5809 Commits

Author SHA1 Message Date
minyue-webrtc
12d30840d8 Correct the calculation of discard rate.
Bug: webrtc:7903
Change-Id: Ib5d6fd882a994dd542b616e5fe1c75710346dd31
Reviewed-on: https://chromium-review.googlesource.com/575057
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19101}
2017-07-20 09:15:46 +00:00
pbos
1777c5fec5 Move temporal-layer properties to FrameConfig.
Removes keying on pattern_idx inside TemporalLayers implementations for
several properties that are different between screencast temporal layers
and normal/default temporal layers.

This is a step towards sharing PopulateCodecSpecific between the layer
patterns and code deduplication to longer term be able to separate the
packetizer step from encoder settings, so that temporal patterns can be
used for asynchronous hardware encoders where there may be outstanding
frames.

BUG=chromium:702017, webrtc:7349
R=brandtr@webrtc.org

Review-Url: https://codereview.webrtc.org/2924993002
Cr-Commit-Position: refs/heads/master@{#19097}
2017-07-20 00:04:02 +00:00
ehmaldonado
f6a861ab6c Remove remains of webrtc/base
All downstream code have been updated to the new location.

In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS

Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn

BUG=webrtc:7634
TBR=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2976293002
Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 17:40:47 +00:00
saza
b4aa4eb06f Replace WEBRTC_TRACE logging in modules/audio_device/.. mac/ win/
Patch set 1:
Run a script to replace occurrences of WEBRTC_TRACE logging with the new style, on webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc.

Patch set 2:
 - Manually fix log lines not handled by the script
 - Adjust local macros that use WEBRTC_TRACE
 - Adjust some lines to conform with code style
 - Update the included headers
 - Remove the now unused object ID variables

BUG=webrtc:5118

Review-Url: https://codereview.webrtc.org/2985443002
Cr-Commit-Position: refs/heads/master@{#19088}
2017-07-19 08:12:36 +00:00
Zijie He
3e45cb577e Mapping screen id from DirectX capturer to GDI capturer
This change ensures DirectX capturer to return the same ScreenId as GDI capturer
for each monitor. So MouseCursoeMonitor can work correctly with the DirectX
capturer.

This is a temporary fix of webrtc:7950.

Bug: webrtc:7950
Change-Id: Icd3f40556701811c21c773a39260a74db43979f3
Reviewed-on: https://chromium-review.googlesource.com/571101
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19079}
2017-07-18 17:26:58 +00:00
Alex Loiko
b5c1607e92 UBSan fuzzer bug in LowCutFilter::BiqueadFilter::Process
The variable 'tmp_int32' in LowCutFilter::BiqueadFilter::Process can
be negative. This replaces a left shift with multiplication.

Bug: chromium:735593, chromium:743330
Change-Id: Idec7fbcc17495f7241eb4bea44920585740e3695
Reviewed-on: https://chromium-review.googlesource.com/575136
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19074}
2017-07-18 12:23:08 +00:00
saza
43a85f0343 Patch set 1:
Run a script to replace occurrences of WEBRTC_TRACE logging with the new
style, on webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc.

Patch set 2:
 - Manually fix log lines not handled by the script
 - Adjust some lines, to conform with code style
 - Update the included headers
 - Remove the now unused object ID variables
 -  - This explains why there's so many files edited

BUG=webrtc:5118
TBR=henrika@webrtc.org

Review-Url: https://codereview.webrtc.org/2978083002
Cr-Commit-Position: refs/heads/master@{#19071}
2017-07-18 11:12:29 +00:00
gnish
157cbbd3a7 Added implementation of three classes:
1) MaxBandwidthFilter
2) MinRttFilter
3) CongestionWindow

Added unit-tests for those classes.

BUG=webrtc:7713

Review-Url: https://codereview.webrtc.org/2966403002
Cr-Commit-Position: refs/heads/master@{#19067}
2017-07-18 09:50:22 +00:00
Zijie He
44967e41c5 Expose IsCurrentSessionSupported() from ScreenCapturerWinDirectx
IsCurrentSessionSupported() is useful to decide whether Windows version should
be used to evaluate the capability of DirectX capturer on the system.

Bug: 741926
Change-Id: Iaaf6011a9e464d7cf5e7dda097007778c73953e0
Reviewed-on: https://chromium-review.googlesource.com/571378
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19060}
2017-07-17 18:58:03 +00:00
tschumim
9d11764344 Reimplemeted "Test and fix for huge bwe drop after alr state"
BUG=webrtc:7746

Test and fix for huge bwe drop after alr state.

BUG=webrtc:7746

Review-Url: https://codereview.webrtc.org/2931873002
Cr-Commit-Position: refs/heads/master@{#18692}
Committed: 37aa8ba616

patch from issue 2931873002 at patchset 320001 (http://crrev.com/2931873002#ps320001)

Review-Url: https://codereview.webrtc.org/2970653004
Cr-Commit-Position: refs/heads/master@{#19055}
2017-07-17 08:41:41 +00:00
tommi
058aa719ff Fix incorrect DCHECK in generic_decoder.cc.
When ownership is not external, the decoder pointer should be valid.

BUG=b/63658384
TBR=magjed@webrtc.org

Review-Url: https://codereview.webrtc.org/2975383002
Cr-Commit-Position: refs/heads/master@{#19041}
2017-07-15 18:33:35 +00:00
saza
9b1367f233 Patch set 1:
Run a script to replace occurrences of WEBRTC_TRACE logging with the new
style, on webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc.

Patch set 2:
Manually fix log lines not handled by the script, remove unused header
and variable.

I would like to do this will the following files, too:
webrtc/modules/audio_device/..
.../linux/audio_device_alsa_linux.cc
.../linux/audio_device_pulse_linux.cc
.../linux/audio_mixer_manager_alsa_linux.cc
.../linux/audio_mixer_manager_pulse_linux.cc
.../linux/latebindingsymboltable_linux.cc
.../mac/audio_device_mac.cc
.../mac/audio_mixer_manager_mac.cc
.../win/audio_device_core_win.cc

BUG=webrtc:5118

Review-Url: https://codereview.webrtc.org/2978953003
Cr-Commit-Position: refs/heads/master@{#19019}
2017-07-14 12:22:33 +00:00
philipel
9bd1d66995 RtpFrameReferenceFinder VP8 reference calculation fix.
BUG=webrtc:7768

Review-Url: https://codereview.webrtc.org/2980943003
Cr-Commit-Position: refs/heads/master@{#19018}
2017-07-14 11:52:01 +00:00
zijiehe
4ff4208c21 Output DeviceName from various windows ScreenCapturer related implementations
Both DXGI_OUTPUT_DESC and DISPLAY_DEVICE contain the DeviceName, which may be
able to map a DirectX screen id with the GDI screen id.
So this change exports the field from both DirectX and GDI implementations.

BUG=webrtc:7950

Review-Url: https://codereview.webrtc.org/2971393002
Cr-Commit-Position: refs/heads/master@{#19010}
2017-07-13 23:05:24 +00:00
henrika
abcf112ae0 Adds sanity check for sample rate on iOS
BUG=b/62909493

Review-Url: https://codereview.webrtc.org/2978913002
Cr-Commit-Position: refs/heads/master@{#19000}
2017-07-13 11:42:50 +00:00
sprang
f03ea04176 Fix potential incorrect sync flags used with screenshare TL stream.
BUG=webrtc:7980

Review-Url: https://codereview.webrtc.org/2978743002
Cr-Commit-Position: refs/heads/master@{#18999}
2017-07-13 10:53:51 +00:00
zijiehe
67e438ffe5 Fix http://crbug.com/741252
The root cause is when a DxgiContext is reseted, it has not been correctly
cleared from DxgiOutputDuplicator. So during next CaptureFrame() function call,
DxgiOutputDuplicator will spread the context change to a deleted instance.

BUG=741252

Review-Url: https://codereview.webrtc.org/2975103002
Cr-Commit-Position: refs/heads/master@{#18992}
2017-07-13 00:29:36 +00:00
jianj
23173a372b vp9: Use 2 threads for 360p.
BUG=None

Review-Url: https://codereview.webrtc.org/2977843002
Cr-Commit-Position: refs/heads/master@{#18990}
2017-07-12 23:11:09 +00:00
jianj
822e593437 vp9: Turn off FRAME_PARALLEL_DECODING.
This is removed from libvpx.

BUG=None

Review-Url: https://codereview.webrtc.org/2975113002
Cr-Commit-Position: refs/heads/master@{#18989}
2017-07-12 23:09:58 +00:00
minyue-webrtc
9c6430f950 Reland "Adding ANA config event to debug dump."
A revert of the CL was made in https://chromium-review.googlesource.com/c/559429/

Now upstream conflicts have been solved, and this is ready to reland.

TBR=ossu@webrtc.org, tschumim@webrtc.org

Bug: webrtc:7854
Change-Id: Idd08f28cf26036ad91eb7b7f73b2466dd9300f76
Reviewed-on: https://chromium-review.googlesource.com/565507
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18985}
2017-07-12 12:51:37 +00:00
saza
f91b0b49f9 Set memory to all-zero floats in ExtractExtendedBlock in AEC2 if buffer is empty.
This avoids a memcopy call which may corrupt audio handling and, in rare cases, crash WebRTC with a buffer over-read.

BUG=webrtc:7845

Review-Url: https://codereview.webrtc.org/2980723002
Cr-Commit-Position: refs/heads/master@{#18984}
2017-07-12 08:43:14 +00:00
saza
10b0d03673 Removes internal buffer memory check in AEC module.
The check triggered in 30 / 1000 cases of running PeerConnectionIntegrationTest.CallTransferredForCaller locally, far more often than expected.

It will soon be replaced by more graceful handling.

BUG=webrtc:7845

Review-Url: https://codereview.webrtc.org/2975043002
Cr-Commit-Position: refs/heads/master@{#18983}
2017-07-12 07:29:36 +00:00
ilnik
29d0840b5c Reland of Refactor timing frame logic to work with encoders with internal sources (patchset #1 id:1 of https://codereview.webrtc.org/2980533002/ )
BUG=webrtc:7594,webrtc:7893

Review-Url: https://codereview.webrtc.org/2974893002
Cr-Commit-Position: refs/heads/master@{#18974}
2017-07-11 15:08:12 +00:00
tschumim
82c5593cb6 Let alr detector use a budged to detect underuse.
BUG=webrtc:7947

Review-Url: https://codereview.webrtc.org/2965233002
Cr-Commit-Position: refs/heads/master@{#18972}
2017-07-11 13:56:04 +00:00
jianjun.zhu
c024740b5e Use relative paths in GN files.
BUG=webrtc:7952
TBR=kjellander@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2974863003
Cr-Commit-Position: refs/heads/master@{#18970}
2017-07-11 13:20:45 +00:00
tommi
c45d6d9c85 Remove dependency on rtc::Thread and rtc_base from audio_mixer_unittests.
Instead, use a TaskQueue in the only test that required it.

BUG=none

Review-Url: https://codereview.webrtc.org/2975883002
Cr-Commit-Position: refs/heads/master@{#18969}
2017-07-11 13:17:10 +00:00
peah
14c11a4712 Add adaptive notch filter to remove narrowband echo components in AEC3
This CL adds detection of components in the render signal that are of
strong narrowband nature and therefore may cause problems for the AEC.
This CL also adds functionality in the echo suppressor to suppress
these signals

BUG=webrtc:7967

Review-Url: https://codereview.webrtc.org/2980493002
Cr-Commit-Position: refs/heads/master@{#18968}
2017-07-11 13:13:43 +00:00
peah
5e6685ff35 Robustification of the AEC3 echo removal in the first part of the call
This CL robustifies the echo removal in AEC3 during the initial parts
of a call in two ways:
-By extending the period until which a headset is deemed to be used.
-By increasing the assumed echo path gain for unknown echo paths at
higher frequencies.

BUG=webrtc:7971

Review-Url: https://codereview.webrtc.org/2974883002
Cr-Commit-Position: refs/heads/master@{#18967}
2017-07-11 11:19:58 +00:00
henrika
7fc3f15686 Now uses CallStaticObjectMethodV to an variable argument list argument
BUG=webrtc:7965

Review-Url: https://codereview.webrtc.org/2974913002
Cr-Commit-Position: refs/heads/master@{#18965}
2017-07-11 10:52:09 +00:00
peah
2910357621 Transparency improvements in the echo canceller 3
This CL adds two changes:
-Adaptive adjustment of the echo suppression to both cover the cases
when the echo path well covers the room, and when when it does not.
-Identification of the case when the echo is too low to be audible
and adaptive handling of this case in the echo suppression.

BUG=webrtc:7519, webrtc:7956,webrtc:7957

Review-Url: https://codereview.webrtc.org/2974583004
Cr-Commit-Position: refs/heads/master@{#18962}
2017-07-11 09:54:02 +00:00
eladalon
48956a195c Remove unused headers from remote_bitrate_estimator_single_stream.cc
BUG=None

Review-Url: https://codereview.webrtc.org/2980473003
Cr-Commit-Position: refs/heads/master@{#18958}
2017-07-10 19:46:25 +00:00
ilnik
0b1e2f3279 Revert of Refactor timing frame logic to work with encoders with internal sources (patchset #2 id:20001 of https://codereview.webrtc.org/2968153002/ )
Reason for revert:
Failing chromoting tests

Original issue's description:
> Refactor timing frame logic to work with encoders with internal sources
>
> BUG=webrtc:7594,webrtc:7893
>
> Review-Url: https://codereview.webrtc.org/2968153002
> Cr-Commit-Position: refs/heads/master@{#18955}
> Committed: a7a4535a35

TBR=sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7594,webrtc:7893

Review-Url: https://codereview.webrtc.org/2980533002
Cr-Commit-Position: refs/heads/master@{#18957}
2017-07-10 19:25:29 +00:00
Noah Richards
bc8ee33658 Remove verbose logs from audio_coding_module.cc.
PlayoutFrequency(), at least, is called ~200 times a second. The others
appear to not be in practice, but it's unclear what value they serve.

They were traces before https://chromium-review.googlesource.com/c/518133/,
which was more reasonable, as you could enable them for just audio
coding traces. But now that they are just logs, they make all VERBOSE
logging unusable.

Bug: webrtc:7959
Change-Id: I190a61c8ff4c0f047798087e80adbb41d791fc29
Reviewed-on: https://chromium-review.googlesource.com/563881
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Noah Richards <noahric@chromium.org>
Cr-Commit-Position: refs/heads/master@{#18956}
2017-07-10 17:36:28 +00:00
ilnik
a7a4535a35 Refactor timing frame logic to work with encoders with internal sources
BUG=webrtc:7594,webrtc:7893

Review-Url: https://codereview.webrtc.org/2968153002
Cr-Commit-Position: refs/heads/master@{#18955}
2017-07-10 17:03:23 +00:00
ehmaldonado
370dd47973 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
Reason for revert:
Breaks lots of downstream projects.

Original issue's description:
> Remove remains of webrtc/base
>
> All downstream code have been updated to the new location.
>
> In PRESUBMIT.py:
> * Remove webrtc/rtc_base from CPP_BLACKLIST
> * Add webrtc/rtc_base to LEGACY_API_DIRS
>
> Fix some duplicated paths in
> webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
>
> BUG=webrtc:7634
> TBR=kwiberg@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2973183002
> Cr-Commit-Position: refs/heads/master@{#18948}
> Committed:
9483b49baf

TBR=kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7634

Review-Url: https://codereview.webrtc.org/2976633002
Cr-Commit-Position: refs/heads/master@{#18949}
2017-07-10 12:58:42 +00:00
ehmaldonado
9483b49baf Remove remains of webrtc/base
All downstream code have been updated to the new location.

In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS

Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn

BUG=webrtc:7634
TBR=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2973183002
Cr-Commit-Position: refs/heads/master@{#18948}
2017-07-10 11:50:54 +00:00
eladalon
6c9556e7f1 Prevent warnings in PacketRouterTest and PacketRouterRembTest
Change MockRtpRtcp to NiceMock<MockRtpRtcp> throughout PacketRouterTest and PacketRouterRembTest (12 tests in total), to suppress a large number of warnings which are currently ignored.

BUG=None

Review-Url: https://codereview.webrtc.org/2977533002
Cr-Commit-Position: refs/heads/master@{#18946}
2017-07-10 10:33:00 +00:00
magjed
6cc25614a9 Remove webrtc::VideoEncoderFactory
Replace the use of webrtc::VideoEncoderFactory with
cricket::WebRtcVideoEncoderFactory and remove the adapter classes
between these two factory types.

Some code changes were necessary in order to accomplish this:
 * Move SimulcastEncoderAdapter from
   webrtc/modules/video_coding/codecs/vp8 to webrtc/media/engine (that's
   where it's used).
 * Rename simulcast_unittest.h to simulcast_test_utility.h and make it
   into it's own target, because it's used from both
   simulcast_unittest.cc and simulcast_encoder_adapter_unittest.cc.
 * Remove ownership of the encoder factory from SimulcastEncoderAdapter,
   and make the necessary changes in surrounding code.

The goal with this CL is to clean up the code, and also to free up
the name webrtc::VideoEncoderFactory for future use.

BUG=webrtc:7925

Review-Url: https://codereview.webrtc.org/2964953002
Cr-Commit-Position: refs/heads/master@{#18945}
2017-07-10 10:26:36 +00:00
saza
bffe597e69 Convert occurrences of deprecated WEBRTC_TRACE logging to LOG style logging in webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc.
First patch set uses a script attached in an issue comment:
https://bugs.chromium.org/p/webrtc/issues/detail?id=5118#c24
This discards the boilerplate prefix of WEBRTC_TRACE log strings, but it appears to be discarded anyway by all users.

Second patch set removes the header and makes small fixes to four of the log messages.

BUG=webrtc:5118

Review-Url: https://codereview.webrtc.org/2958273002
Cr-Commit-Position: refs/heads/master@{#18941}
2017-07-10 08:05:45 +00:00
saza
5de068082b Add a check in the BlockBuffer of AEC2 to guard for buffer overflows.
Ensure that the ring buffer does not return a pointer into the buffer if
no data is available to read.

The ring buffer fix is not directly applicable to issue webrtc:7845, but may cause related memory errors.

BUG=webrtc:7845

Review-Url: https://codereview.webrtc.org/2971313002
Cr-Commit-Position: refs/heads/master@{#18940}
2017-07-10 08:01:09 +00:00
Jonathan Yu
3ffa72d0f0 Add AudioFrame::ResetWithoutMuting() to address performance regression.
Prior to https://codereview.webrtc.org/2750783004/ Reset() intentionally
did not zero out the buffer. After that change, callers calling Reset()
and then mutable_data() were performing a wasteful zeroing.

This change adds ResetWithoutMuting() to match the old behavior and
switches the sole non-test caller of Reset() to use ResetWithoutMuting()
instead.

Prior to this change (optimized, Linux):
$ out/Default/webrtc_perf_tests --gtest_filter=NetEqPerformanceTest.Run* \
  --gtest_repeat=10 | grep neteq_performance
*RESULT neteq_performance: 10_pl_10_drift= 4051 ms
*RESULT neteq_performance: 0_pl_0_drift= 1768 ms
*RESULT neteq_performance: 10_pl_10_drift= 3666 ms
*RESULT neteq_performance: 0_pl_0_drift= 1690 ms
*RESULT neteq_performance: 10_pl_10_drift= 3685 ms
*RESULT neteq_performance: 0_pl_0_drift= 1693 ms
*RESULT neteq_performance: 10_pl_10_drift= 3720 ms
*RESULT neteq_performance: 0_pl_0_drift= 1690 ms
*RESULT neteq_performance: 10_pl_10_drift= 3780 ms
*RESULT neteq_performance: 0_pl_0_drift= 1728 ms
*RESULT neteq_performance: 10_pl_10_drift= 3733 ms
*RESULT neteq_performance: 0_pl_0_drift= 1737 ms
*RESULT neteq_performance: 10_pl_10_drift= 3781 ms
*RESULT neteq_performance: 0_pl_0_drift= 1744 ms
*RESULT neteq_performance: 10_pl_10_drift= 3712 ms
*RESULT neteq_performance: 0_pl_0_drift= 1731 ms
*RESULT neteq_performance: 10_pl_10_drift= 3681 ms
*RESULT neteq_performance: 0_pl_0_drift= 1691 ms
*RESULT neteq_performance: 10_pl_10_drift= 3681 ms
*RESULT neteq_performance: 0_pl_0_drift= 1690 ms

With this change:
$ out/Default/webrtc_perf_tests --gtest_filter=NetEqPerformanceTest.Run* \
  --gtest_repeat=10 | grep neteq_performance
*RESULT neteq_performance: 10_pl_10_drift= 3824 ms
*RESULT neteq_performance: 0_pl_0_drift= 1632 ms
*RESULT neteq_performance: 10_pl_10_drift= 3502 ms
*RESULT neteq_performance: 0_pl_0_drift= 1521 ms
*RESULT neteq_performance: 10_pl_10_drift= 3520 ms
*RESULT neteq_performance: 0_pl_0_drift= 1534 ms
*RESULT neteq_performance: 10_pl_10_drift= 3517 ms
*RESULT neteq_performance: 0_pl_0_drift= 1530 ms
*RESULT neteq_performance: 10_pl_10_drift= 3521 ms
*RESULT neteq_performance: 0_pl_0_drift= 1527 ms
*RESULT neteq_performance: 10_pl_10_drift= 3511 ms
*RESULT neteq_performance: 0_pl_0_drift= 1533 ms
*RESULT neteq_performance: 10_pl_10_drift= 3518 ms
*RESULT neteq_performance: 0_pl_0_drift= 1523 ms
*RESULT neteq_performance: 10_pl_10_drift= 3503 ms
*RESULT neteq_performance: 0_pl_0_drift= 1524 ms
*RESULT neteq_performance: 10_pl_10_drift= 3514 ms
*RESULT neteq_performance: 0_pl_0_drift= 1534 ms
*RESULT neteq_performance: 10_pl_10_drift= 3501 ms
*RESULT neteq_performance: 0_pl_0_drift= 1530 ms

BUG=webrtc:7343,chromium:738852,chromium:738839

Change-Id: Idcbb276ca0ed27fff95164a73f1c1fa310175ee5
Reviewed-on: https://chromium-review.googlesource.com/563021
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18939}
2017-07-08 23:36:26 +00:00
tommi
cf39dd5d82 Add RTC_FROM_HERE location information to two DCHECKs in ProcessThread.
BUG=none
TBR=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2967693002
Cr-Commit-Position: refs/heads/master@{#18937}
2017-07-07 23:24:34 +00:00
peah
fb660ae633 Decreased the adaptation rate for the adaptive filter in the echo canceller 3
BUG=webrtc:7955

Review-Url: https://codereview.webrtc.org/2968223003
Cr-Commit-Position: refs/heads/master@{#18932}
2017-07-07 14:59:24 +00:00
ehmaldonado
eaaae9e91b base->rtc_base: Update .c, .mm and .java files.
TBR=kwiberg@webrtc.org
BUG=webrtc:7634

Review-Url: https://codereview.webrtc.org/2974613003
Cr-Commit-Position: refs/heads/master@{#18926}
2017-07-07 10:09:51 +00:00
braveyao
4a494ffd12 desktop_capture: crop border in window_capture on Win8/10
On Windows8/10, we prefer cropping desired window out from a whole screen
capture due to some reasons. The problem is Win10 has an invisible border
around the window. If we leave the border, it will expose background a bit.

This cl is about to always remove the border of desired window on Win8/10.
This will help a lot to capturing still windows during window sharing.
This cl still can't handle the background exposure issue when you move the
target window around during capturing. More investigation is needed.

BUG=chromium:737278

Review-Url: https://codereview.webrtc.org/2973853002
Cr-Commit-Position: refs/heads/master@{#18921}
2017-07-07 03:20:27 +00:00
Edward Lemur
c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00
Sebastian Jansson
9e3f1e4ca2 Fixed a miscalculation of sent bitrate caused by mixup of time units
Bug: webrtc:7949
Change-Id: Ia57fdd3d1de0952b80e77c30b0a6cfe44515eff2
Reviewed-on: https://chromium-review.googlesource.com/561460
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18917}
2017-07-06 15:22:58 +00:00
henrik.lundin
a44910787b Let NetEq reset the AudioFrame during muted state
In practice, this change will make AudioFrame::muted_ replicate the
explicit muted variable, passed as a pointer to NetEq::GetAudio.

BUG=webrtc:7944

Review-Url: https://codereview.webrtc.org/2965203002
Cr-Commit-Position: refs/heads/master@{#18914}
2017-07-06 12:23:53 +00:00
sprang
168794c43c Implement RTP keepalive in native stack.
BUG=webrtc:7907

Review-Url: https://codereview.webrtc.org/2960363002
Cr-Commit-Position: refs/heads/master@{#18912}
2017-07-06 11:38:06 +00:00
mbonadei
5c0d703382 Moving asm code out of isac_fix_c sources list
BUG=webrtc:7743

Review-Url: https://codereview.webrtc.org/2973613002
Cr-Commit-Position: refs/heads/master@{#18911}
2017-07-06 10:48:55 +00:00