NetEq: Fix a negative shift value
In some rare occations (very low energy signal), a shift value happened
to be negative. This is now fixed by using the WEBRTC_SPL_SHIFT_W32,
which in essence checks the sign of the number of shifts and performs a
right or left shift accordingly.

The fix reverts to how the code was written in old NetEq; see
4d363ae305/webrtc/modules/audio_coding/neteq/normal.c (165).

BUG=webrtc:5490

Review URL: https://codereview.webrtc.org/1675293002

Cr-Commit-Position: refs/heads/master@{#11546}
2016-02-10 10:47:56 +00:00
sergeyu
cc9669c6b8 Cleanup shared memory handling in DesktopCapturer interface.
Previously shared memory buffers for DesktopCapturer were created
using DesktopCapturer::Callback::CreateSharedBuffer(). That made it
difficult to proxy DesktopCapturer interface from one thread to another.
This CL adds SharedBufferFactory interface that's allowed to be called
on a background thread. This also simplifies clients that don't
need to use shared memory, as they no longer need to override
CreateSharedBuffer().

Review URL: https://codereview.webrtc.org/1678073003

Cr-Commit-Position: refs/heads/master@{#11543}
2016-02-09 23:13:32 +00:00
Alejandro Luebs
fa639f0bb3 Surface the noise estimate of the NS to be used by other components
R=henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://codereview.webrtc.org/1654443004 .

Cr-Commit-Position: refs/heads/master@{#11541}
2016-02-09 19:24:51 +00:00
kjellander
78ddd733b0 Update path for audioproc_debug proto output.
This will make it align with protoc tools that use the relative
path from the project root to the files in the output path.
Having this, no hacks will need to be applied downstream.

TBR=henrik.lundin@webrtc.org
NOTRY=True

Review URL: https://codereview.webrtc.org/1673263002

Cr-Commit-Position: refs/heads/master@{#11540}
2016-02-09 16:13:16 +00:00
kjellander
4bba35f735 Switch third_party/gflags to use updated GitHub repo.
This pulls in several fixes and gets Visual Studio 2015 support.
The new repo is located at https://github.com/gflags/gflags
which is mirrored in Chrome infrastructure at
https://chromium.googlesource.com/external/github.com/gflags/gflags

New configuration headers were generated according to README.webrtc
on Windows and Linux. I verified the Linux generated ones are working
on Mac. The generating headers on Mac are identical with only a minor
difference (an __unused attribute) that doesn't effect the build.

BUG=webrtc:5185
NOTRY=True
NOPRESUBMIT=True
TESTED=Successfully ran:
out/Release/video_quality_measurement --input_filename=resources/foreman_cif.yuv  --width=352 --height=288
to verify flags are still being parsed properly.
I also ran the compile trybots and the baremetal bots
(since they run tests that have gflags flags).

Review URL: https://codereview.webrtc.org/1679263002

Cr-Commit-Position: refs/heads/master@{#11539}
2016-02-09 14:47:47 +00:00
danilchap
09fef9e6f7 [rtp_rtcp] Added Sender Report Request rtcp packet.
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1555543005

Cr-Commit-Position: refs/heads/master@{#11538}
2016-02-09 13:57:56 +00:00
hbos
0715a83a07 Avoid OpenH264 encoder bug for #threads > 1 on Mac and Chromium+Sandbox.
Until the bug has been further investigated, we're limiting the number
of threads to 1 to avoid problems. See crbug.com/583348.

BUG=chromium:500605, chromium:468365, chromium:583348

Review URL: https://codereview.webrtc.org/1677543002

Cr-Commit-Position: refs/heads/master@{#11536}
2016-02-09 10:34:34 +00:00
henrik.lundin
e594213a2b Fix div-by-0 in NetEq's StatisticsCalculator
If a StatisticsCalculator::PeriodicUmaAverage object was created and
then deleted without any samples being logged, the destructor would call
the Metric() method, which calculated sum_/counter_. However, with no
samples logged, counter_ is 0.

This was found and verified using UBSan tests; see the bug for more info.

BUG=webrtc:5490
R=ivoc@webrtc.org

Review URL: https://codereview.webrtc.org/1678773003

Cr-Commit-Position: refs/heads/master@{#11534}
2016-02-09 08:36:02 +00:00
jbauch
d2a22960c3 Enable cpplint for webrtc/modules/pacing and fix all uncovered cpplint errors.
This CL enableds cpplint for webrtc/modules/pacing.

BUG=webrtc:5460
NOTRY=true

TESTED=Fixed issues reported by:
find webrtc/modules/pacing -type f -name *.cc -o -name *.h | xargs cpplint.py
followed by 'git cl presubmit'.

Review URL: https://codereview.webrtc.org/1674423002

Cr-Commit-Position: refs/heads/master@{#11531}
2016-02-09 07:18:30 +00:00
Peter Boström
c0ae305a9e Fix null-pointer dereference in RTPSenderVideo.
Since the address of the dereference is taken this inputs a garbage
almost-null pointer into RtpPacketizer. Not likely that a load/store is
performed on the address, but UBSan fires and it's a source of potential
future errors.

BUG=webrtc:5124, webrtc:5490
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1677003002 .

Cr-Commit-Position: refs/heads/master@{#11528}
2016-02-08 14:00:22 +00:00
Stefan Holmer
58c664c13d Clean up of CongestionController.
Removes unused methods and moves out ViERemb to Call.

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

Review URL: https://codereview.webrtc.org/1663413003 .

Cr-Commit-Position: refs/heads/master@{#11527}
2016-02-08 13:31:53 +00:00
danilchap
7336eeb690 [rtp_rtcp] rtcp::Tmmbn cleaned and got Parse function
Added accessor and Parse function
removed dependencies on structures from rtcp_utility.h (except RtcpCommonHeader)
removed limitation of 50 items per TMMBN.

BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1670973002

Cr-Commit-Position: refs/heads/master@{#11524}
2016-02-08 11:35:20 +00:00