Commit Graph

1080 Commits

Author SHA1 Message Date
84f6a3fc6b Move optional.h to webrtc/api/
We use Optional in our public API, so its header should be in
webrtc/api/.

BUG=webrtc:8205

Review-Url: https://codereview.webrtc.org/3011943002
Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 15:43:13 +00:00
ba09f79ba3 Make UBSan warnings fatal and fix the existing ones
The warnings were (all signed integer overflow):
webrtc/common_audio/signal_processing/levinson_durbin.c:46:25
12 * 268435456 cannot be represented in type 'int'
webrtc/modules/audio_processing/aecm/aecm_core.cc:930:69
522240 * 6115 cannot be represented in type 'int'
webrtc/modules/audio_processing/aecm/aecm_core_c.cc:455:36
72293096 * 50 cannot be represented in type 'int'
webrtc/modules/pacing/alr_detector.cc:70:48
1000000000 * 65 cannot be represented in type 'int'
webrtc/modules/rtp_rtcp/source/rtp_sender.cc:947:20
1929277286 + 321546521 cannot be represented in type 'int'

BUG=webrtc:8195

Review-Url: https://codereview.webrtc.org/3005003002
Cr-Commit-Position: refs/heads/master@{#19670}
2017-09-04 15:32:43 +00:00
a8ae6f2aca Add flag enabling more packets to be retransmittable.
If not indicated otherwise, allow adding a packet to the retransmission
history at least every 1/15s in order to reduce frame dropping.

BUG=webrtc:7694

Review-Url: https://codereview.webrtc.org/2999063002
Cr-Commit-Position: refs/heads/master@{#19665}
2017-09-04 14:23:56 +00:00
529662a44c Move array_view.h to webrtc/api/
We use ArrayView in our public API, so its header should be in
webrtc/api/.

BUG=none

Review-Url: https://codereview.webrtc.org/3007763002
Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 12:43:17 +00:00
1acbd68718 Move RtpExtension to api/ directory and config.h/.cc to call/.
BUG=webrtc:5876
R=deadbeef@webrtc.org, solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3004723002 .
Cr-Commit-Position: refs/heads/master@{#19639}
2017-09-01 13:29:30 +00:00
16adf03d25 Recently we moved webrtc/base to webrtc/rtc_base, so these
directives in our DEPS files are not needed anymore.

Includes from webrtc/rtc_base are also whitelisted in webrtc/DEPS
so we don't have to whitelist it in all the others DEPS files.

BUG=webrtc:7634
NOTRY=True

Review-Url: https://codereview.webrtc.org/3006583002
Cr-Commit-Position: refs/heads/master@{#19601}
2017-08-30 11:45:58 +00:00
6d5b4d6fe1 Piggybacking simulcast id and ALR experiment id into video content type extension.
Use it to slice UMA video receive statis.

BUG=8032

Review-Url: https://codereview.webrtc.org/2986893002
Cr-Commit-Position: refs/heads/master@{#19598}
2017-08-30 10:32:14 +00:00
b32aaf97bd Reland of Verify sender ssrc when receiving rtcp target bitrate (patchset #1 id:1 of https://codereview.webrtc.org/3005633002/ )
Reason for revert:
Landed fix in upstream project.

Original issue's description:
> Revert of Verify sender ssrc when receiving rtcp target bitrate (patchset #3 id:40001 of https://codereview.webrtc.org/3000373002/ )
>
> Reason for revert:
> Might be the root cause of an internal test failure.
>
> Original issue's description:
> > Verify sender ssrc when receiving rtcp target bitrate
> >
> > BUG=webrtc:8137
> >
> > Review-Url: https://codereview.webrtc.org/3000373002
> > Cr-Commit-Position: refs/heads/master@{#19524}
> > Committed: a7a4beb419
>
> TBR=danilchap@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:8137
>
> Review-Url: https://codereview.webrtc.org/3005633002
> Cr-Commit-Position: refs/heads/master@{#19529}
> Committed: 95a64ca8aa

TBR=danilchap@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8137

Review-Url: https://codereview.webrtc.org/3006683002
Cr-Commit-Position: refs/heads/master@{#19557}
2017-08-28 12:49:12 +00:00
95a64ca8aa Revert of Verify sender ssrc when receiving rtcp target bitrate (patchset #3 id:40001 of https://codereview.webrtc.org/3000373002/ )
Reason for revert:
Might be the root cause of an internal test failure.

Original issue's description:
> Verify sender ssrc when receiving rtcp target bitrate
>
> BUG=webrtc:8137
>
> Review-Url: https://codereview.webrtc.org/3000373002
> Cr-Commit-Position: refs/heads/master@{#19524}
> Committed: a7a4beb419

TBR=danilchap@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:8137

Review-Url: https://codereview.webrtc.org/3005633002
Cr-Commit-Position: refs/heads/master@{#19529}
2017-08-25 19:59:42 +00:00
41476e014c When Ulpfec recovers a packet, set |returned| flag earlier.
This avoids infinite recursion in case the recovered packet carries a
RED header.

BUG=chromium:754748

Review-Url: https://codereview.webrtc.org/3004553002
Cr-Commit-Position: refs/heads/master@{#19525}
2017-08-25 16:08:44 +00:00
a7a4beb419 Verify sender ssrc when receiving rtcp target bitrate
BUG=webrtc:8137

Review-Url: https://codereview.webrtc.org/3000373002
Cr-Commit-Position: refs/heads/master@{#19524}
2017-08-25 16:06:16 +00:00
2b706343de Add audio_level member to RtpSource and set it from RtpReceiverImpl::IncomingRtpPacket.
BUG=webrtc:7987

Review-Url: https://codereview.webrtc.org/3000713002
Cr-Commit-Position: refs/heads/master@{#19503}
2017-08-24 21:52:17 +00:00
ba050a6d6d Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )
Reason for revert:
Create reland CL to add fix to.

Original issue's description:
> Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )
>
> Reason for revert:
> Speculative revet for breaking remoting_unittests in fyi bots.
> https://build.chromium.org/p/chromium.webrtc.fyi/waterfall?builder=Win7%20Tester
>
> Original issue's description:
> > Add a flags field to video timing extension.
> >
> > The rtp header extension for video timing shuold have an additional
> > field for signaling metadata, such as what triggered the extension for
> > this particular frame. This will allow separating frames select because
> > of outlier sizes from regular frames, for more accurate stats.
> >
> > This implementation is backwards compatible in that it can read video
> > timing extensions without the new flag field, but it always sends with
> > it included.
> >
> > BUG=webrtc:7594
> >
> > Review-Url: https://codereview.webrtc.org/3000753002
> > Cr-Commit-Position: refs/heads/master@{#19353}
> > Committed: cf5d485e14
>
> TBR=danilchap@webrtc.org,kthelgason@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7594
>
> Review-Url: https://codereview.webrtc.org/2995953002
> Cr-Commit-Position: refs/heads/master@{#19360}
> Committed: f0f7378b05

TBR=danilchap@webrtc.org,kthelgason@webrtc.org,stefan@webrtc.org,emircan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/2996153002
Cr-Commit-Position: refs/heads/master@{#19405}
2017-08-18 09:51:12 +00:00
5b9746ef10 When using clang, switch on -Wc++11-narrowing
See
https://clang.llvm.org/docs/DiagnosticsReference.html#wc-11-narrowing
for datails. This catches a narrowing bug that broke a downstream
project in https://codereview.webrtc.org/2995523002/.

BUG=none

Review-Url: https://codereview.webrtc.org/2995073002
Cr-Commit-Position: refs/heads/master@{#19366}
2017-08-16 11:52:35 +00:00
f0f7378b05 Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )
Reason for revert:
Speculative revet for breaking remoting_unittests in fyi bots.
https://build.chromium.org/p/chromium.webrtc.fyi/waterfall?builder=Win7%20Tester

Original issue's description:
> Add a flags field to video timing extension.
>
> The rtp header extension for video timing shuold have an additional
> field for signaling metadata, such as what triggered the extension for
> this particular frame. This will allow separating frames select because
> of outlier sizes from regular frames, for more accurate stats.
>
> This implementation is backwards compatible in that it can read video
> timing extensions without the new flag field, but it always sends with
> it included.
>
> BUG=webrtc:7594
>
> Review-Url: https://codereview.webrtc.org/3000753002
> Cr-Commit-Position: refs/heads/master@{#19353}
> Committed: cf5d485e14

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

Review-Url: https://codereview.webrtc.org/2995953002
Cr-Commit-Position: refs/heads/master@{#19360}
2017-08-15 19:31:23 +00:00
037f3e42f2 Replace absolute path with relative path for GN files.
Bug: webrtc:7952
Change-Id: I45d889bd976f58386f803d0dc27147ea00a52e56
Reviewed-on: https://chromium-review.googlesource.com/612786
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19357}
2017-08-15 15:57:36 +00:00
cf5d485e14 Add a flags field to video timing extension.
The rtp header extension for video timing shuold have an additional
field for signaling metadata, such as what triggered the extension for
this particular frame. This will allow separating frames select because
of outlier sizes from regular frames, for more accurate stats.

This implementation is backwards compatible in that it can read video
timing extensions without the new flag field, but it always sends with
it included.

BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/3000753002
Cr-Commit-Position: refs/heads/master@{#19353}
2017-08-15 12:33:27 +00:00
ec86be0962 Reduce locking when collecting receive statistic
BUG=None

Review-Url: https://codereview.webrtc.org/2997803002
Cr-Commit-Position: refs/heads/master@{#19336}
2017-08-14 12:51:02 +00:00
0bc8423fe5 Move RtcpReportBlocks implementation from ReceiveStatistics to ReceiveStatisticsImpl
BUG=webrtc:8016

Review-Url: https://codereview.webrtc.org/2997783002
Cr-Commit-Position: refs/heads/master@{#19327}
2017-08-11 15:12:54 +00:00
ee89e7870c Replace CHECK(x && y) with two separate CHECK() calls
That way, the debug printout will tell us which of x and y that was false.

BUG=none

Review-Url: https://codereview.webrtc.org/2988153003
Cr-Commit-Position: refs/heads/master@{#19297}
2017-08-10 00:22:01 +00:00
3e69e5c2c0 Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock
Continues on https://codereview.webrtc.org/2992043002

BUG=webrtc:8033

Review-Url: https://codereview.webrtc.org/2994633002
Cr-Commit-Position: refs/heads/master@{#19286}
2017-08-09 13:13:45 +00:00
eb94436b38 Modify VP8 RTP to always use 2 bytes for picture Id
Bug: webrtc:7877
Change-Id: Ic40a7e142918399d05d02e8858313fe9b62d042b
Reviewed-on: https://chromium-review.googlesource.com/596967
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19282}
2017-08-09 11:17:48 +00:00
26622d3ff8 Audit of kConstants missing the const qualifier
Found via supersize query:
size_info.symbols.WhereFullNameMatches(r'\bk[A-Z]').WhereInSection('d')

This moves 90 symbols from .data -> .data.rel.ro (5.50kb)

BUG=chromium:747064

Review-Url: https://codereview.webrtc.org/2986163002
Cr-Commit-Position: refs/heads/master@{#19274}
2017-08-08 17:48:15 +00:00
186d9c3873 Renamed fields in common_types.h/RtcpStatistics.
BUG=webrtc:8033

Review-Url: https://codereview.webrtc.org/2992043002
Cr-Commit-Position: refs/heads/master@{#19247}
2017-08-04 12:03:53 +00:00
c0d481a4a6 Protected streams report RTP messages directly to the FlexFec streams
In preparation of making RTP packet demuxing many-to-one (one SSRC goes to one sink, but one sink may have multiple SSRCs), we need to remove FlexFEC's dependence on being able to register itself with the demuxer. Instead, we register FlexFEC streams with the streams they protect; when those streams get packets, they'll forward them to their associated FlexFEC streams, too.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2974453002
Cr-Commit-Position: refs/heads/master@{#19219}
2017-08-02 14:39:07 +00:00
a25a69582e Enable large-scale FEC tests on iOS.
Also change the loss rates to 5% and 1%, instead of 50%.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2950313002
Cr-Commit-Position: refs/heads/master@{#19199}
2017-08-01 12:01:07 +00:00
8a1d2a315f Remove NullReceiveStatistics
rtcp_sender accepts nullptr as indication statistics shouldn't be used,
Other uses of NullReceiveStatistcs were already deleted.

BUG=webrtc:8016

Review-Url: https://codereview.webrtc.org/2988143002
Cr-Commit-Position: refs/heads/master@{#19197}
2017-08-01 10:21:37 +00:00
f5f793c2ed Take smaller interface for RtpRtcp::Configuration::receive_statistics
BUG=webrtc:8016

Review-Url: https://codereview.webrtc.org/2988763002
Cr-Commit-Position: refs/heads/master@{#19167}
2017-07-27 11:44:18 +00:00
96b69bdbee Refactor composing report blocks for rtcp Sender/Receiver reports.
Compose them while creating sr/rr instead of presaving in temporary
member variable

BUG=webrtc:5565, webrtc:8016

Review-Url: https://codereview.webrtc.org/2979413002
Cr-Commit-Position: refs/heads/master@{#19138}
2017-07-25 16:15:14 +00:00
7fb11d7376 Shrink critical-section scope in ReceiveStatisticsImpl::GetActiveStatisticians()
The critical-section's scope can be shrunk (we can hold the lock for a shorter time).

BUG=None

Review-Url: https://codereview.webrtc.org/2984973002
Cr-Commit-Position: refs/heads/master@{#19137}
2017-07-25 15:25:23 +00:00
6209dcdeb1 Add SetReportBlocks to rtcp Sender/Receive Report classes.
BUG=None

Review-Url: https://codereview.webrtc.org/2991623002
Cr-Commit-Position: refs/heads/master@{#19136}
2017-07-25 15:07:13 +00:00
83377270dc Remove deprecated RtpRtcp::SetAudioPacketSize
was deprecated in https://codereview.webrtc.org/2545753002

BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2986793002
Cr-Commit-Position: refs/heads/master@{#19134}
2017-07-25 14:46:54 +00:00
d3f3c3497b Remove NullObjectReceiveStatistics() in rtp_rtcp module
use (already supported) nullptr as indication for no statistics

BUG=webrtc:8016

Review-Url: https://codereview.webrtc.org/2983363002
Cr-Commit-Position: refs/heads/master@{#19129}
2017-07-25 11:20:12 +00:00
a04d9c31a0 Remove RtpRtcp::RemoteRTCPStat(RTCPSenderInfo*) as unused
BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2986543002
Cr-Commit-Position: refs/heads/master@{#19128}
2017-07-25 11:03:39 +00:00
d14d9f7414 Use array declaration for extension URIs.
Allows using sizeof() on the class constants and reduces space usage by
a pointer.

Bug: None
Change-Id: Ie919b13094903d50bdadc92b23a5aa5b6cc100ec
Reviewed-on: https://chromium-review.googlesource.com/581878
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19113}
2017-07-21 19:36:14 +00:00
a3251dd83f Add parsing/serializing for MID RTP header extension.
This is the first in a series of CLs to add support for media
identification as part of unified plan SDP.

Bug: webrtc:4050
Change-Id: I0eb5639d240a9a1412c2b047a33d5112e4901f26
Reviewed-on: https://chromium-review.googlesource.com/576374
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19111}
2017-07-21 17:33:25 +00:00
f6a861ab6c Remove remains of webrtc/base
All downstream code have been updated to the new location.

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

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

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

Review-Url: https://codereview.webrtc.org/2976293002
Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 17:40:47 +00:00
c024740b5e Use relative paths in GN files.
BUG=webrtc:7952
TBR=kjellander@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2974863003
Cr-Commit-Position: refs/heads/master@{#18970}
2017-07-11 13:20:45 +00:00
370dd47973 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
Reason for revert:
Breaks lots of downstream projects.

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

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

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

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

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

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

Review-Url: https://codereview.webrtc.org/2973183002
Cr-Commit-Position: refs/heads/master@{#18948}
2017-07-10 11:50:54 +00:00
c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00
168794c43c Implement RTP keepalive in native stack.
BUG=webrtc:7907

Review-Url: https://codereview.webrtc.org/2960363002
Cr-Commit-Position: refs/heads/master@{#18912}
2017-07-06 11:38:06 +00:00
2edc6845ac Report timing frames info in GetStats.
Some frames are already marked as 'timing frames' via video-timing RTP header extension. Timestamps along full WebRTC pipeline are gathered for these frames. This CL implements reporting of these timestamps for a single
timing frame since the last GetStats(). The frame with the longest end-to-end delay between two consecutive GetStats calls is reported.

The purpose of this timing information is not to provide a realtime statistics but to provide debugging information as it will help identify problematic places in video pipeline for outliers (frames which took longest to process).

BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/2946413002
Cr-Commit-Position: refs/heads/master@{#18909}
2017-07-06 10:06:50 +00:00
7c7796b8ec Register FlexFEC SSRC to receive RTCP on sending side.
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2965883002
Cr-Commit-Position: refs/heads/master@{#18877}
2017-07-03 13:02:53 +00:00
dca1e09db7 Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"
This reverts commit c8fa692ec44fd6ba4fa3d085ac3161a262fc18c5.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2964773002 .
Cr-Commit-Position: refs/heads/master@{#18872}
2017-07-01 14:42:25 +00:00
c8fa692ec4 Update includes for webrtc/{base => rtc_base} rename (1/3)
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
2017-06-30 21:02:00 +00:00
fa8567868e Fix FecTest.FlexfecTest flakiness caused by seq. num. wraparound.
The CL in https://codereview.webrtc.org/2918333002/ enabled
FecTest.FlexfecTest and also added a sequence number offset between
the FEC packets and the media packets. This was to simulate that the
sequence numbers were generated from different spaces, i.e., that they
belong to different SSRCs.

The test does not account for sequence number wraparound, which means
that it could fail when the sequence number offset realization was large.
This CL fixes the problem by ensuring that the offset always lies in
[0, 2^15].

This CL also fixes spelling of UlpfecTest.

BUG=webrtc:7912
TESTED=ninja -C out/Debug && third_party/gtest-parallel/gtest-parallel --gtest_filter="*Flexfec*" -r 1000 out/Debug/modules_tests

Review-Url: https://codereview.webrtc.org/2966753002
Cr-Commit-Position: refs/heads/master@{#18863}
2017-06-30 14:22:15 +00:00
5f8b04d53a Higher logging severity for RED packets in UlpfecReceiverImpl.
As requested by holmer@ in https://codereview.webrtc.org/2918333002.

BUG=webrtc:5654
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2965533003
Cr-Commit-Position: refs/heads/master@{#18846}
2017-06-30 08:52:24 +00:00
d726a3f487 Reland of Only compare sequence numbers from the same SSRC in ForwardErrorCorrection. (patchset #1 id:1 of https://codereview.webrtc.org/2919313005/ )
Reason for revert:
Fix RtpStreamReceiver to not recover RTX packets with incorrect SSRC.

Original issue's description:
> Revert of Only compare sequence numbers from the same SSRC in ForwardErrorCorrection. (patchset #5 id:120001 of https://codereview.webrtc.org/2893293003/ )
>
> Reason for revert:
> Breaks fuzzer.
>
> Original issue's description:
> > Only compare sequence numbers from the same SSRC in ForwardErrorCorrection.
> >
> > Prior to this CL, the ForwardErrorCorrection state would be reset whenever
> > the difference in sequence numbers of the last recovered media packet
> > and the new packet (media or FEC) was too large. This comparison did not
> > take into account that FlexFEC uses a different SSRC for the FEC packets,
> > meaning that the the state would be reset very frequently when FlexFEC
> > is used. This should not have led to any major problems, except for a
> > decreased decoding efficiency.
> >
> > This CL verifies that whenever we compare sequence numbers in
> > ForwardErrorCorrection, they do indeed belong to the same SSRC.
> >
> > BUG=webrtc:5654
> >
> > Review-Url: https://codereview.webrtc.org/2893293003
> > Cr-Commit-Position: refs/heads/master@{#18399}
> > Committed: 1476a9d789
>
> TBR=stefan@webrtc.org,holmer@google.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5654
>
> Review-Url: https://codereview.webrtc.org/2919313005
> Cr-Commit-Position: refs/heads/master@{#18446}
> Committed: 92732ecc5c

R=stefan@webrtc.org
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2918333002
Cr-Commit-Position: refs/heads/master@{#18827}
2017-06-29 09:45:35 +00:00
e4350197ec Don't disable FEC if timing frames are disabled.
Don't disable fec for packets without timing frames extension
even if they are marked as belonging to timing frames.

BUG=webrtc:7894

Review-Url: https://codereview.webrtc.org/2956263002
Cr-Commit-Position: refs/heads/master@{#18826}
2017-06-29 09:27:48 +00:00