5000 Commits

Author SHA1 Message Date
guoweis
318166bed7 Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ )
Reason for revert:
Broke chromium fyi build.

Original issue's description:
> Convert internal representation of Srtp cryptos from string to int.
>
> Note that the coversion from int to string happens in 3 places
> 1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
> 2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
> 3) stats collection also needs external names.
>
> External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
> Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.
>
> The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().
>
> BUG=webrtc:5043
>
> Committed: https://crrev.com/2764e1027a08a5543e04b854a27a520801faf6eb
> Cr-Commit-Position: refs/heads/master@{#10701}

TBR=juberti@webrtc.org,pthatcher@webrtc.org,juberti@google.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5043

Review URL: https://codereview.webrtc.org/1455233005

Cr-Commit-Position: refs/heads/master@{#10702}
2015-11-19 03:03:46 +00:00
guoweis
2764e1027a Convert internal representation of Srtp cryptos from string to int.
Note that the coversion from int to string happens in 3 places
1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
3) stats collection also needs external names.

External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.

The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().

BUG=webrtc:5043

Review URL: https://codereview.webrtc.org/1416673006

Cr-Commit-Position: refs/heads/master@{#10701}
2015-11-19 02:02:40 +00:00
kjellander@webrtc.org
3c652b6746 modules/audio_coding: Remove some codec include dirs
Also clean up some include_dir entries and update the few
references to them with absolute include paths instead.
Finally fixed a few lint errors and invalid header guards.

None of these are used downstream.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1438663003 .

Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 22:08:46 +00:00
kjellander@webrtc.org
b7ce96470b modules/video_coding/utility: Remove include
This makes it clearer this code not meant to be used as an API.
I could not find any use of this in downstream code.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=stefan@webrtc.org
TBR=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1440873005 .

Cr-Commit-Position: refs/heads/master@{#10699}
2015-11-18 22:04:20 +00:00
Henrik Kjellander
0f59a88b32 modules/video_processing: refactor interface->include + more.
Moved/renamed:
webrtc/modules/video_processing/main/interface -> webrtc/modules/video_processing/include
webrtc/modules/video_processing/main/source/* -> webrtc/modules/video_processing
webrtc/modules/video_processing/main/test/unit_test -> webrtc/modules/video_processing/test

No downstream code seems to use this module.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1410663004 .

Cr-Commit-Position: refs/heads/master@{#10697}
2015-11-18 21:31:33 +00:00
Henrik Kjellander
ed7d6ec63e WebRTC: Add compability header for video_coding refactoring.
It turns out there were downstream use of the encoded_frame.h header
that was moved in https://codereview.webrtc.org/1417283007/.
Add a copy of it in the old location to allow a seamless transition.

BUG=webrtc:5095
TBR=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1447163006 .

Cr-Commit-Position: refs/heads/master@{#10696}
2015-11-18 21:26:38 +00:00
Henrik Kjellander
2557b86e76 modules/video_coding refactorings
The main purpose was the interface-> include rename, but other files
were also moved, eliminating the "main" dir.

To avoid breaking downstream, the "interface" directories were copied
into a new "video_coding/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).

Other files also moved:
video_coding/main/source -> video_coding
video_coding/main/test -> video_coding/test

BUG=webrtc: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 --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc

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

Review URL: https://codereview.webrtc.org/1417283007 .

Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 21:00:33 +00:00
kwiberg
223692aa85 Remove dead code
Review URL: https://codereview.webrtc.org/1452153003

Cr-Commit-Position: refs/heads/master@{#10692}
2015-11-18 16:27:56 +00:00
kwiberg
e1a27d48ad Move CNG/RED payload type extraction to Rent-A-Codec
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1450883002

Cr-Commit-Position: refs/heads/master@{#10691}
2015-11-18 15:32:57 +00:00
peah
2446e5a2de Fixed the render queue item size preallocation and verification, moved constant declarations, removed redundant queue allocation
BUG=

Review URL: https://codereview.webrtc.org/1454683002

Cr-Commit-Position: refs/heads/master@{#10689}
2015-11-18 14:11:18 +00:00
danilchap
0219c9b4bf rtcp::App moved into own file and got Parse function
Review URL: https://codereview.webrtc.org/1437353003

Cr-Commit-Position: refs/heads/master@{#10688}
2015-11-18 13:56:57 +00:00
andrew
f70568c04b So long and thanks for all the code reviews!
- Remove myself from OWNERS.
- Add myself to AUTHORS (I signed a CLA).
- Add minyue to audio_conference_mixer which would otherwise be empty.
- Add missing comma in WATCHLISTS.

Review URL: https://codereview.webrtc.org/1458763002

Cr-Commit-Position: refs/heads/master@{#10686}
2015-11-18 11:07:45 +00:00
asapersson
cb50c96be2 Set temporal up switch bit to false for flexible mode (one temporal layer is configured currently).
BUG=webrtc:5214

Review URL: https://codereview.webrtc.org/1453693002

Cr-Commit-Position: refs/heads/master@{#10685}
2015-11-18 09:58:59 +00:00
Guo-wei Shieh
310b093aa4 Fix active tcp port to 9
In tcp only call:
Tested with hangout.
Tested with firefox.

To test firefox, goto about:config, search for media.peerconnection.ice.tcp and turn it on.

Existing test case should be suffice to cover this.

R=juberti@google.com
TBR=jubert@webrtc.org
BUG=webrtc:3849

Review URL: https://codereview.webrtc.org/1217463004 .

Cr-Commit-Position: refs/heads/master@{#10683}
2015-11-18 03:15:57 +00:00
thaloun
2935e01419 Several Tick counter improvements try #2."
This reverts commit c91d1738709b038fee84d569180cba2bbcbfe5d7.

BUG=

Review URL: https://codereview.webrtc.org/1452843003

Cr-Commit-Position: refs/heads/master@{#10682}
2015-11-17 23:02:59 +00:00
davidben
c073615d56 Update references to TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305, etc.
In preparation for implementing the standardized variant of CHACHA20_POLY1305
(it changed slightly in the standardization process),
TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305 and TLS1_CK_ECDHE_ECDSA_CHACHA20_POLY1305
were renamed to have an _OLD suffix with compatibility unsuffixed #defines
temporarily available.

Update references to include the _OLD suffixed ones. Once we've cycled through
the few consumers of the unsuffixed names (just WebRTC and QUIC), the unsuffixed
names can refer to the to-be-implemented standardized variant and eventually
the draft version will be removed.

(This has no effect on upstream OpenSSL compatibility as OpenSSL never defined
these symbols to begin with. Though probably they will once standardization is
done.)

BUG=none

Review URL: https://codereview.webrtc.org/1412803010

Cr-Commit-Position: refs/heads/master@{#10681}
2015-11-17 20:58:17 +00:00
honghaiz
32f39968ce Re-apply change https://codereview.webrtc.org/1426673007/
Do not delete the turn port entry right away when the respective
connection is deleted. The dependency on asyncinvoker has been added
in chromium libjingle-nacl.

BUG=webrtc:5120

Review URL: https://codereview.webrtc.org/1450263002

Cr-Commit-Position: refs/heads/master@{#10679}
2015-11-17 19:36:37 +00:00
henrika
5c489c9d3e Add OpenSL ES enable setting to AppRTCDemo (part 2).
It is now possible to enable OpenSL ES on devices that supports it.

Fix for https://codereview.webrtc.org/1449083002/

Review URL: https://codereview.webrtc.org/1455563002

Cr-Commit-Position: refs/heads/master@{#10678}
2015-11-17 18:12:46 +00:00
Peter Boström
2be7c54b13 Remove ViEEncoder::ScaleInputImage.
BUG=webrtc:1695
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1447073002 .

Cr-Commit-Position: refs/heads/master@{#10677}
2015-11-17 14:52:59 +00:00
Peter Boström
bd05f0ba52 Unconditionally build VP9 support.
Broken for PeerConnection either way (since VP9 support is announced)
and would fail on a CHECK apart from generating incorrect
offers/answers. This isn't a flag that we want to support, so it's
better to remove the foot-shooting gun.

BUG=
R=asapersson@webrtc.org, kjellander@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1451663002 .

Cr-Commit-Position: refs/heads/master@{#10676}
2015-11-17 14:27:41 +00:00
stefan
18adf0a79d Add UMA for send bwe and pacer bitrate.
Review URL: https://codereview.webrtc.org/1434403004

Cr-Commit-Position: refs/heads/master@{#10675}
2015-11-17 14:25:02 +00:00
pbos
d9eec762ce Trace encoding/decoding time in a generic way.
Removes VP8::Encode trace in favor of VCMGenericEncoder ones and adds
one to InitEncode. Also adds an instant event to ::Encoded since this
can be done on a different thread.

Also adds the corresponding traces to VCMGenericDecoder.

BUG=webrtc:5167
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1412573010

Cr-Commit-Position: refs/heads/master@{#10674}
2015-11-17 14:03:52 +00:00
henrika
5a71f03f8b Deactivate the audio session after a call ends using the AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation constant
since it is recommended for VoIP apps.

BUG=b/23356406
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1418483004 .

Cr-Commit-Position: refs/heads/master@{#10673}
2015-11-17 13:54:58 +00:00
henrika
fd614c2149 Adding thread timeout for audio recorer thread in Java
BUG=NONE

Review URL: https://codereview.webrtc.org/1444313002

Cr-Commit-Position: refs/heads/master@{#10671}
2015-11-17 12:28:33 +00:00
glaznev
e66339296b Add OpenSL ES enable setting to AppRTCDemo.
Disable OpenSL ES by default.
Plus remove no longer used CPU overuse detection option.

Review URL: https://codereview.webrtc.org/1449083002

Cr-Commit-Position: refs/heads/master@{#10670}
2015-11-17 12:05:35 +00:00
pbos
3c12f4dadb Revert of Create rtc::AtomicInt POD struct. (patchset #12 id:220001 of https://codereview.webrtc.org/1420043008/ )
Reason for revert:
Caused static initializers.

BUG=chromium:556866
TBR=tommi@webrtc.org

Original issue's description:
> Create rtc::AtomicInt POD struct.
>
> Prevents accidental non-atomic reads, increments and stores since
> "volatile int" doesn't enforce atomic usage.
>
> BUG=
> R=kwiberg@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/b27f590ece487819c3d1fda400315e582fb975b6
> Cr-Commit-Position: refs/heads/master@{#10657}

TBR=kwiberg@webrtc.org,tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review URL: https://codereview.webrtc.org/1453093002

Cr-Commit-Position: refs/heads/master@{#10669}
2015-11-17 11:21:07 +00:00
peah
192164eebc Preparational work before introducing the locks in order to harmonize the code:
-Moved the initialize function
-Moved api_format into the shared state

BUG=

Review URL: https://codereview.webrtc.org/1413093002

Cr-Commit-Position: refs/heads/master@{#10668}
2015-11-17 10:16:51 +00:00
peah
4d291f7d5e Applied the render queueing to the agc.
BUG=webrtc:5099

Review URL: https://codereview.webrtc.org/1416583003

Cr-Commit-Position: refs/heads/master@{#10667}
2015-11-17 07:52:32 +00:00
pbos
740c4f11e0 Remove packet initializer in RtpRtcpRtxNackTest.
Fixes RtpRtcpRtxNackTest to not use uninitialized data when not sending
RTX.

BUG=webrtc:3183
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1427653007

Cr-Commit-Position: refs/heads/master@{#10665}
2015-11-17 01:19:39 +00:00
pbos
854e84c7fb Use webrtc/base/logging.h for video coding/processing.
Replaces system_wrappers' logging.h in video_coding and
video_processing.

BUG=webrtc:5118
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1435873003

Cr-Commit-Position: refs/heads/master@{#10664}
2015-11-17 00:39:10 +00:00
thaloun
c91d173870 Revert of Several Tick counter improvements. (patchset #8 id:140001 of https://codereview.webrtc.org/1415923010/ )
Reason for revert:
Potentially breaks a threading test under DrMemory.  Rolling back while I investigate.

Original issue's description:
> Several Tick counter improvements.
>
> Move logic into cc file
> Simplify interval calculation
> Remove unused QUERY_PERFORMANCE_COUNTER windows implementation
> Remove double divide on each ::Now() invocation on mac
>
> Move TickTime and TickInterval funcitons to cc file in prep for refactoring.
>
> BUG=
> R=mflodman@webrtc.org, pbos@webrtc.org
>
> Committed: https://crrev.com/4c27e4b62da2047063d88eedfeec3e939fea7843
> Cr-Commit-Position: refs/heads/master@{#10661}

TBR=pbos@webrtc.org,mflodman@webrtc.org,noahric@chromium.org,thaloun@google.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review URL: https://codereview.webrtc.org/1450203002

Cr-Commit-Position: refs/heads/master@{#10663}
2015-11-17 00:28:56 +00:00
peah
fa6228e221 Introduced the render sample queue for the aec and aecm.
BUG=webrtc:5099

Review URL: https://codereview.webrtc.org/1410833002

Cr-Commit-Position: refs/heads/master@{#10662}
2015-11-17 00:27:50 +00:00
Tim Haloun
4c27e4b62d Several Tick counter improvements.
Move logic into cc file
Simplify interval calculation
Remove unused QUERY_PERFORMANCE_COUNTER windows implementation
Remove double divide on each ::Now() invocation on mac

Move TickTime and TickInterval funcitons to cc file in prep for refactoring.

BUG=
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1415923010 .

Cr-Commit-Position: refs/heads/master@{#10661}
2015-11-16 22:37:59 +00:00
Alex Glaznev
eb8b388273 Fix VP9 support in AppRTCDemo.
Default VP9 selection is no longer triggered by field trial string
after https://codereview.webrtc.org/1432673002, so VP9 need to
be selected now through SDP mangling.

R=wzh@webrtc.org

Review URL: https://codereview.webrtc.org/1452783002 .

Cr-Commit-Position: refs/heads/master@{#10660}
2015-11-16 22:12:01 +00:00
kjellander
6f8ce060a2 common_video: rename interface -> include
To avoid breaking downstream, the "interface" directories were copied
into a new "common_video/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
The header guards are also identical to avoid mixing them up in the transition.

BUG=webrtc: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 --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc

Review URL: https://codereview.webrtc.org/1418913006

Cr-Commit-Position: refs/heads/master@{#10659}
2015-11-16 21:52:31 +00:00
pbos
b27f590ece Create rtc::AtomicInt POD struct.
Prevents accidental non-atomic reads, increments and stores since
"volatile int" doesn't enforce atomic usage.

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

Review URL: https://codereview.webrtc.org/1420043008

Cr-Commit-Position: refs/heads/master@{#10657}
2015-11-16 19:03:06 +00:00
brucedawson
3528a27b1b Flesh out webrtc/.gitignore
Chromium only checks out the webrtc directory so it misses the root
level .gitignore file which leads to messy "git status" reports inside
third_party/webrtc. This copies the root level .gitignore so that
.vcxproj files, the OSX equivalent, and other files will be ignored.

Some of the entries are irrelevant, but it is better too have a few
irrelevant entries than to be missing some, and the simplicity of
copying is valuable.

NOTRY=True
TBR=henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1432413002

Cr-Commit-Position: refs/heads/master@{#10656}
2015-11-16 19:02:02 +00:00
solenberg
8b85de2ba1 Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1442483003

Cr-Commit-Position: refs/heads/master@{#10654}
2015-11-16 17:48:12 +00:00
minyue
9a7c838ec4 Adding stddef.h to opus_inst.h.
This is to prevent size_t from undefined. This does not happen in current WebRTC since the sources that opus_inst.h gets used have proper definitions. But it would be good to add the definition in itself.

Review URL: https://codereview.webrtc.org/1446093003

Cr-Commit-Position: refs/heads/master@{#10653}
2015-11-16 16:07:04 +00:00
solenberg
3a94154035 Move some send stream configuration into webrtc::AudioSendStream.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1418503010

Cr-Commit-Position: refs/heads/master@{#10652}
2015-11-16 15:34:59 +00:00
kwiberg
e155ae671c Move CNG and RED management into the Rent-A-Codec
This leaves CodecOwner without a job, so we eliminate it.

BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1443653004

Cr-Commit-Position: refs/heads/master@{#10650}
2015-11-16 12:50:02 +00:00
tommi
54e92326af Revert of Do not delete the turn port entry right away when the respective connection is deleted. (patchset #5 id:260001 of https://codereview.webrtc.org/1426673007/ )
Reason for revert:
I have to revert this unfortunately because it adds a dependency on AsyncInvoker, which is not included when building libjingle_nacl in Chromium.
AsyncInvoker needs to first be added to the list of sources in Chromium.

Original issue's description:
> Do not delete the turn port entry right away when the respective connection is deleted.
> BUG=webrtc:5120
>
> Committed: https://crrev.com/e58fe8ef0e6d959f54adee3ed77764927d3845cc
> Cr-Commit-Position: refs/heads/master@{#10641}

TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5120

Review URL: https://codereview.webrtc.org/1449863002

Cr-Commit-Position: refs/heads/master@{#10649}
2015-11-16 12:13:02 +00:00
Henrik Kjellander
0b9e29c87d Remove include dirs from modules/{media_file,pacing}
Also move files out of media_file/source.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=asapersson@webrtc.org, perkj@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1435093002 .

Cr-Commit-Position: refs/heads/master@{#10647}
2015-11-16 10:12:32 +00:00
nisse
d9b75bef5d Fix a data race in the thread unit tests.
The flag used in thread_unittest.cc:FunctorB is subject to a (mostly
harmless) data race. In a tsan build, reproduce using

  out/Release/rtc_unittests --gtest_filter=AsyncInvokeTest.FireAndForget

There are additional tsan warnings, not all deterministic, when
running all the rtc_unittets: Some data races related to destructors,
and a locking-order-inversion warning. Hence applying this patch does
not make the unit tests tsan-clean.

I should also add that this is my very first cl, so I'm not at all
familiar with the process.

Review URL: https://codereview.webrtc.org/1439613004

Cr-Commit-Position: refs/heads/master@{#10645}
2015-11-16 08:54:10 +00:00
asapersson
6f14be8df8 Add limit for minimum number of required samples before recording input and sent framerate stats.
BUG=

Review URL: https://codereview.webrtc.org/1446443002

Cr-Commit-Position: refs/heads/master@{#10644}
2015-11-16 08:40:57 +00:00
honghaiz
e58fe8ef0e Do not delete the turn port entry right away when the respective connection is deleted.
BUG=webrtc:5120

Review URL: https://codereview.webrtc.org/1426673007

Cr-Commit-Position: refs/heads/master@{#10641}
2015-11-14 01:54:47 +00:00
tfarina
fa5d0dbd1e cleanup: get rid of basicdefs.h include
The ARRAY_SIZE macro it defines is not used anymore, as all the usages
were converted to arraysize macro from arraysize.h.

BUG=None
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1443273002

Cr-Commit-Position: refs/heads/master@{#10640}
2015-11-13 22:37:43 +00:00
honghaiz
a4845efe14 Fix flaky tests
BUG=webrtc:5190,webrtc:5136

Review URL: https://codereview.webrtc.org/1405103006

Cr-Commit-Position: refs/heads/master@{#10639}
2015-11-13 17:53:01 +00:00
danilchap
f8506cbdd8 rtcp::Ij renamed to rtcp::ExtendedJitterReport
to match name given in the RFC5450
  private member renamed to inter_arrival_jitters_ for the same reason.
rtcp::ExtendedJitterReport moved into own file
accessors and Parse function added
  to make class usable for parsing packet

Review URL: https://codereview.webrtc.org/1434213004

Cr-Commit-Position: refs/heads/master@{#10636}
2015-11-13 15:33:26 +00:00
pbos
0fa9b22789 Remove scoped_ptrs for VCM sender_ and receiver_.
Put VideoSender/VideoReceiver flat within the object, not as
scoped_ptrs, giving fewer allocations and looking a bit nicer.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1443613002

Cr-Commit-Position: refs/heads/master@{#10634}
2015-11-13 13:59:59 +00:00