Commit Graph

605 Commits

Author SHA1 Message Date
1c392cc5cf Avoid rescheduling the next RTCP packet if the RTCP sender status doesn't change.
The change made in https://codereview.webrtc.org/1757683002 introduced an extra call to RTCPSender::SetRTCPStatus after the video receive stream is created. The SetRTCPStatus call results in no state change, as the RTCP sender is already enabled, however, it reschedules the next RTCP packet to be RTCP_INTERVAL_VIDEO_MS/2 (500) ms in the future.
Before the change, the next packet time was only set by the previous call to RTCPSender::SetSSRC, which placed it 100 ms in the future. The change, therefore, changed the timing of multiple performance tests - as it now takes a different length of time to ramp up to the same bandwidth.

BUG=chromium:597332

Review URL: https://codereview.webrtc.org/1826093004

Cr-Commit-Position: refs/heads/master@{#12203}
2016-04-01 21:46:54 +00:00
86101e9c08 Remove deprecated RtpReceiver::CreateAudioReceiver() function.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1848813003

Cr-Commit-Position: refs/heads/master@{#12192}
2016-04-01 08:01:33 +00:00
4a206a96c1 Remove webrtc::ScopedVector
We can (and should) use std::vector<std::unique_ptr<T>> instead.
Because it's standard, and because it's safer since callers have to
manually wrap elements in std::unique_ptr before inserting them and
manually unwrap them after inserting them.

Review URL: https://codereview.webrtc.org/1839603002

Cr-Commit-Position: refs/heads/master@{#12182}
2016-03-31 17:24:31 +00:00
4b9cad86ce [rtcp] Sdes::Parse updated not to use RTCPUtility
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1846433003

Cr-Commit-Position: refs/heads/master@{#12169}
2016-03-30 20:34:38 +00:00
41befcee7d Make rtcp sender use max transfer unit.
Remove packet overhead from rtp sender as unused.

R=philipel, åsapersson

Review URL: https://codereview.webrtc.org/1827953002

Cr-Commit-Position: refs/heads/master@{#12165}
2016-03-30 18:11:55 +00:00
367bbbf76d [rtcp] ReceiverReport::Parse updated not to use RTCPUtility
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1837213002

Cr-Commit-Position: refs/heads/master@{#12164}
2016-03-30 16:43:09 +00:00
f1f87203d7 Split ByteBuffer into writer/reader objects.
This allows the reader to reference data, thus avoiding unnecessary
allocations and memory copies.

BUG=webrtc:5155,webrtc:5670

Review URL: https://codereview.webrtc.org/1821083002

Cr-Commit-Position: refs/heads/master@{#12160}
2016-03-30 13:43:44 +00:00
1d0313916b Reland https://codereview.webrtc.org/1802993002/
Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.

BUG=webrtc:4690

Committed: https://crrev.com/69a81999ace08e40e2b2ec526b0e111aa11b9538
Cr-Commit-Position: refs/heads/master@{#12015}

Review URL: https://codereview.webrtc.org/1840893004

Cr-Commit-Position: refs/heads/master@{#12157}
2016-03-30 09:42:37 +00:00
2f36c2399e [rtcp] SenderReport::Parse updated not to use RTCPUtility
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1825353002

Cr-Commit-Position: refs/heads/master@{#12140}
2016-03-29 15:02:33 +00:00
f752f85f3d [rtcp] Pli::Parse updated not to use RTCPUtility
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1811933002

Cr-Commit-Position: refs/heads/master@{#12104}
2016-03-23 15:25:30 +00:00
56036ffc45 cleanup RTCPSender
rtc::scoped_ptr -> std::unique_ptr.
CrticalSectionWrapper -> CriticalSection.
assert -> DCHECK.
removed unused headers.
removed unused using.
removed unused member field.

BUG=webrtc:5520, webrtc:5565
R=åsapersson

Review URL: https://codereview.webrtc.org/1806603002

Cr-Commit-Position: refs/heads/master@{#12091}
2016-03-22 18:14:16 +00:00
1d1944187f Replace RefCountImpl with rtc::RefCountedObject.
Removes code duplication and use of the dangerous public destructor in
RefCountImpl.

Also making wider use of scoped_refptr and fixing various leaks in the
process.

BUG=webrtc:5229
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1477013005 .

Cr-Commit-Position: refs/heads/master@{#12075}
2016-03-21 15:44:41 +00:00
5f0b83b7fb Enabling rtcp-rsize negotiation and fixing some issues with it.
Sending of reduced size RTCP packets should be enabled only if it's
enabled in the send parameters (which corresponds to the remote description).

Since the RTCPReceiver's RtcpMode isn't used at all, I removed it to ease
confusion.

BUG=webrtc:4868
R=pbos@webrtc.org, pthatcher@google.com, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1713493003 .

Cr-Commit-Position: refs/heads/master@{#12057}
2016-03-18 22:02:13 +00:00
94a23f04af Reland "Add check_deps rules in DEPS files."
Relanding https://codereview.webrtc.org/1796413002/
without the change to the openmax_dl include path
(which broke downstream code).

TBR=tommi@webrtc.org
BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc

Review URL: https://codereview.webrtc.org/1804333002 .

Cr-Commit-Position: refs/heads/master@{#12031}
2016-03-17 11:05:50 +00:00
b69395b374 Revert of Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code. (patchset #2 id:20001 of https://codereview.webrtc.org/1802993002/ )
Reason for revert:
Revert because it breaks downstream code.

Original issue's description:
> Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/69a81999ace08e40e2b2ec526b0e111aa11b9538
> Cr-Commit-Position: refs/heads/master@{#12015}

TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1812453002

Cr-Commit-Position: refs/heads/master@{#12016}
2016-03-16 14:05:21 +00:00
69a81999ac Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1802993002

Cr-Commit-Position: refs/heads/master@{#12015}
2016-03-16 12:59:04 +00:00
56cf60e717 Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
Reason for revert:
The openmax_dl include change breaks downstream projects.

Original issue's description:
> Add check_deps rules in DEPS files.
>
> Add fine-grained check_deps rules for all of WebRTC.
> This will help both maintaining sane dependencies and provides a way
> to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
>
> Example:
> buildtools/checkdeps/graphdeps.py --root=. --format=png \
> --out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
> --excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
>
> will produce a neat webrtc.png image showcasing the dependencies
> (according to the DEPS file) for the bitrate_controller module.
> Some dependencies are filtered out for readability.
>
> BUG=webrtc:5623
> TESTED=Passing runs using:
> buildtools/checkdeps/checkdeps.py --root=. talk
> buildtools/checkdeps/checkdeps.py --root=. webrtc
>
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/086f851b7b9b4bcbd4fe507c3bf83b760bd7f4d9
> Cr-Commit-Position: refs/heads/master@{#12008}

TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5623

Review URL: https://codereview.webrtc.org/1808573002

Cr-Commit-Position: refs/heads/master@{#12009}
2016-03-16 00:41:04 +00:00
086f851b7b Add check_deps rules in DEPS files.
Add fine-grained check_deps rules for all of WebRTC.
This will help both maintaining sane dependencies and provides a way
to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.

Example:
buildtools/checkdeps/graphdeps.py --root=. --format=png \
--out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
--excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'

will produce a neat webrtc.png image showcasing the dependencies
(according to the DEPS file) for the bitrate_controller module.
Some dependencies are filtered out for readability.

BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc

R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1796413002 .

Cr-Commit-Position: refs/heads/master@{#12008}
2016-03-16 00:22:53 +00:00
6021fe2b1e Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1803923003

Cr-Commit-Position: refs/heads/master@{#12003}
2016-03-15 18:41:58 +00:00
7021b92525 introduced rtcp::CommonHeader class
this class replace and extend RTCPUtility::RtcpCommonHeader structure and RTCPUtility::RtcpParseCommonHeader function.
In addition to header fields, payload pointer is stored because rtcp header without payload is rarely useful.
Sample usage can be checked in 'RTCP Parser sketched' CL: https://codereview.webrtc.org/1555683002/

BUG=webrtc:5260
R=asapersson@webrtc.org, sprang@webrtc.org

Review URL: https://codereview.webrtc.org/1575413002 .

Cr-Commit-Position: refs/heads/master@{#11999}
2016-03-15 16:39:45 +00:00
f8cdd184d5 Add histogram stats for AV sync stream offset:
"WebRTC.Video.AVSyncOffsetInMs"

The absolute value of the sync offset between a rendered video frame and the latest played audio frame is measured per video frame. The average offset per received video stream is recorded when a stream is removed.

Updated sync tests in call_perf_tests.cc to use this implementation.

BUG=webrtc:5493

Review URL: https://codereview.webrtc.org/1756193005

Cr-Commit-Position: refs/heads/master@{#11993}
2016-03-15 08:00:54 +00:00
7a4116aa54 [rtp_rtcp] Append functionality moved from base RtcpPacket class to CompoundPacket
BUG=webrtc:5260

Review URL: https://codereview.webrtc.org/1582323005

Cr-Commit-Position: refs/heads/master@{#11981}
2016-03-14 15:19:31 +00:00
83f831a919 Experiment for the nack module.
Testing the nack module by implementing it into the current jitter buffer
under the experiment WebRTC-NewVideoJitterBuffer.

BUG=webrtc:5514

Review URL: https://codereview.webrtc.org/1778503002

Cr-Commit-Position: refs/heads/master@{#11969}
2016-03-12 11:30:31 +00:00
8842c3e41b Relanding https://codereview.webrtc.org/1715883002/ in pieces.
- Use better types in AudioSendStream::SendTelephoneEvent() and related methods.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1782053002

Cr-Commit-Position: refs/heads/master@{#11953}
2016-03-11 11:06:48 +00:00
50da1d329a Fixed busy loop in case of partially malformed rtcp packet
BUG=webrtc:5574

Review URL: https://codereview.webrtc.org/1726573003

Cr-Commit-Position: refs/heads/master@{#11946}
2016-03-10 21:13:57 +00:00
3ecb5c8698 Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ )
Reason for revert:
Breaks Chromium FYI bots for Android. E.g. https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/4486/steps/content_browsertests/logs/stdio

Original issue's description:
> - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs.
> - Use better types in AudioSendStream::SendTelephoneEvent() and related methods.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/8886c816582a7c6190c5429222cb8096fca302a6
> Cr-Commit-Position: refs/heads/master@{#11927}

TBR=tina.legrand@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1776243003

Cr-Commit-Position: refs/heads/master@{#11930}
2016-03-09 15:32:05 +00:00
c1e55c7136 rtt calculation handles time go backwards
CompactNtpIntervalToMs renamed to CompactNtpRttToMs and handle special cases:
large values consider negative/invalid and result in value of 1.
0 result consider too small and increases to 1.

BUG=590996
R=asapersson@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1763823003 .

Cr-Commit-Position: refs/heads/master@{#11928}
2016-03-09 14:14:45 +00:00
8886c81658 - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs.
- Use better types in AudioSendStream::SendTelephoneEvent() and related methods.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1722253002

Cr-Commit-Position: refs/heads/master@{#11927}
2016-03-09 11:32:53 +00:00
96150a6322 [cleanup] fixed macros and includes in rtp_cvo.h
assert macro replaced with RTC_NOTREACHED, added proper include for it.

R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1763973003 .

Cr-Commit-Position: refs/heads/master@{#11886}
2016-03-07 09:55:28 +00:00
b65f3e39d7 [cleanup] Remove unused fields/functions from rtcp module.
Removed fields are initialized but unused.
Removed functions are not called, sometimes are not defined.

BUG=webrtc:5565
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1726403003 .

Cr-Commit-Position: refs/heads/master@{#11839}
2016-03-02 12:26:19 +00:00
a4f31bd03a TMMBRSet become vector<rtcp::TmmbItem>
this is a slice of https://codereview.webrtc.org/1474693002/
All TMMBRSet functions intentionally left unchanged. Goal to make them obsolete, not to clear.

BUG=webrtc:5565

Review URL: https://codereview.webrtc.org/1669323002

Cr-Commit-Position: refs/heads/master@{#11813}
2016-02-29 13:26:05 +00:00
8b79b07a55 Move RTP module activation into PayloadRouter.
Simplifies PayloadRouter to not accept dynamically-changing modules as
well as usage of PayloadRouter inside ViEChannel::SetSendCodec.

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

Review URL: https://codereview.webrtc.org/1725363003 .

Cr-Commit-Position: refs/heads/master@{#11787}
2016-02-26 15:31:44 +00:00
9c01725e37 Simplify registration of RTP-header extensions.
Removes per-extension functions in ViEChannel/ViEReceiver and instead
register extensions directly on the RTP module by mapping extension
string to RTP-header-extension type.

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

Review URL: https://codereview.webrtc.org/1740133002 .

Cr-Commit-Position: refs/heads/master@{#11786}
2016-02-26 15:26:29 +00:00
c63f79a0a5 Fix ubsan warning in byteio_unittest
BUG=webrtc:5490

Review URL: https://codereview.webrtc.org/1739753002

Cr-Commit-Position: refs/heads/master@{#11782}
2016-02-26 13:13:51 +00:00
a2644c06ee Disable tests failing under UBSan to enable deployment to main waterfall.
modules_unittests: https://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20UBSan/builds/1138/steps/modules_unittests/logs/stdio
[ RUN      ] ByteIoTest.Test64SBitBigEndian
../../webrtc/modules/rtp_rtcp/source/byte_io_unittest.cc:34:33: runtime error: shift exponent 64 is too large for 64-bit type 'long'

rtc_unittests: https://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20UBSan/builds/1138/steps/rtc_unittests/logs/stdio
[ RUN      ] IPAddressTest.TestCountIPMaskBits
../../webrtc/base/ipaddress.cc:415:20: runtime error: negation of -2147483648 cannot be represented in type 'int32_t' (aka 'int'); cast to an unsigned type to negate this value to itself

[ RUN      ] BandwidthSmootherTest.TestSampleRollover
../../webrtc/base/rollingaccumulator.h:73:22: runtime error: signed integer overflow: 2147483647 * 2147483647 cannot be represented in type 'int'

[ RUN      ] RandomNumberGeneratorTest.UniformSignedInterval
../../webrtc/base/random_unittest.cc:121:50: runtime error: signed integer overflow: 2147483647 - -2147483648 cannot be represented in type 'int'

rtc_media_unittests: https://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20UBSan/builds/1138/steps/rtc_media_unittests/logs/stdio
[ RUN      ] VideoCommonTest.TestComputeScaleWithHighFps
../../webrtc/media/base/videocommon.cc:75:34: runtime error: signed integer overflow: 2621440 - -2147483648 cannot be represented in type 'int'

BUG=webrtc:5487, webrtc:5490, webrtc:5491
NOTRY=True
R=pbos@webrtc.org
TBR=henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1727233005 .

Cr-Commit-Position: refs/heads/master@{#11764}
2016-02-25 13:23:29 +00:00
a26ac925f7 Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ )
Reason for revert:
Revert breaks other uses, a fix will be rolled into Chromium instead.

Original issue's description:
> Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ )
>
> Reason for revert:
> Breaks Chromium.
>
> Original issue's description:
> > Remove ignored return code from modules.
> >
> > ModuleProcessImpl doesn't act on return codes and having them around is
> > confusing (it's unclear what an error return code here would do even).
> >
> > BUG=
> > R=tommi@webrtc.org
> >
> > Committed: f14c47a58c
>
> TBR=tommi@webrtc.org,pbos@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/da33a8a2a22f6d19ba2a8cce963beafbdbaa8fd8
> Cr-Commit-Position: refs/heads/master@{#11761}

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

Review URL: https://codereview.webrtc.org/1737013002

Cr-Commit-Position: refs/heads/master@{#11762}
2016-02-25 12:50:09 +00:00
da33a8a2a2 Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ )
Reason for revert:
Breaks Chromium.

Original issue's description:
> Remove ignored return code from modules.
>
> ModuleProcessImpl doesn't act on return codes and having them around is
> confusing (it's unclear what an error return code here would do even).
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: f14c47a58c

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

Review URL: https://codereview.webrtc.org/1736663004

Cr-Commit-Position: refs/heads/master@{#11761}
2016-02-25 12:34:12 +00:00
f6ff9714c0 Fix division by zero in FindTMMBRBoundingSet
BUG=webrtc:5490

Review URL: https://codereview.webrtc.org/1727273003

Cr-Commit-Position: refs/heads/master@{#11749}
2016-02-24 17:23:57 +00:00
f14c47a58c Remove ignored return code from modules.
ModuleProcessImpl doesn't act on return codes and having them around is
confusing (it's unclear what an error return code here would do even).

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1703833002 .

Cr-Commit-Position: refs/heads/master@{#11747}
2016-02-24 15:51:23 +00:00
a094fd1550 RTT intermediate calculation use ntp time instead of milliseconds.
Compact NTP representation was designed exactly for that purpose: calculate RTT. No need to map to ms before doing arithmetic on this values.
  Because of this change there is no need to keep mapping between compact ntp presentation and milliseconds in the RTCPSender.

BUG=webrtc:5565
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1491843004 .

Cr-Commit-Position: refs/heads/master@{#11710}
2016-02-22 17:59:47 +00:00
e2d83d6560 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()
Also move some stats reporting from vie_channel to send stats proxy

BUG=

Review URL: https://codereview.webrtc.org/1669623004

Cr-Commit-Position: refs/heads/master@{#11688}
2016-02-19 17:03:34 +00:00
fc968a283c Fix sequence-number replay race for padding.
Prevents allocating sequence numbers for packets that go out on the
network even though sending media is disabled.

This race caused a replay of sequence numbers when GetRtpState() on a
stopped stream would not return the last sequence number sent, since the
pacer thread could request and send padding on a later sequence number
before the modules are disconnected from the pacer.

BUG=webrtc:5543
R=stefan@webrtc.org
TEST=Repeating EndToEndTest.RestartingSendStreamPreservesRtpState 1000 times under TSan.

Review URL: https://codereview.webrtc.org/1715703002 .

Cr-Commit-Position: refs/heads/master@{#11685}
2016-02-19 15:14:44 +00:00
1e80ce438e webrtc::RtpPacket name freed for better RtpPacket
There were two different structures named RtpPacket in webrtc namespace:
RtpPacket defined in fec_test_helper renamed to test::RawRtpPacket
RtpPacket defined in rtp_sender_video and producer_fec removed as unused

BUG=webrtc:5261
R=sprang@google.com, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1710103004 .

Cr-Commit-Position: refs/heads/master@{#11682}
2016-02-19 15:02:24 +00:00
0e2e50ca1c Always append the BYE packet type at the end
When composing a RTCP packet, if there is a BYE
to be appended, preserve it and append it at the
end after all other packet types are added.

BUG=webrtc:5498
NOTRY=true

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

Cr-Commit-Position: refs/heads/master@{#11672}
2016-02-18 16:33:33 +00:00
69e59e619a [rtp_rtcp] rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer
rtcp::RawPacket is rtc::Buffer, it had no extra functionality.
rtc::Buffer is a movable class - no point to wrap it into rtc::scoped_ptr
change is large, but straightforward:
  rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer
  ->Buffer() replaced with .data()
  ->Length() replaced with .size()

BUG=webrtc:5260

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

Cr-Commit-Position: refs/heads/master@{#11649}
2016-02-17 11:11:50 +00:00
0206000a66 iOS: Add resource files for tests and implement OutputPath
With this change the following tests have been successfully
passing in the iOS Simulator for iPhone 5 and iOS 9:
* audio_decoder_unittests
* common_video_unittests
* modules_tests
* rtc_api_objc_tests
* rtc_pc_unittests
* system_wrappers_unittests
* voice_engine_unittests

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

BUG=webrtc:4755
NOTRY=True

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

Cr-Commit-Position: refs/heads/master@{#11646}
2016-02-17 06:06:17 +00:00
9d0c43242b Remove video-codec max bitrate from TMMBN.
TMMBN was capped by configured max bitrate for no apparent reason.
Removing this to not require payload-type reconfiguration on new
video-codec settings. Actual removal of payload-type reconfiguration
will happen in a pending CL.

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

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

Cr-Commit-Position: refs/heads/master@{#11639}
2016-02-16 16:59:36 +00:00
9f35d55c58 Added accessor and Parse function.
Create function merged into one.

BUG=webrtc:5260

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

Cr-Commit-Position: refs/heads/master@{#11581}
2016-02-11 16:19:06 +00:00
09fef9e6f7 [rtp_rtcp] Added Sender Report Request rtcp packet.
BUG=webrtc:5260
R=åsapersson

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

Cr-Commit-Position: refs/heads/master@{#11538}
2016-02-09 13:57:56 +00:00
c0ae305a9e Fix null-pointer dereference in RTPSenderVideo.
Since the address of the dereference is taken this inputs a garbage
almost-null pointer into RtpPacketizer. Not likely that a load/store is
performed on the address, but UBSan fires and it's a source of potential
future errors.

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

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

Cr-Commit-Position: refs/heads/master@{#11528}
2016-02-08 14:00:22 +00:00