9173 Commits

Author SHA1 Message Date
deadbeef
1e23461d5e Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ )
Reason for revert:
Broke chromium FYI bot because the chromium mock PC overrides the method whose signature is changing.

Also broke a downstream internal test, which I need to investigate further.

Original issue's description:
> Adding error output param to SetConfiguration, using new RTCError type.
>
> Most notably, will return "INVALID_MODIFICATION" if a field in the
> configuration was modified and modification of that field isn't supported.
>
> Also changing RTCError to a class that wraps an enum type, because it will
> eventually need to hold other information (like SDP line number), to match
> the RTCError that was recently added to the spec:
> https://github.com/w3c/webrtc-pc/pull/850
>
> BUG=webrtc:6916
>
> Review-Url: https://codereview.webrtc.org/2587133004
> Cr-Commit-Position: refs/heads/master@{#15777}
> Committed: 7a5fa6cd61

TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6916

Review-Url: https://codereview.webrtc.org/2600813002
Cr-Commit-Position: refs/heads/master@{#15778}
2016-12-24 09:43:32 +00:00
deadbeef
7a5fa6cd61 Adding error output param to SetConfiguration, using new RTCError type.
Most notably, will return "INVALID_MODIFICATION" if a field in the
configuration was modified and modification of that field isn't supported.

Also changing RTCError to a class that wraps an enum type, because it will
eventually need to hold other information (like SDP line number), to match
the RTCError that was recently added to the spec:
https://github.com/w3c/webrtc-pc/pull/850

BUG=webrtc:6916

Review-Url: https://codereview.webrtc.org/2587133004
Cr-Commit-Position: refs/heads/master@{#15777}
2016-12-24 08:47:59 +00:00
minyue
d01ed1fe8f Fix an error in Audio Network Adaptor: time constant passed wrong.
BUG=wbrtc:6303

Review-Url: https://codereview.webrtc.org/2595283003
Cr-Commit-Position: refs/heads/master@{#15767}
2016-12-23 09:49:37 +00:00
deadbeef
e97389c505 If network enumeration fails, try binding to the "ANY" address.
This isn't as good as being able to enumerate all networks, but it's better
than doing nothing; it still will provide STUN/TURN candidates for the default
route if one exists.

BUG=webrtc:6932

Review-Url: https://codereview.webrtc.org/2599673003
Cr-Commit-Position: refs/heads/master@{#15766}
2016-12-23 09:43:45 +00:00
deadbeef
40610e24ce Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define.
This allows building without SCTP support (and even building/running
tests). The "HAVE_SCTP" define has been functional for a while, but there
wasn't any easy way to turn it on/off.

NOTRY=True
BUG=webrtc:6933

Review-Url: https://codereview.webrtc.org/2593313002
Cr-Commit-Position: refs/heads/master@{#15763}
2016-12-22 18:53:38 +00:00
asapersson
fe50b4d750 Make class of static functions in rtp_to_ntp.h:
- UpdateRtcpList
- RtpToNtp

class RtpToNtpEstimator
- UpdateMeasurements
- Estimate

List with rtcp measurements is now private.

BUG=none

Review-Url: https://codereview.webrtc.org/2574133003
Cr-Commit-Position: refs/heads/master@{#15762}
2016-12-22 15:53:51 +00:00
danilchap
bf5f5297c5 Disable flaky VideoSendStreamTest.RemoveOverheadeFromBandwidth
BUG=webrtc:6886
NOTRY=True
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2596223002
Cr-Commit-Position: refs/heads/master@{#15761}
2016-12-22 15:51:54 +00:00
mbonadei
ebafdc8484 Refactor webrtc/modules/rtp_rtcp for GN check
This moves some GN check configurations out of .gn to individual
targets.

This commit also removes the source file 'mocks/mock_rtp_rtcp.h' from
the static_library 'rtp_rtcp' because it depends on a 'testonly = true'
target. After a check this seems only included in the unitest code:

$ grep -Rn "mocks/mock_rtp_rtcp.h" webrtc/modules/rtp_rtcp/
webrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc:18:#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc:17:#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"

This commit also removes the dependency on
'//webrt/modules/video_coding' because it seems that the following
include can be removed:

#include "webrtc/modules/video_coding/include/video_coding_defines.h"

The now checked target is:
"//webrtc/modules/rtp_rtcp/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2598963002
Cr-Commit-Position: refs/heads/master@{#15760}
2016-12-22 15:35:39 +00:00
mbonadei
000d16396e Refactor webrtc/modules/audio_conference_mixer for GN check
This moves some GN check configurations out of .gn to individual
targets.

The now checked target is:
"//webrtc/modules/audio_conference_mixer/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2593003002
Cr-Commit-Position: refs/heads/master@{#15759}
2016-12-22 14:56:21 +00:00
Henrik Kjellander
0de11aa130 Landmine to clobber failing Android x86/x64 builds
Clobber to fix Android x86/x64 builds after
https://codereview.webrtc.org/1414343008/

They started failing with errors like
../../third_party/android_tools/ndk/platforms/android-21/arch-x86_64/usr/include/stdint.h:32:20: fatal error: stddef.h: No such file or directory
   #include <stddef.h>
             ^
from https://build.chromium.org/p/tryserver.webrtc/builders/android_compile_x64_dbg/builds/10032/steps/compile/logs/stdio
A clobbered build solved the problem.

BUG=webrtc:5006
TBR=mbonadei@webrtc.org

Review-Url: https://codereview.webrtc.org/2601473002 .
Cr-Commit-Position: refs/heads/master@{#15757}
2016-12-22 11:40:56 +00:00
henrika
526248779a Disables AudioDeviceTest.StartStopPlayout on iOS
BUG=webrtc:6889
NOTRY=True

Review-Url: https://codereview.webrtc.org/2595303002
Cr-Commit-Position: refs/heads/master@{#15753}
2016-12-22 09:36:49 +00:00
asapersson
8d5608880f Do not call OnDecoderTiming before timing values are set.
Wait until first frame is decoded to avoid include zeros in stats.

BUG=b/32659204

Review-Url: https://codereview.webrtc.org/2582313002
Cr-Commit-Position: refs/heads/master@{#15752}
2016-12-22 09:26:18 +00:00
kjellander
c37ad499da Revert of Make P2PTransportChannel inherit from IceTransportInternal. (patchset #3 id:80001 of https://codereview.webrtc.org/2590063002/ )
Reason for revert:
Breaks Chromium WebRTC FYI bots:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/12337
The error was masked by another breaking change that was committer earlier. This is the first build showing the error.

Original issue's description:
> Make P2PTransportChannel inherit from IceTransportInternal.
>
> Make P2PTransportChannel inherit from IceTransportInternal instead of
> TransportChannelImpl and TransportChannel, so that the DTLS-related methods can
> be separated from P2PTransportChannel.
>
> BUG=none
>
> Review-Url: https://codereview.webrtc.org/2590063002
> Cr-Commit-Position: refs/heads/master@{#15743}
> Committed: 12749d89d9

TBR=deadbeef@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=none

Review-Url: https://codereview.webrtc.org/2594343002
Cr-Commit-Position: refs/heads/master@{#15751}
2016-12-22 07:52:00 +00:00
kjellander
d943c48454 Revert of Refactor webrtc/modules/desktop_capture for GN check (patchset #1 id:1 of https://codereview.webrtc.org/2593713002/ )
Reason for revert:
Apparently breaks Chromium compile for unknown reason:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/12314

Original issue's description:
> Refactor webrtc/modules/desktop_capture for GN check
>
> This moves some GN check configurations out of .gn to individual
> targets.
>
> The now checked target is:
> "//webrtc/modules/desktop_capture/*"
>
> BUG=webrtc:6828
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2593713002
> Cr-Commit-Position: refs/heads/master@{#15725}
> Committed: 70870b9211

TBR=sergeyu@chromium.org,mbonadei@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828

Review-Url: https://codereview.webrtc.org/2597923002
Cr-Commit-Position: refs/heads/master@{#15750}
2016-12-22 07:19:59 +00:00
zhihuang
dcccda7e7c Created a java wrapper for the callback OnAddTrack to PeerConnection.Observer
Created a java wrapper for the callback OnAddTrack in this CL since it has been added to native C++ API
The callback function is called when a track is signaled by remote side and a new RtpReceiver is created.
The application can tell when tracks are added to the streams by listening to this callback.

BUG=webrtc:6112

Review-Url: https://codereview.webrtc.org/2513723002
Cr-Commit-Position: refs/heads/master@{#15745}
2016-12-21 22:08:03 +00:00
zhihuang
12749d89d9 Make P2PTransportChannel inherit from IceTransportInternal.
Make P2PTransportChannel inherit from IceTransportInternal instead of
TransportChannelImpl and TransportChannel, so that the DTLS-related methods can
be separated from P2PTransportChannel.

BUG=none

Review-Url: https://codereview.webrtc.org/2590063002
Cr-Commit-Position: refs/heads/master@{#15743}
2016-12-21 18:26:18 +00:00
brandtr
b29e652b10 Revert "Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ )"
Problem fixed: RTP header extensions were not properly set in tests.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2593963003
Cr-Commit-Position: refs/heads/master@{#15741}
2016-12-21 14:37:18 +00:00
hbos
23368e1aef RTCStatsCollectorTest: ExpectReportContainsCertificateInfo /w EXPECT_EQ
Modify ExpectReportContainsCertificateInfo to use EXPECT_EQ checks of
RTCCertificateStats objects.

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2594553003
Cr-Commit-Position: refs/heads/master@{#15738}
2016-12-21 12:29:17 +00:00
hbos
c42ba32877 RTCStatsCollectorTest: Remove ExpectReportContainsCandidate.
Remove ExpectReportContainsCandidate in favor of EXPECT_EQ checks of
RTC[Local/Remote]IceCandidateStats objects.

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2594753002
Cr-Commit-Position: refs/heads/master@{#15737}
2016-12-21 11:31:45 +00:00
nisse
e55b16c664 Drop unneeded include of media_file.h.
BUG=None

Review-Url: https://codereview.webrtc.org/2587403002
Cr-Commit-Position: refs/heads/master@{#15736}
2016-12-21 11:05:44 +00:00
brandtr
504b95eff8 Avoid creating receiver_time outliers in the VideoAnalyzer.
Prior to this change, the receiver_time metric had huge outliers
whenever FlexFEC was enabled. This was due to a measurement problem,
where the time of the incoming packet was incorrectly set to zero.
This happened for packets that were lost in transit, but recovered
through FEC.

This CL fixes this problem by simply not recording samples where the
incoming packet time is undefined. The CL also removes the possibility
of timestamp collisions in the data structures.

TESTED=Ran './webrtc_perf_tests --gtest_filter="*ForemanCifPlr5H264Flexfec*" | grep receiver_time' locally 10 times, without experiencing any outliers.
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2596793002
Cr-Commit-Position: refs/heads/master@{#15735}
2016-12-21 10:54:35 +00:00
mbonadei
d39e16ac30 Revert of Refactor webrtc/modules/video_processing for GN check (patchset #3 id:40001 of https://codereview.webrtc.org/2595543002/ )
Reason for revert:
This CL broke some buildbots. I will investigate it later.

Original issue's description:
> Refactor webrtc/modules/video_processing for GN check
>
> This moves some GN check configurations out of .gn to individual
> targets.
>
> The now checked target is:
> "//webrtc/modules/video_processing/*"
>
> BUG=webrtc:6828
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2595543002
> Cr-Commit-Position: refs/heads/master@{#15732}
> Committed: 00a810b844

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828

Review-Url: https://codereview.webrtc.org/2594973002
Cr-Commit-Position: refs/heads/master@{#15733}
2016-12-21 10:18:54 +00:00
mbonadei
00a810b844 Refactor webrtc/modules/video_processing for GN check
This moves some GN check configurations out of .gn to individual
targets.

The now checked target is:
"//webrtc/modules/video_processing/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2595543002
Cr-Commit-Position: refs/heads/master@{#15732}
2016-12-21 10:01:26 +00:00
hbos
dbb64d8f27 RTCStatsCollectorTest: Remove ExpectReportContainsDataChannel.
Remove ExpectReportContainsDataChannel in favor of EXPECT_EQ checks of
RTCDataChannelStats objects.

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2597433002
Cr-Commit-Position: refs/heads/master@{#15731}
2016-12-21 09:57:46 +00:00
hbos
02d2a92d92 RTCStatsReport::AddStats DCHECKs that the ID is unique.
Previously it was allowed to call AddStats with stats of the same ID
multiple times.

This revealed a few things:
- Local and remote streams can have the same label.
  RTCMediaStreamStats's ID is updated to include "local"/"remote".
- The same certificate can show up multiple times (e.g. for local and
  remote in a loopback), so we skip creating RTCCertificateStats for the
  same certificate multiple times

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2593503003
Cr-Commit-Position: refs/heads/master@{#15730}
2016-12-21 09:29:05 +00:00
mbonadei
2a495ca297 Refactor webrtc/modules/pacing for GN check
This moves some GN check configurations out of .gn to individual
targets.

The now checked target is:
"//webrtc/modules/pacing/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2594523003
Cr-Commit-Position: refs/heads/master@{#15729}
2016-12-21 08:26:58 +00:00
nisse
01c715096f Move nat-related code to target rtc_base_tests_utils.
BUG=None

Review-Url: https://codereview.webrtc.org/2591733002
Cr-Commit-Position: refs/heads/master@{#15728}
2016-12-21 08:23:08 +00:00
brandtr
70e4053844 Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ )
Reason for revert:
Unexpected perf regressions.

Original issue's description:
> Parse FlexFEC RTP headers in Call and add integration with BWE.
>
> BUG=webrtc:5654
>
> Review-Url: https://codereview.webrtc.org/2553863003
> Cr-Commit-Position: refs/heads/master@{#15709}
> Committed: ab2ffa3b28

TBR=philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2589393002
Cr-Commit-Position: refs/heads/master@{#15727}
2016-12-21 08:22:03 +00:00
mbonadei
ba96730bd8 Refactor webrtc/modules/media_file for GN check
This moves some GN check configurations out of .gn to individual
targets.

The now checked target is:
"//webrtc/modules/media_file/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2593693002
Cr-Commit-Position: refs/heads/master@{#15726}
2016-12-21 08:20:52 +00:00
mbonadei
70870b9211 Refactor webrtc/modules/desktop_capture for GN check
This moves some GN check configurations out of .gn to individual
targets.

The now checked target is:
"//webrtc/modules/desktop_capture/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2593713002
Cr-Commit-Position: refs/heads/master@{#15725}
2016-12-21 07:42:05 +00:00
deadbeef
fe4a8a41ad Implement current/pending session description methods.
BUG=webrtc:6917

Review-Url: https://codereview.webrtc.org/2590753002
Cr-Commit-Position: refs/heads/master@{#15722}
2016-12-21 01:56:17 +00:00
erikchen
494dff4c07 Fix a screen capture issue on retina macOS devices.
The CGDisplayStream API returns rects in physical pixel coordinates, not
Density-Independent Pixel coordinates. The code was incorrectly re-applying the
dip_to_pixel scaling.

BUG=chromium:675490

Review-Url: https://codereview.webrtc.org/2588973002
Cr-Commit-Position: refs/heads/master@{#15720}
2016-12-21 01:00:22 +00:00
peah
1b08dc33eb To verify the upcoming code changes it is required
that the level of the output in the audio processing
module is monitored. This CL adds that.

BUG=webrtc:6181, webrtc:6183, webrtc:6220

Review-Url: https://codereview.webrtc.org/2549143004
Cr-Commit-Position: refs/heads/master@{#15718}
2016-12-20 21:45:58 +00:00
stefan
0838327ec9 Add method needed to extract frame capture and arrival timestamps from rtc event logs.
BUG=None

Review-Url: https://codereview.webrtc.org/2557073002
Cr-Commit-Position: refs/heads/master@{#15717}
2016-12-20 16:51:52 +00:00
stefan
64427e563e Add back video_replay. Disappeared in the gn conversion.
BUG=webrtc:6323

Review-Url: https://codereview.webrtc.org/2595533002
Cr-Commit-Position: refs/heads/master@{#15715}
2016-12-20 15:26:58 +00:00
magjed
a3f2d30182 Remove media/base header files from rtc_media target
Chromium has now been updated, so we can remove the base headers from
rtc_media.

BUG=None

Review-Url: https://codereview.webrtc.org/2590813002
Cr-Commit-Position: refs/heads/master@{#15712}
2016-12-20 13:26:59 +00:00
nisse
306127635e Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros.
TBR=pthatcher@webrtc.org
BUG=webrtc:6733

Review-Url: https://codereview.webrtc.org/2515653002
Cr-Commit-Position: refs/heads/master@{#15711}
2016-12-20 13:03:58 +00:00
philipel
022b54e86a Wire up H264 fmtp sprop-parameter-sets with H264SpsPpsTracker.
To avoid making this CL large unittests will be added in a followup CL.

BUG=webrtc:5948

patch from issue 2570073002 at patchset 20001 (http://crrev.com/2570073002#ps20001)

Review-Url: https://codereview.webrtc.org/2565173009
Cr-Commit-Position: refs/heads/master@{#15710}
2016-12-20 12:15:59 +00:00
brandtr
ab2ffa3b28 Parse FlexFEC RTP headers in Call and add integration with BWE.
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2553863003
Cr-Commit-Position: refs/heads/master@{#15709}
2016-12-20 11:33:58 +00:00
nisse
b36ee8d498 New method StatsObserver::OnCompleteReports, passing ownership.
The new name, OnCompleteReports rather than OnComplete, is needed
because in C++ method lookup, overriding a method hides all otherwise
inherited methods with the same name, even if they have a different
signature. And here, the intention is that each subclass should
override one or the other of the two methods, and inherit the method it
doesn't override.

This cl is a prerequisite for
https://codereview.webrtc.org/2567143003/, because the Chrome glue
code needs to retain the stats report after the OnComplete method has
returned.

Currently, Chrome makes a copy of the stats mapping (which breaks when
changing ValuePtr from an rtc::linked_ptr to an std::unique_ptr). After
this cl, Chrome can be fixed to take ownership and no longer needs to
copy anything, unblocking cl 2567143003.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2584553002
Cr-Commit-Position: refs/heads/master@{#15708}
2016-12-20 11:30:00 +00:00
nisse
5206667dad Delete unused method PayloadRouter::MaxPayloadLength.
Documentation was also unclear, it seems it returned the RTP packet
size including RTP headers.

BUG=None.

Review-Url: https://codereview.webrtc.org/2588343002
Cr-Commit-Position: refs/heads/master@{#15707}
2016-12-20 11:12:04 +00:00
danilchap
8bab796db7 Style cleanup in RTCPReceiver
Rename variables and private functions to follow style,
replace remaining asserts with DCHECKs.
add 'ms' suffix to time variables derived from clock_
add 'ntp' suffix to time variables derived from ntp time.
No functional changes expected.

BUG=None

Review-Url: https://codereview.webrtc.org/2588753002
Cr-Commit-Position: refs/heads/master@{#15706}
2016-12-20 10:46:46 +00:00
magjed
d5236e2948 Revert of Add disabled certificate check support to IceServer PeerConnection API. (patchset #8 id:140001 of https://codereview.webrtc.org/2557803002/ )
Reason for revert:
This CL broke all Chromium WebRTC FYI bots. A roll+fix was attempted here: https://codereview.chromium.org/2590783003/, but failed to land. I'm reverting this CL now to make the tree green again. Make the API change gradual when you reland so that we can update Chromium between.

Original issue's description:
> Add disabled certificate check support to IceServer PeerConnection API.
>
> Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
> that it's not actually some kind of SSL over TCP. Also making it clear
> that it's mutually exclusive with OPT_TLS.
>
> Add "OPT_TLS_INSECURE" that implements the new certificate-check
> disabled TLS mode, which is also mutually exclusive with the other
> TLS options.
>
> PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
> the new insecure mode and added it as a RelayCredentials member.
>
> TurnPort: Add new TLS policy member with appropriate getter and setter
> to avoid constructor bloat. Initialize it from the RelayCredentials
> after the TurnPort is created.
>
> Expose the new feature in the PeerConnection API via
> IceServer.tls_certificate_policy as well as via the Android JNI
> PeerConnection API.
>
> For security reasons we ensure that:
>
> 	1) The policy is always explicitly initialized to secure.
>         2) API users have to explicitly integrate with the feature to
>            use it, and will otherwise get no change in behavior.
> 	3) The feature is not immediately exposed in non-native
> 	   contexts. For example, disabling of certificate validation
>            is not implemented via URI parsing since this would
>            immediately allow it to be used from a web page.
>
> BUG=webrtc:6840
>
> Review-Url: https://codereview.webrtc.org/2557803002
> Cr-Commit-Position: refs/heads/master@{#15670}
> Committed: b0f04fdb9e

TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,hnsl@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6840

Review-Url: https://codereview.webrtc.org/2590153002
Cr-Commit-Position: refs/heads/master@{#15703}
2016-12-20 10:22:06 +00:00
henrik.lundin
59dbfe6d32 Add a unit test for Opus complexity adaptation
The test verifies that the hysteresis window in the configuration works
as intended.

BUG=webrtc:6708

Review-Url: https://codereview.webrtc.org/2594563002
Cr-Commit-Position: refs/heads/master@{#15700}
2016-12-20 09:17:55 +00:00
zhihuang
86abd6f2fd Add an abstract class for IceTransport
P2PTransportChannl will be renamed to IceTransport and this class will be
the base class of IceTransport. By doing this, the Dtls related methods
can be separated from the IceTransport.

For more detail, https://docs.google.com/document/d/1g9RA0s4RV7hFAcWiAM2b6H5ZohAVpBqcYienEDj6IcY/edit

BUG=none

Review-Url: https://codereview.webrtc.org/2577183004
Cr-Commit-Position: refs/heads/master@{#15690}
2016-12-19 19:54:05 +00:00
brandtr
8b5c345ee5 Add GUARDED_BY's in FlexfecReceiver.
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2589583004
Cr-Commit-Position: refs/heads/master@{#15688}
2016-12-19 18:02:30 +00:00
brandtr
bb7066f966 Clean up storage of FlexFEC payload type in webrtc::VideoCodecSettings.
No need to pass a whole struct around, when only one member is used.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2589833002
Cr-Commit-Position: refs/heads/master@{#15687}
2016-12-19 17:41:04 +00:00
danilchap
0ad21111fc Revert of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. (patchset #1 id:1 of https://codereview.webrtc.org/2574943003/ )
Reason for revert:
breaks downstream project.

Can you make this change in a compatible way using anonymous union:
union {
  bool is_first_packet_in_frame;
  RTC_DEPRECATED bool isFirstPacket;
};
(unfortunetly this this treak breaks braced initialization in rtp_rtcp_impl_unittest.cc,
so that should be rewritting in a more classic way)

Original issue's description:
> Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
>
> Name should represent the actual meaning.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2574943003
> Cr-Commit-Position: refs/heads/master@{#15684}
> Committed: efde908380

TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2589783003
Cr-Commit-Position: refs/heads/master@{#15686}
2016-12-19 17:36:33 +00:00
hbos
7472dc3b94 Removed undefined method from webrtcsession.h.
It was accidentally added in https://codereview.webrtc.org/2583883002/
(added in one patch set, removed in another but forgot about the header
declaration).

BUG=webrtc:6875, chromium:627816

Review-Url: https://codereview.webrtc.org/2583123003
Cr-Commit-Position: refs/heads/master@{#15685}
2016-12-19 17:34:14 +00:00
johan
efde908380 Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
Name should represent the actual meaning.

BUG=None

Review-Url: https://codereview.webrtc.org/2574943003
Cr-Commit-Position: refs/heads/master@{#15684}
2016-12-19 16:32:24 +00:00