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Author SHA1 Message Date
sprang
0365a27f56 Use RtcpPacket to send SLI in RtcpSender 
BUG=webrtc:2450

Review URL: https://codereview.webrtc.org/1268383002

Cr-Commit-Position: refs/heads/master@{#9695}
 2015-08-11 08:02:44 +00:00
Peter Boström
d6f1a38165 Remove ViEChannel simulcast lock. 
Since the number of streams is now known on construction we can
initialize all RTP modules on construction. They are internally locked
so we don't nede a simulcast lock anymore.

BUG=1695
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52639004 .

Cr-Commit-Position: refs/heads/master@{#9577}
 2015-07-14 14:08:14 +00:00
Erik Språng
11beccd712 Remove external report blocks from RtcpSender and rtp_rtcp interface. 
Feature does not seem to be used and complicates other refactoring of
the rtcp module.

BUG=
R=asapersson@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54569004

Cr-Commit-Position: refs/heads/master@{#9304}
 2015-05-28 09:10:34 +00:00
Shao Changbin
e62202fedf Support handling multiple RTX but only generate SDP with RTX associated with VP8. 
This implementation registers RTX-APT map inside RTP sender and receiver.
While it only generates SDP with RTX associated with VP8 to make it
compatible with previous Chrome versions.

Should add following changes after reaches stable,
* Use RTX-APT map for building and restoring RTP packets.
* Add RTX support for RED or VP9 in Video engine.
* Set RTX payload type for RED inside FecConfig in EndToEndTest.

BUG=4024
R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36889004

Cr-Commit-Position: refs/heads/master@{#9040}
 2015-04-21 12:25:42 +00:00
sprang@webrtc.org
779c3d16b9 Use ByteReader/ByteWriter instead of rtputility and manual shift/add. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41289004

Cr-Commit-Position: refs/heads/master@{#8761}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8761 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 16:44:54 +00:00
kjellander@webrtc.org
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro 
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 13:04:54 +00:00
kwiberg@webrtc.org
00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-26 14:43:50 +00:00
pkasting@chromium.org
d324546ced Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ : 
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36179004

Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-23 21:29:45 +00:00
pbos@webrtc.org
7c4d20fd6c Remove potential deadlock in RTPSenderAudio. 
Removes lock-order inversion formed by RTPSenderAudio->RTPSender calls
by doing a lot shorter locking which fetches a current state of
RTPSenderAudio variables before sending.

Thread annotates locked variables and removes one lock in
RTPSenderAudio, bonus fixes data races reported in voe_auto_test
--automated under TSan (DTMF data race).

Also includes some bonus cleanup of RTPSenderVideo which removes the
send critsect completely as all methods using it was always called
from RTPSender under its send_critsect.

R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
BUG=3001, chromium:454654

Review URL: https://webrtc-codereview.appspot.com/41869004

Cr-Commit-Position: refs/heads/master@{#8348}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-12 12:20:50 +00:00
changbin.shao@intel.com
9691b36995 Cleanup for Rtp Rtcp API test. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8098 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-20 05:42:52 +00:00
pbos@webrtc.org
0b0c24177b Only return Rtx mode in RTXSendStatus(). 
There is no need to return 'ssrc' and 'payloadtype' inside this function
since they are never used.

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

Review URL: https://webrtc-codereview.appspot.com/38569004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-13 14:15:15 +00:00
pkasting@chromium.org
16825b1a82 Use int64_t more consistently for times, in particular for RTT values. 
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-12 21:51:21 +00:00
andrew@webrtc.org
8f27fcce79 Revert 8028 "Support associated payload type when registering Rt..." 
Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.

> Support associated payload type when registering Rtx payload type.
> 
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
> 
> BUG=4024
> R=pbos@webrtc.org, stefan@webrtc.org
> TBR=mflodman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/26259004
> 
> Patch from Changbin Shao <changbin.shao@intel.com>.

TBR=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 20:22:46 +00:00
pbos@webrtc.org
2a169640a3 Support associated payload type when registering Rtx payload type. 
Major changes include,
- Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
- Receiver: Restore RTP packets by the new RTX-APT map.
- Sender: Send RTP packets by checking RTX-APT map.
- Add RTX payload type for RED in the default codec list.

BUG=4024
R=pbos@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26259004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 15:16:10 +00:00
pbos@webrtc.org
d16e839c6d Rtp-Rtcp sender cleanup. 
Some setter functions from Rtp and Rtcp Sender never return negative values. Remove return results from those functions.

Also removed const on non-pointer/reference types for related files.

BUG=
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34469004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7962 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 13:49:55 +00:00
pbos@webrtc.org
9334ac2d78 Use vector of CSRCs for DeliverFrame & SetCSRCs. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28029004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 08:25:50 +00:00
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths. 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-20 22:28:14 +00:00
pbos@webrtc.org
ece3890d3a Report total bitrate for all streams in GetStats. 
This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.

R=stefan@webrtc.org, xians@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/27179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-14 11:52:04 +00:00
xians@webrtc.org
3cefbc99f4 Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. 
This also marks all virtual overrides of other classes in the same files. 

This will make a subsequent change I intend to do safer, where I'll change the 
argument types of the base Transport functions, by breaking the compile if I 
miss any overrides. 

This also highlighted a number of unused functions. I've removed some of these. 

TBR=mflodman@webrtc.org, pkasting@chromium.org
BUG=none 
TEST=none

Review URL: https://webrtc-codereview.appspot.com/28709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-10 09:42:53 +00:00
stefan@webrtc.org
4ef438e2de Remove the send-side cname getter APIs from voice and video engine. 
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.

R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 09:55:30 +00:00
pbos@webrtc.org
62bafae661 Some refactoring inside rtp_rtcp/. 
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.

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

Review URL: https://webrtc-codereview.appspot.com/19899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 12:10:51 +00:00
stefan@webrtc.org
ef92755780 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. 
This makes it easier to disable RTX by filtering out the RTX codec during call setup/signaling, and won't require that also the SSRCs are filtered out.

BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15629005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6335 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 08:25:29 +00:00
andresp@webrtc.org
dc80bae2a6 Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. 
Clean some logs and add asserts in the way.

BUG=3153
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-08 11:06:12 +00:00
sprang@webrtc.org
54ae4ffb9e Add callbacks for receive channel RTCP statistics. 
This allows a listener to receive new statistics as it is generated - avoiding the need to poll. This also makes handling stats from multiple RTP streams more tractable.
The change is primarily targeted at the new video engine API.

TEST=Unit test in ReceiveStatisticsTest. Integration tests to follow as call tests when fully wired up.

BUG=2235
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5323 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-19 13:26:02 +00:00
stefan@webrtc.org
7e9315b42e Adds support for sending redundant payloads over RTX. 
TEST=trybots
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 10:24:26 +00:00
stefan@webrtc.org
48df38114d Fix for making sure that the packet in order checks are done prior to updating the last received packet state. 
Without this fix all packets are considered out-of-order by the rtp receiver, causing the last received state
in the rtp receiver to never get valid.

Also makes sure that only valid timestamps and receive times are used for audio/video sync.

BUG=2608
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 15:18:52 +00:00
pbos@webrtc.org
be63fd644f Initialize CodecInst structs in test_api_audio.cc. 
Fixes errors detected by DrMemory on Windows.

BUG=2382
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2228004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4764 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-17 09:33:17 +00:00
stefan@webrtc.org
7bb8f02274 Adds support for combining RTX and FEC/RED. 
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX.

Enables retransmissions over RTX by default in the loopback test.

BUG=1811
TESTS=voe/vie_auto_test --automated and trybots.
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2154004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-06 13:40:11 +00:00
stefan@webrtc.org
286fe0b04d Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" 
...and fixes the RTCP bug.

BUG=2277
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4588 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-21 20:58:21 +00:00
henrike@webrtc.org
a0218a84d1 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." 
> Reverts a second set of reverts caused by a bug in a dependency.
> 
> Revert "Revert r4328"
> 
> Revert "Revert r4322 "Support sending multiple report blocks and keeping track
> of statistics on"
> 
> BUG=1811
> R=henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/2072004

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2087004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4585 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-21 19:44:13 +00:00
stefan@webrtc.org
1a65d6c36b Reverts a second set of reverts caused by a bug in a dependency. 
Revert "Revert r4328"

Revert "Revert r4322 "Support sending multiple report blocks and keeping track
of statistics on"

BUG=1811
R=henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2072004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4582 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-21 16:22:21 +00:00
wu@webrtc.org
822fbd8b68 Update talk to 50918584. 
Together with Stefan's http://review.webrtc.org/1960004/.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2048004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-15 23:38:54 +00:00
tnakamura@webrtc.org
64e2cbf184 clean up incomplete revert in r4357 
Also revert r4319, will follow up with pbos

Reason for recent series of reverts: video freezes when testing with packet loss

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1817004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 21:52:59 +00:00
tnakamura@webrtc.org
aa4d96a134 Revert r4301 
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 19:25:04 +00:00
elham@webrtc.org
b7eda43810 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on 
several SSRCs"

R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1774006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 21:08:27 +00:00
pbos@webrtc.org
96edd56170 Sorted headers under rtp_rtcp/. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1781005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 15:40:42 +00:00
stefan@webrtc.org
717d147ebb Support sending multiple report blocks and keeping track of statistics on several SSRCs. 
BUG=1811
TEST=vie_auto_test --automated, voe_auto_test --automated, trybots
R=andresp@webrtc.org, tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1768004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 13:39:27 +00:00
stefan@webrtc.org
66b2e5c05a Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the 
rtp_rtcp implementation.

This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.

With this change the dead-or-alive and packet timeout APIs are removed.

TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1745004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-05 14:30:48 +00:00
solenberg@webrtc.org
91811e2b04 Remove unused multi stream bandwidth estimator. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1712004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4264 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-25 20:36:14 +00:00
pbos@webrtc.org
a048d7cb0a Include files from webrtc/.. paths in rtp_rtcp/ 
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1557004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 14:27:38 +00:00
stefan@webrtc.org
a5cb98cbbd Breaking out RTP header parsing from the RTP module. 
This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video.

Moving bandwidth estimation before the RTP module is also required for RTX.

TEST=vie_auto_test, voe_auto_test, trybots.
BUG=1811
R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1545004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 12:12:51 +00:00
mikhal@webrtc.org
6cfa3907c8 Updating NACK RTX test 
BUG=1513
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1274006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-15 20:17:43 +00:00
pbos@webrtc.org
3004c79c6a Fix clang errors in non-GYP_DEFINES=clang=1 build 
BUG=1623
R=stefan@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1368004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 12:36:21 +00:00
mflodman@webrtc.org
9f5ebb5251 Adding a payload type for RTX. 
BUG=736
TEST=Modified RTP unittests.

Review URL: https://webrtc-codereview.appspot.com/1278004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 14:55:46 +00:00
pbos@webrtc.org
2f44673d66 WebRtc_Word32 => int32_t for rtp_rtcp/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1279007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 11:08:41 +00:00
stefan@webrtc.org
41211466d8 Revert the deletion of test_api_nack.cc in r3674. 
TBR=mflodman@webrtc.org, mikhal@webrtc.org

BUG=1513

Review URL: https://webrtc-codereview.appspot.com/1217004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3677 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 15:00:50 +00:00
mikhal@webrtc.org
bda7f305c5 Adding RTX on source 
Review URL: https://webrtc-codereview.appspot.com/1190004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-15 23:21:52 +00:00
turaj@webrtc.org
b7edd06530 Remove DTMF detection. Talk team has been in the loop and there is no need for 
DTMF detection at the receiver side.

test=voe_auto_test, VoE extended test DTMF
Review URL: https://webrtc-codereview.appspot.com/1168004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 22:27:27 +00:00
stefan@webrtc.org
1dc0aa2de2 Fix for build error on android introduced with r3609. 
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1164004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3611 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 09:30:47 +00:00
stefan@webrtc.org
a27107004d Split the NACK list into multiple RTCPs if it's too big. 
TEST=rtp_rtcp_unittests
BUG=1434

Review URL: https://webrtc-codereview.appspot.com/1148006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3609 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 09:02:06 +00:00