757 Commits

Author SHA1 Message Date
Stefan Holmer
4654d204e4 Add test which verifies that the RTP header extensions are set correctly for FEC packets.
Also taking the opportunity to do a little bit of clean up.

BUG=webrtc:705
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1506863002 .

Cr-Commit-Position: refs/heads/master@{#10927}
2015-12-08 08:10:58 +00:00
Danil Chapovalov
fc47ed6c05 rtcp::Rrtr block moved into own file and got Parse function
BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1496883002 .

Cr-Commit-Position: refs/heads/master@{#10912}
2015-12-07 13:46:42 +00:00
Stefan Holmer
b86d4e4a8d Prepare the AudioSendStream to be hooked up to send-side BWE.
This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.

BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1479023002 .

Cr-Commit-Position: refs/heads/master@{#10909}
2015-12-07 09:26:32 +00:00
Danil Chapovalov
97f7e13c23 rtcp::ReceiverReport moved into own file and got Parse function
BUG=webrtc:5260
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1453083002 .

Cr-Commit-Position: refs/heads/master@{#10897}
2015-12-04 15:13:40 +00:00
Erik Språng
f7c5776d42 Refactorings to send RTCP packets directly via the RtcpPacket callback, with some simplifications enabled by this. NACK now also sent via RtcpPacket.
BUG=webrtc:2450
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1309833002 .

Cr-Commit-Position: refs/heads/master@{#10888}
2015-12-04 09:40:54 +00:00
danilchap
f8385aded0 rtcp::Pli moved into own file and got a Parse function
Created rtcp::Psfb abstract class between rtcp::Pli and rtcp::RtcpPacket to hold common data for Feedback Message.

BUG=webrtc:5260

Review URL: https://codereview.webrtc.org/1446513002

Cr-Commit-Position: refs/heads/master@{#10823}
2015-11-27 13:36:17 +00:00
stefan
64c0a0a111 Revert of Make overuse estimator one dimensional. (patchset #5 id:80001 of https://codereview.webrtc.org/1376423002/ )
Reason for revert:
Broke webrtc_perf_tests on bots.

Original issue's description:
> Make overuse estimator one dimensional.
>
> This drops the payload size difference dimension of the Kalman filter,
> which doesn't improve the quality of the estimation when pacing packets
> on the send-side.
>
> R=gaetano.carlucci@gmail.com, mflodman@webrtc.org, terelius@webrtc.org
>
> Committed: https://crrev.com/06e05a85b9e4def75ed5e6b582c4df842616f25f
> Cr-Commit-Position: refs/heads/master@{#10809}

TBR=terelius@webrtc.org,mflodman@webrtc.org,gaetano.carlucci@gmail.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1481003002

Cr-Commit-Position: refs/heads/master@{#10816}
2015-11-27 09:02:35 +00:00
Stefan Holmer
06e05a85b9 Make overuse estimator one dimensional.
This drops the payload size difference dimension of the Kalman filter,
which doesn't improve the quality of the estimation when pacing packets
on the send-side.

R=gaetano.carlucci@gmail.com, mflodman@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/1376423002 .

Cr-Commit-Position: refs/heads/master@{#10809}
2015-11-26 14:35:10 +00:00
danilchap
50c5136cb2 RTCP Bye packet moved to own file
Bye class got support for Parsing
 Reason field implemented

Review URL: https://codereview.webrtc.org/1430013003

Cr-Commit-Position: refs/heads/master@{#10741}
2015-11-22 17:03:16 +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
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
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
danilchap
df948f03b3 rtcp::ReportBlock refactored to contain parsing
Review URL: https://codereview.webrtc.org/1420283022

Cr-Commit-Position: refs/heads/master@{#10633}
2015-11-13 11:03:18 +00:00
tfarina
5237aaf243 Convert usage of ARRAY_SIZE to arraysize.
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.

BUG=None
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1405023016

Cr-Commit-Position: refs/heads/master@{#10594}
2015-11-11 07:44:39 +00:00
philipel
cfc319be1d Reland of Work on flexible mode and screen sharing. (patchset #1 id:1 of https://codereview.webrtc.org/1438543002/ )
Reason for revert:
Failed test not related to this CL (test fails on
master at an earlier date), re-landing original CL..

(This time from my @webrtc account.)

Original issue's description:
> Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ )
>
> Reason for revert:
> Seems to break VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Linux Memcheck buildbot.
>
> Original issue's description:
> > Work on flexible mode and screen sharing.
> >
> > Implement VP8 style screensharing but with spatial layers.
> > Implement flexible mode.
> >
> > Files from other patches:
> > generic_encoder.cc
> > layer_filtering_transport.cc
> >
> > BUG=webrtc:4914
> >
> > Committed: https://crrev.com/77ccfb4d16c148e61a316746bb5d9705e8b39f4a
> > Cr-Commit-Position: refs/heads/master@{#10572}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,philipel@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4914
>
> Committed: https://crrev.com/0be8f1d347bdb171462df89c2a4c69b3f3eb7519
> Cr-Commit-Position: refs/heads/master@{#10578}

TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,terelius@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4914

Review URL: https://codereview.webrtc.org/1431283002

Cr-Commit-Position: refs/heads/master@{#10581}
2015-11-10 15:17:26 +00:00
terelius
0be8f1d347 Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ )
Reason for revert:
Seems to break VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Linux Memcheck buildbot.

Original issue's description:
> Work on flexible mode and screen sharing.
>
> Implement VP8 style screensharing but with spatial layers.
> Implement flexible mode.
>
> Files from other patches:
> generic_encoder.cc
> layer_filtering_transport.cc
>
> BUG=webrtc:4914
>
> Committed: https://crrev.com/77ccfb4d16c148e61a316746bb5d9705e8b39f4a
> Cr-Commit-Position: refs/heads/master@{#10572}

TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,philipel@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4914

Review URL: https://codereview.webrtc.org/1438543002

Cr-Commit-Position: refs/heads/master@{#10578}
2015-11-10 13:31:22 +00:00
philipel
77ccfb4d16 Work on flexible mode and screen sharing.
Implement VP8 style screensharing but with spatial layers.
Implement flexible mode.

Files from other patches:
generic_encoder.cc
layer_filtering_transport.cc

BUG=webrtc:4914

Review URL: https://codereview.webrtc.org/1328113004

Cr-Commit-Position: refs/heads/master@{#10572}
2015-11-10 10:19:20 +00:00
mflodman
c4a1c370aa Removed vie_defines.h
The defines still in use was only used in single files, so they were
moved to these specific cc-files.

Review URL: https://codereview.webrtc.org/1411573007

Cr-Commit-Position: refs/heads/master@{#10539}
2015-11-06 12:33:56 +00:00
asapersson
c253a1c00e Reland of "Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile."
BUG=webrtc:5144
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1409753007

Cr-Commit-Position: refs/heads/master@{#10533}
2015-11-06 08:12:09 +00:00
asapersson
394c537b21 Update layer indices for non-flexible mode according to updates in the RTP payload profile.
https://tools.ietf.org/id/draft-ietf-payload-vp9-01.txt

BUG=chromium:500602
TBR=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1426813002

Cr-Commit-Position: refs/heads/master@{#10522}
2015-11-05 14:07:09 +00:00
asapersson
5d4e944391 Revert of Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile. (patchset #3 id:40001 of https://codereview.webrtc.org/1427253002/ )
Reason for revert:
Breaks bot.

Original issue's description:
> Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile. Max p_diff is 8 bits.
>
> Change type of number of reference pictures (size_t -> uint8_t). Max is 2 bits.
>
> Size of WebRtcRTPHeader: 4352 -> 1784 bytes.
>
> BUG=webrtc:5144, chromium:500602
>
> Committed: https://crrev.com/81c5c7f8157f767747bd97419eb0a589207354cf
> Cr-Commit-Position: refs/heads/master@{#10504}

TBR=stefan@webrtc.org,mflodman@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5144, chromium:500602

Review URL: https://codereview.webrtc.org/1423493005

Cr-Commit-Position: refs/heads/master@{#10508}
2015-11-04 15:40:44 +00:00
asapersson
81c5c7f815 Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile. Max p_diff is 8 bits.
Change type of number of reference pictures (size_t -> uint8_t). Max is 2 bits.

Size of WebRtcRTPHeader: 4352 -> 1784 bytes.

BUG=webrtc:5144, chromium:500602

Review URL: https://codereview.webrtc.org/1427253002

Cr-Commit-Position: refs/heads/master@{#10504}
2015-11-04 10:19:45 +00:00
Henrik Kjellander
ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=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 -m tryserver.webrtc

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

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00
tfarina
a41ab9326c Switch usage of _DEBUG macro to NDEBUG.
http://stackoverflow.com/a/29253284/5237416

BUG=None
R=tommi@webrtc.org
NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1429513004

Cr-Commit-Position: refs/heads/master@{#10468}
2015-10-30 23:08:54 +00:00
asapersson
6449990387 Update scalability structure data according to updates in the RTP payload profile.
https://tools.ietf.org/id/draft-ietf-payload-vp9-01.txt

BUG=chromium:500602
TBR=mflodman

Review URL: https://codereview.webrtc.org/1411923004

Cr-Commit-Position: refs/heads/master@{#10445}
2015-10-29 10:35:16 +00:00
asapersson
32df5efc6d Update reference indices according to updates in the RTP payload profile.
https://tools.ietf.org/id/draft-ietf-payload-vp9-01.txt

BUG=chromium:500602

Review URL: https://codereview.webrtc.org/1406283008

Cr-Commit-Position: refs/heads/master@{#10442}
2015-10-29 08:45:47 +00:00
Henrik Kjellander
98f53510b2 system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1413333002 .

Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
Peter Boström
ebc0b4e993 Use webrtc/base/logging.h for rtp_rtcp.
BUG=webrtc:5118
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1422023002 .

Cr-Commit-Position: refs/heads/master@{#10437}
2015-10-28 15:39:43 +00:00
stefan
f116bd0d7a Call OnSentPacket for all packets sent in the test framework.
Required a bit of refactoring to make it possible to pass a Call to DirectTransport on construction. This also lead to me having to remove the shared lock between PacketTransport and RtpRtcpObserver. Now RtpRtcpObserver has a SetTransports method instead of a SetReceivers method.

BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1419193002

Cr-Commit-Position: refs/heads/master@{#10430}
2015-10-27 15:29:47 +00:00
pbos
a99069db63 Fix win32 header include order in rtp_utility.h.
Matches the include order in webrtc/base/criticalsection.h and makes use
of winsock2.h instead of winsock.h for consistency.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1407053008

Cr-Commit-Position: refs/heads/master@{#10389}
2015-10-23 13:32:44 +00:00
stefan
bbe876f0d3 Set send times in send time history via OnSentPacket.
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1419503004

Cr-Commit-Position: refs/heads/master@{#10384}
2015-10-23 09:05:43 +00:00
sprang
affa39cb39 Remove time constraint on first retransmit of a packet.
We don't allow more than one retransmission within one RTT, but the RTT
estimate might be off. Reasonably, the remote end will not send a NACK
until the packet after has been received - so always resend on first
request.

Review URL: https://codereview.webrtc.org/1414563003

Cr-Commit-Position: refs/heads/master@{#10362}
2015-10-21 20:46:42 +00:00
Stefan Holmer
0a87ffcaad Fix bug in how send timestamps are converted to 24 bits.
BUG=webrtc:4173
R=sprang@webrtc.org

Review URL: https://codereview.webrtc.org/1412683004 .

Cr-Commit-Position: refs/heads/master@{#10356}
2015-10-21 11:42:09 +00:00
tommi
e4f96501fc Remove system_wrappers/interface/trace_event.h
BUG=

Review URL: https://codereview.webrtc.org/1417773002

Cr-Commit-Position: refs/heads/master@{#10346}
2015-10-21 06:00:57 +00:00
pbos
22993e1a0c Unify FrameType and VideoFrameType.
Prevents some heap allocation and frame-type conversion since interfaces
mismatch. Also it's less confusing to have one type for this.

BUG=webrtc:5042
R=magjed@webrtc.org, mflodman@webrtc.org, henrik.lundin@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1371043003

Cr-Commit-Position: refs/heads/master@{#10320}
2015-10-19 09:39:15 +00:00
noahric
3eab10d629 Add back an override of RestoreOriginalPacket.
External consumers may have a dependency on the old name, so this will give them the opportunity to switch over.

BUG=

Review URL: https://codereview.webrtc.org/1414543002

Cr-Commit-Position: refs/heads/master@{#10310}
2015-10-17 08:06:46 +00:00
sprang
861c55e583 Transport sequence number should be set also for retransmissions.
This is a reland of https://codereview.webrtc.org/1385563005 which was
reverted since the test was flaky. The reason was a race condition (in
the test) and that NACK wasn't properly set up.

BUG=

Review URL: https://codereview.webrtc.org/1406193002

Cr-Commit-Position: refs/heads/master@{#10303}
2015-10-16 17:01:25 +00:00
stefan
c1aeaf0dc3 Wire up packet_id / send time callbacks to webrtc via libjingle.
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1363573002

Cr-Commit-Position: refs/heads/master@{#10289}
2015-10-15 14:26:17 +00:00
noahric
65220a70a3 Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
Also updated the RTPPayloadRegistry::RestoreOriginalPacket signature to not take the first arg as a **, since it isn't modified.

Review URL: https://codereview.webrtc.org/1394573004

Cr-Commit-Position: refs/heads/master@{#10276}
2015-10-14 18:29:56 +00:00
sprang
7dc39f331a Avoid data race in RtcpReceiver.
See eg https://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/3930/steps/video_engine_tests/logs/stdio

Also some cleanup, lock annotations.

BUG=

Review URL: https://codereview.webrtc.org/1401463003

Cr-Commit-Position: refs/heads/master@{#10266}
2015-10-13 16:17:56 +00:00
Peter Boström
e23e737177 Disable pacer disabling.
Since the pacer is always enabled, removing enable/disable which makes
all packet queueing succeed. Also renaming one of the ::SendPackets
::InsertPacket to avoid confusion.

BUG=webrtc:1695, webrtc:2629
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1392513002 .

Cr-Commit-Position: refs/heads/master@{#10211}
2015-10-08 09:44:29 +00:00
Peter Boström
0c4e06b4c6 Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.

BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1362503003 .

Cr-Commit-Position: refs/heads/master@{#10196}
2015-10-07 10:23:32 +00:00
Alejandro Luebs
10950692d6 Revert "Transport sequence number should be set also for retransmissions."
After this CL, video_engine_test started failing flakily in different bots for different CLs.

TBR=sprang@webrtc.org

Review URL: https://codereview.webrtc.org/1393553003 .

Cr-Commit-Position: refs/heads/master@{#10188}
2015-10-06 19:27:12 +00:00
sprang
af4ced986b Transport sequence number should be set also for retransmissions.
When fetching a packet from the rtp packet history, cuased by a
retransmission, the transport seq extension header is enabled but the
sequence number is set to 0. A new transport seq should be assigned in
this case.

BUG=

Review URL: https://codereview.webrtc.org/1385563005

Cr-Commit-Position: refs/heads/master@{#10183}
2015-10-06 13:02:57 +00:00
stefan
1d8a506405 Add a PacketOptions struct to webrtc::Transport.
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.

BUG=4173

Review URL: https://codereview.webrtc.org/1376673004

Cr-Commit-Position: refs/heads/master@{#10144}
2015-10-02 10:39:40 +00:00
pbos
da903eaabb Unify newapi::RtcpMode and RTCPMethod.
BUG=webrtc:1695
R=solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1373903003

Cr-Commit-Position: refs/heads/master@{#10143}
2015-10-02 09:37:18 +00:00
sprang
49f9cdba02 Fix bug where rtcp::TransportFeedback may generate incorrect messages.
In particular, if 14 short deltas were inserted (2 * capacity of status
vector chunk with 2bit items) followed by a large delta, that status
item would be dropped.

BUG=

Review URL: https://codereview.webrtc.org/1367193002

Cr-Commit-Position: refs/heads/master@{#10132}
2015-10-01 10:07:04 +00:00
sprang
38778b046f Add unit test for nack bandwidth constraint.
BUG=

Review URL: https://codereview.webrtc.org/1341743002

Cr-Commit-Position: refs/heads/master@{#10111}
2015-09-29 16:48:30 +00:00
sprang
86fd9ed6f9 Set RtcpSender transport at construction.
BUG=

Review URL: https://codereview.webrtc.org/1365043002

Cr-Commit-Position: refs/heads/master@{#10106}
2015-09-29 11:45:51 +00:00
pbos
2d566686a2 Unify Transport and newapi::Transport interfaces.
BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1369263002

Cr-Commit-Position: refs/heads/master@{#10096}
2015-09-28 16:59:36 +00:00