f8c2baca4e
Add a gyp/gn variable for whether to use iLBC or not
...
BUG=webrtc:5415
Review URL: https://codereview.webrtc.org/1578953003
Cr-Commit-Position: refs/heads/master@{#11291}
2016-01-18 14:38:40 +00:00
34ed2b95a5
[rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function
...
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1544983002
Cr-Commit-Position: refs/heads/master@{#11288}
2016-01-18 10:43:38 +00:00
cec0a08275
Add a new interface for creating a udp socket in which it binds the socket to a network if the network handle is set.
...
Plus, in stunport, turnport and allocation sequence, create a socket using the new interface.
BUG=
Review URL: https://codereview.webrtc.org/1556743002
Cr-Commit-Position: refs/heads/master@{#11279}
2016-01-15 22:49:15 +00:00
56271ed889
fix bug 5430
...
Fixed misusage of Connection function and also fixed the test case.
BUG=webrtc:5430
Review URL: https://codereview.webrtc.org/1592763003
Cr-Commit-Position: refs/heads/master@{#11278}
2016-01-15 22:45:11 +00:00
884f58523a
Storing raw audio sink for default audio track.
...
BUG=webrtc:5250
Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
Cr-Commit-Position: refs/heads/master@{#11230}
Review URL: https://codereview.webrtc.org/1551813002
Cr-Commit-Position: refs/heads/master@{#11275}
2016-01-15 17:20:08 +00:00
1567d0bd98
[rtp_rtcp] rtcp::Sdes moved into own file
...
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1439553003/
BUG=webrtc:5260
R=asapersson@webrtc.org , åsapersson
Review URL: https://codereview.webrtc.org/1592763002 .
Cr-Commit-Position: refs/heads/master@{#11274}
2016-01-15 16:34:32 +00:00
79a5a83e10
Adapt to boringssl's new defaults.
...
This is now a merge with patchset #2 of https://codereview.webrtc.org/1550773002 after that CL was reverted.
BUG=webrtc:5381
Review URL: https://codereview.webrtc.org/1589493004
Cr-Commit-Position: refs/heads/master@{#11273}
2016-01-15 15:16:54 +00:00
2c13297bf5
[rtp_rtcp] rtcp::Rpsi moved into own file
...
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1550293003/
BUG=webrtc:5260
R=asapersson@webrtc.org , åsapersson
Review URL: https://codereview.webrtc.org/1583233007 .
Cr-Commit-Position: refs/heads/master@{#11272}
2016-01-15 14:21:34 +00:00
256e5b23f8
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1557593002/
...
BUG=webrtc:5260
R=asapersson@webrtc.org , åsapersson
Review URL: https://codereview.webrtc.org/1579213005 .
Cr-Commit-Position: refs/heads/master@{#11271}
2016-01-15 13:16:36 +00:00
5679da1291
[rtp_rtcp] rtcp::Fir moved into own file
...
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1544403002
BUG=webrtc:5260
R=asapersson@webrtc.org , åsapersson
Review URL: https://codereview.webrtc.org/1581983003 .
Cr-Commit-Position: refs/heads/master@{#11269}
2016-01-15 12:19:59 +00:00
a5eba6c98b
[rtp_rtcp] rtcp::Remb moved into own file
...
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1552773002/
BUG=webrtc:5260
R=asapersson@webrtc.org , åsapersson
Review URL: https://codereview.webrtc.org/1590883002 .
Cr-Commit-Position: refs/heads/master@{#11268}
2016-01-15 11:40:27 +00:00
d66b44d565
Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
...
The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
Original review: https://codereview.webrtc.org/1413483003/
The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.
TBR=glaznev@webrtc.org , henrik.lundin@webrtc.org , solenberg@google.com , henrikg@webrtc.org , perkj@webrtc.org
BUG=webrtc:4741
Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
Cr-Commit-Position: refs/heads/master@{#11093}
Review URL: https://codereview.webrtc.org/1540103002
Cr-Commit-Position: refs/heads/master@{#11267}
2016-01-15 11:06:41 +00:00
d9e62f5837
Fixed sending Rtp packets with non zero csrcs and certain extensions.
...
Added test that fails because of given issue.
BUG=webrtc:5413
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1586523003
Cr-Commit-Position: refs/heads/master@{#11258}
2016-01-14 22:55:23 +00:00
67b1e1ab0b
Put options as the argument of the java PeerConnectionFactory constructor.
...
BUG=
Review URL: https://codereview.webrtc.org/1581903002
Cr-Commit-Position: refs/heads/master@{#11257}
2016-01-14 22:45:44 +00:00
5d332ac8ff
Fix expectation bug in the RTPSender unit test.
...
The current expectation for InsertPacket(...) uses WillRepeatedly, which accepts if the function is called zero or more times. This CL changes this to either a fixed number of calls, or at least a positive number of calls.
Review URL: https://codereview.webrtc.org/1585783003
Cr-Commit-Position: refs/heads/master@{#11256}
2016-01-14 22:37:43 +00:00
04cb763955
Add tests for verifying transport feedback for audio and video.
...
BUG=webrtc:5263
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1589523002 .
Cr-Commit-Position: refs/heads/master@{#11255}
2016-01-14 19:34:39 +00:00
fcfc804e43
Eliminate defines in talk/
...
Replace LINUX, OSX and IOS defines with WEBRTC_ prefixed versions.
Remove no longer used defines from talk/build/common.gypi due to
previously migrated sources (into webrtc/p2p and webrtc/libjingle).
When this is rolled into Chromium, we can also clean up the platform
defines in
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp
NOTRY=True
BUG=webrtc:5420
TESTED=Ran all compile trybots with --clobber flag.
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1588453005
Cr-Commit-Position: refs/heads/master@{#11254}
2016-01-14 19:01:25 +00:00
3542013f58
Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
...
Reason for revert:
We're getting boringssl version conflicts. Reverting for now.
Original issue's description:
> Update with new default boringssl no-aes cipher suites. Re-enable tests.
>
> This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part).
>
> BUG=webrtc:5381
> R=davidben@webrtc.org , henrika@webrtc.org
>
> Committed: https://crrev.com/31c8d2eac5aec977f584ab0ae5a1d457d674f101
> Cr-Commit-Position: refs/heads/master@{#11250}
TBR=davidben@webrtc.org ,henrika@webrtc.org ,torbjorng@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5381
Review URL: https://codereview.webrtc.org/1586183002
Cr-Commit-Position: refs/heads/master@{#11253}
2016-01-14 17:14:06 +00:00
2734d77c95
Remove assert which was incorrectly added to TcpPort::OnSentPacket.
...
TBR=pthatcher@webrtc.org
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1588083002 .
Cr-Commit-Position: refs/heads/master@{#11252}
2016-01-14 16:04:04 +00:00
55674ffb32
Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
...
Chromium reported errors when building libjingle_nacl due to some methods used virtual instead of override when they were overriding the base class. My guess is that when one method starts using override, all other in the same class must too.
R=tommi@webrtc.org
TBR=pthatcher@webtrc.org
BUG=4173
Review URL: https://codereview.webrtc.org/1589563003 .
Cr-Commit-Position: refs/heads/master@{#11251}
2016-01-14 14:49:23 +00:00
31c8d2eac5
Update with new default boringssl no-aes cipher suites. Re-enable tests.
...
This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part).
BUG=webrtc:5381
R=davidben@webrtc.org , henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1550773002 .
Cr-Commit-Position: refs/heads/master@{#11250}
2016-01-14 14:18:02 +00:00
e5e0e57bdf
Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
...
Reason for revert:
Broke Chrome:
https://build.chromium.org/p/tryserver.chromium.linux/builders/linux_chromium_chromeos_compile_dbg_ng/builds/143025/steps/compile%20%28with%20patch%29/logs/stdio
FAILED: cd ../../third_party/libjingle; python ../../native_client/build/build_nexe.py --root ../.. --product-dir ../../out/Debug/xyz --config-name Debug -t ../../native_client/toolchain/ --arch pnacl --build newlib_plib --name ../../out/Debug/gen/tc_pnacl_newlib/lib/libjingle_nacl.a --objdir ../../out/Debug/obj/third_party/libjingle/libjingle_nacl.gen/pnacl_newlib-pnacl/libjingle_nacl "--include-dirs=../../out/Debug/gen/tc_pnacl_newlib/include ../.. \"../../out/Debug/gen\" ./source ../ ../../native_client_sdk/src/libraries ../../native_client_sdk/src/libraries/nacl_io/include ../../native_client_sdk/src/libraries/third_party/newlib-extras ../expat/files/lib ../boringssl/src/include" "--compile_flags=-O2 -g -Wall -fdiagnostics-show-option -Werror -Wno-unused-function -Wno-char-subscripts -Wno-c++11-extensions -Wno-unnamed-type-template-args -Wno-extra-semi -Wno-unused-private-field -Wno-char-subscripts -Wno-unused-function \"-std=gnu++11\" " --gomadir /b/build/goma "--defines=\"__STDC_LIMIT_MACROS=1\" \"__STDC_FORMAT_MACROS=1\" \"_GNU_SOURCE=1\" \"_POSIX_C_SOURCE=199506\" \"_XOPEN_SOURCE=600\" \"DYNAMIC_ANNOTATIONS_ENABLED=1\" \"DYNAMIC_ANNOTATIONS_PREFIX=NACL_\" \"NACL_BUILD_ARCH=x86\" V8_DEPRECATION_WARNINGS \"CLD_VERSION=2\" \"_FILE_OFFSET_BITS=64\" CHROMIUM_BUILD \"CR_CLANG_REVISION=255169-1\" COMPONENT_BUILD UI_COMPOSITOR_IMAGE_TRANSPORT \"USE_AURA=1\" \"USE_ASH=1\" \"USE_PANGO=1\" \"USE_CAIRO=1\" \"USE_DEFAULT_RENDER_THEME=1\" \"USE_LIBJPEG_TURBO=1\" \"USE_X11=1\" \"IMAGE_LOADER_EXTENSION=1\" \"ENABLE_WEBRTC=1\" \"ENABLE_MEDIA_ROUTER=1\" USE_PROPRIETARY_CODECS ENABLE_PEPPER_CDMS ENABLE_CONFIGURATION_POLICY ENABLE_NOTIFICATIONS \"ENABLE_HIDPI=1\" \"ENABLE_TOPCHROME_MD=1\" USE_UDEV DONT_EMBED_BUILD_METADATA \"DCHECK_ALWAYS_ON=1\" FIELDTRIAL_TESTING_ENABLED \"ENABLE_TASK_MANAGER=1\" \"ENABLE_EXTENSIONS=1\" \"ENABLE_PDF=1\" \"ENABLE_PLUGINS=1\" \"ENABLE_SESSION_SERVICE=1\" \"ENABLE_THEMES=1\" \"ENABLE_AUTOFILL_DIALOG=1\" \"ENABLE_BACKGROUND=1\" \"ENABLE_PRINTING=1\" \"ENABLE_PRINT_PREVIEW=1\" \"ENABLE_SPELLCHECK=1\" \"ENABLE_CAPTIVE_PORTAL_DETECTION=1\" \"ENABLE_APP_LIST=1\" \"ENABLE_SUPERVISED_USERS=1\" \"ENABLE_MDNS=1\" \"ENABLE_SERVICE_DISCOVERY=1\" V8_USE_EXTERNAL_STARTUP_DATA FULL_SAFE_BROWSING SAFE_BROWSING_CSD SAFE_BROWSING_DB_LOCAL EXPAT_RELATIVE_PATH FEATURE_ENABLE_SSL GTEST_RELATIVE_PATH HAVE_OPENSSL_SSL_H NO_MAIN_THREAD_WRAPPING NO_SOUND_SYSTEM WEBRTC_POSIX SRTP_RELATIVE_PATH SSL_USE_OPENSSL USE_WEBRTC_DEV_BRANCH \"timezone=_timezone\" XML_STATIC \"USE_LIBPCI=1\" \"USE_OPENSSL=1\" \"USE_OPENSSL_CERTS=1\"" "--link_flags=-B../../out/Debug/gen/tc_pnacl_newlib/lib " "--source-list=../../out/gypfiles/third_party/libjingle/pnacl_newlib.libjingle_nacl.source_list.gypcmd"
In file included from ../webrtc/p2p/base/tcpport.cc:67:
../webrtc/p2p/base/tcpport.h:50:23: error: 'CreateConnection' overrides a member function but is not marked 'override' [-Werror,-Winconsistent-missing-override]
virtual Connection* CreateConnection(const Candidate& address,
^
../webrtc/p2p/base/portinterface.h:71:23: note: overridden virtual function is here
virtual Connection* CreateConnection(
^
In file included from ../webrtc/p2p/base/tcpport.cc:67:
../webrtc/p2p/base/tcpport.h:53:16: error: 'PrepareAddress' overrides a member function but is not marked 'override' [-Werror,-Winconsistent-missing-override]
virtual void PrepareAddress();
^
../webrtc/p2p/base/portinterface.h:63:16: note: overridden virtual function is here
virtual void PrepareAddress() = 0;
^
(etc)
Original issue's description:
> Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
>
> To reduce the risk of future mistakes when connecting Ports, Port::OnSentPacket was made pure virtual to ensure that new implementations take care of it.
>
> BUG=4173
> R=pthatcher@webrtc.org
>
> Committed: https://crrev.com/7307952a5bf63311e5f9a2a90089a06177e42716
> Cr-Commit-Position: refs/heads/master@{#11247}
TBR=pthatcher@webrtc.org ,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=4173
Review URL: https://codereview.webrtc.org/1586063002
Cr-Commit-Position: refs/heads/master@{#11249}
2016-01-14 12:57:03 +00:00
688e308a35
Re-land: "Use an explicit identifier in Config"
...
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.
Original CL: https://codereview.webrtc.org/1538643004/
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1589573004
Cr-Commit-Position: refs/heads/master@{#11248}
2016-01-14 12:32:51 +00:00
7307952a5b
Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
...
To reduce the risk of future mistakes when connecting Ports, Port::OnSentPacket was made pure virtual to ensure that new implementations take care of it.
BUG=4173
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1577873003 .
Cr-Commit-Position: refs/heads/master@{#11247}
2016-01-14 12:15:56 +00:00
ff2a6351e0
Add ramp-up tests for transport sequence number with and w/o audio.
...
Also add a perf metric tracking the average network latency.
The audio stream test is disabled for now since audio isn't included in bitrate allocation.
BUG=webrtc:5263
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1582833002 .
Cr-Commit-Position: refs/heads/master@{#11244}
2016-01-14 09:00:34 +00:00
beed8280d8
Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
...
Previosly ToSesnsetiveString() wasn't working witn some implementations
of inet_ntop(). Rewrote it to avoid that dependency.
BUG=chromium:577344
R=pthatcher@webrtc.org , tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1584793004 .
Cr-Commit-Position: refs/heads/master@{#11242}
2016-01-14 02:14:59 +00:00
2d110be77f
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
...
Reason for revert:
tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach.
Original issue's description:
> Storing raw audio sink for default audio track.
>
> BUG=webrtc:5250
>
> Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
> Cr-Commit-Position: refs/heads/master@{#11230}
TBR=pthatcher@webrtc.org ,solenberg@webrtc.org ,pbos@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:5250
Review URL: https://codereview.webrtc.org/1588693002
Cr-Commit-Position: refs/heads/master@{#11241}
2016-01-13 20:00:29 +00:00
8432e1f4b8
Re-enable tests that failed under Linux_Msan.
...
Fixed in latest libvpx roll.
Keep EndToEndTest.TransportSeqNumOnAudioAndVideo disabled on
Win_DrMemory for now as it seems to time-out/too slow.
TBR=stefan@webrtc.org , kjellander@webrtc.org
BUG=webrtc:5402
NOTRY=True
Review URL: https://codereview.webrtc.org/1577313003
Cr-Commit-Position: refs/heads/master@{#11240}
2016-01-13 16:35:51 +00:00
fca54f41ad
Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
...
Reason for revert:
Reverting due to problem with roll:
/b/build/slave/linux/build/src/buildtools/linux64/gn gen //out/Release '--args=ffmpeg_branding="Chrome" proprietary_codecs=true is_debug=false is_component_build=false use_goma=true goma_dir="/b/build/goma" symbol_level=1 dcheck_always_on=true' --check --runtime-deps-list-file=/b/build/slave/linux/build/src/out/Release/runtime_deps
-> returned 1
ERROR at //third_party/webrtc/BUILD.gn:245:18: Item not found
configs -= [ "//build/config/clang:find_bad_constructs" ]
^-----------------------------------------
You were trying to remove "//build/config/clang:find_bad_constructs"
from the list but it wasn't there.
GN gen failed: 1
step returned non-zero exit code: 1
@@@STEP_FAILURE@@@
Original issue's description:
> Use an explicit identifier in Config
>
> This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.
>
> Committed: https://crrev.com/25249d92d3cf105bcc7b684c8924ccdbc9afcb93
> Cr-Commit-Position: refs/heads/master@{#11231}
TBR=henrik.lundin@webrtc.org ,stefan@webrtc.org ,tommi@chromium.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/1586563003
Cr-Commit-Position: refs/heads/master@{#11239}
2016-01-13 16:12:07 +00:00
292e192f17
Add build_protobuf variable.
...
This makes it possible to use protobuffers with
an external protobuf library instead of the one that
comes with the WebRTC code.
NOTRY=True
Review URL: https://codereview.webrtc.org/1589433002
Cr-Commit-Position: refs/heads/master@{#11236}
2016-01-13 13:47:07 +00:00
a276e73168
Clean the code for external denoiser.
...
BUG=webrtc:5255
Review URL: https://codereview.webrtc.org/1578373003
Cr-Commit-Position: refs/heads/master@{#11235}
2016-01-13 13:36:40 +00:00
2f7dea164d
[rtp_rtcp] rtcp::Empty moved into own file and renamed to CompoundPacket on the way
...
Class renamed to indicated use of the rtcp::Empty class: it can only be used as container for other rtcp packets.
All tests that use Append function moved from rtcp_packet_unittest, even if they did not use Empty class.
This is because there is plan to make RtcpPacket class lighter by moving Append functionality to CompoundPacket class.
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1582503002
Cr-Commit-Position: refs/heads/master@{#11234}
2016-01-13 10:03:09 +00:00
ea8c0f6fcb
Fix capture ntp time issue introduced with r11187.
...
I think the problem was that I only introduced delay in one direction, and the estimation assumes that the RTT is evenly divided between the send direction and the receive direction, which was true for the old test.
BUG=chromium:576246
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1577853005 .
Cr-Commit-Position: refs/heads/master@{#11233}
2016-01-13 07:58:52 +00:00
25249d92d3
Use an explicit identifier in Config
...
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.
Review URL: https://codereview.webrtc.org/1538643004
Cr-Commit-Position: refs/heads/master@{#11231}
2016-01-13 02:50:31 +00:00
e591f9377f
Storing raw audio sink for default audio track.
...
BUG=webrtc:5250
Review URL: https://codereview.webrtc.org/1551813002
Cr-Commit-Position: refs/heads/master@{#11230}
2016-01-13 00:45:33 +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
92e677a1f8
[rtp_rtcp] rtcp::Sli packet moved into own file and got Parse function
...
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1551893002
Cr-Commit-Position: refs/heads/master@{#11228}
2016-01-12 18:05:00 +00:00
5584bf4c4d
Make :rtc_base_approved a public dep of :rtc_base.
...
It looks to me like targets :rtc_base_approved is logically a subset of
:rtc_base, and so any targets depending on :rtc_base expect to also get
access to the headers in :rtc_base_approved.
Thus I think it's appropriate for :rtc_base to have :rtc_base_approved in
public_deps, so that `gn check` will permit this without clients having to
explicitly depend on both.
NOTRY=True
Review URL: https://codereview.webrtc.org/1578833002
Cr-Commit-Position: refs/heads/master@{#11227}
2016-01-12 17:46:59 +00:00
e84e96e8be
NetEq: Fix a typo in a comment
...
TBR=minyue@webrtc.org
NOTRY=true
Review URL: https://codereview.webrtc.org/1578223003 .
Cr-Commit-Position: refs/heads/master@{#11226}
2016-01-12 15:36:23 +00:00
36220ae24f
Slap deprecation notices on Pass methods
...
There's no reason not to use std::move instead now that we can use the
C++11 standard library.
BUG=webrtc:5373
Review URL: https://codereview.webrtc.org/1531013003
Cr-Commit-Position: refs/heads/master@{#11225}
2016-01-12 15:24:27 +00:00
d20e651327
Fix test bug introduced in r11101.
...
BUG=chromium:572995
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1578223002 .
Cr-Commit-Position: refs/heads/master@{#11224}
2016-01-12 14:51:28 +00:00
2845a02339
Remove unused enum RTPDirections.
...
BUG=
Review URL: https://codereview.webrtc.org/1582523002
Cr-Commit-Position: refs/heads/master@{#11221}
2016-01-12 13:01:02 +00:00
3842c5c7f7
Wire-up BWE feedback for audio receive streams.
...
Also wires up receiving transport sequence numbers.
BUG=webrtc:5263
R=mflodman@webrtc.org , pbos@webrtc.org , solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1535963002 .
Cr-Commit-Position: refs/heads/master@{#11220}
2016-01-12 12:55:11 +00:00
67e94fb6f2
Add unit test for stand-alone denoiser and fixed some bugs.
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The unit test will run the pure C denoiser and SSE2/NEON denoiser (based
on the CPU detection) and compare the denoised frames to ensure the bit
exact.
TBR=tommi@webrtc.org
BUG=webrtc:5255
Review URL: https://codereview.webrtc.org/1492053003
Cr-Commit-Position: refs/heads/master@{#11216}
2016-01-12 05:34:14 +00:00
b2328d11dc
Remove additional channel constraints when Beamforming is enabled in AudioProcessing
...
The general constraints on number of channels for AudioProcessing is:
num_in_channels == num_out_channels || num_out_channels == 1
When Beamforming is enabled and additional constraint was added forcing:
num_out_channels == 1
This artificial constraint was removed by adding upmixing support in CopyTo, since it was already supported for the AudioFrame interface using InterleaveTo.
Review URL: https://codereview.webrtc.org/1571013002
Cr-Commit-Position: refs/heads/master@{#11215}
2016-01-12 04:32:32 +00:00
2a34688f86
Make Beamforming dynamically settable for Android platform builds
...
Review URL: https://codereview.webrtc.org/1563493005
Cr-Commit-Position: refs/heads/master@{#11213}
2016-01-12 02:04:33 +00:00
2bc63a1dd3
clang-format audio_device/mac.
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NOTRY=true
Review URL: https://codereview.webrtc.org/1570063003
Cr-Commit-Position: refs/heads/master@{#11212}
2016-01-11 23:59:25 +00:00
a7446d2a50
Change DTLS default from 1.0 to 1.2 for webrtc.
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This changes for standalone webrtc applications.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1548733002 .
Cr-Commit-Position: refs/heads/master@{#11211}
2016-01-11 23:27:12 +00:00
f6c318ebae
Update API for Objective-C RTCMediaSource.
...
BUG=
R=tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1538263002 .
Patch from Jon Hjelle <hjon@andyet.net >.
Cr-Commit-Position: refs/heads/master@{#11210}
2016-01-11 22:39:05 +00:00
e799badacc
Move Objective-C video renderers to webrtc/api/objc.
...
BUG=
R=tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1542473003 .
Patch from Jon Hjelle <hjon@andyet.net >.
Cr-Commit-Position: refs/heads/master@{#11209}
2016-01-11 21:47:17 +00:00
