admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
7,499 Commits 1 Branch 0 Tags
Commit Graph

296 Commits

Author SHA1 Message Date
asapersson@webrtc.org
4414939954 Add method for incrementing RtpPacketCounter. Removes duplicate code. 
Correction to check if rtx is enabled on send-side (and not receive) when updating rtx send bitrate stat.

Remove unneeded guarded by annotations.

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

Review URL: https://webrtc-codereview.appspot.com/41729004

Cr-Commit-Position: refs/heads/master@{#8239}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-04 08:35:21 +00:00
pkasting@chromium.org
0e81fdf5d2 Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting. 
BUG=chromium:82439
TEST=none
R=henrik.lundin@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40569004

Cr-Commit-Position: refs/heads/master@{#8229}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-02 23:54:40 +00:00
sprang@webrtc.org
c957ffc6dc Fixed potential crash if rtp packet history is completely full. 
Also performance enhanecement in rtp_sender (don't lookup if kDontStore)

BUG=4171
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39759004

Cr-Commit-Position: refs/heads/master@{#8226}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-02 13:08:14 +00:00
pkasting@chromium.org
026b892e72 Using << on an int8_t or uint8_t will output a character rather than a number. 
Places that do this need to cast to int to get the desired behavior.

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

Review URL: https://webrtc-codereview.appspot.com/40579004

Cr-Commit-Position: refs/heads/master@{#8223}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8223 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-30 19:54:19 +00:00
tommi@webrtc.org
4161715e3f Remove ChangeUniqueID. 
This fixes a two year old TODO of deleting dead code :)
In cases where the _id or id_ member variable is being used for tracing,
I changed the member to at least be const.

It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them.

BUG=
R=henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37849004

Cr-Commit-Position: refs/heads/master@{#8201}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-29 12:14:13 +00:00
sprang@webrtc.org
43c883954f Allow rtp packet history to dynamically expand in size. 
When using the paced sender, packets will be put into the rtp packet
history and then retreived from there again when it is time to send.

In some cases (low send bitrate and very large frames created) this
may overflow, causing packets to be overwritten in the packet history
before they have been sent.

Check this condition and expand history size if needed.

This is primarily triggered during screenshare, when
switching to a large picture with lots of high frequency
details in it.

BUG=4171
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34879004

Cr-Commit-Position: refs/heads/master@{#8195}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8195 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-29 09:09:41 +00:00
asapersson@webrtc.org
37c0559c1e Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets). 
Don't copy codec specific header for empty packets in the jitter buffer.

BUG=3135
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37659004

Cr-Commit-Position: refs/heads/master@{#8184}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 13:58:40 +00:00
sprang@webrtc.org
2a6558c2a5 Make sure ByteReader<T>::Read* is properly constified. 
Also, start using it in real code...

BUG=
R=holmer@google.com, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37809004

Cr-Commit-Position: refs/heads/master@{#8181}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 12:38:16 +00:00
asapersson@webrtc.org
273fbbb921 Update StreamDataCounter with FEC bytes. 
Add histograms stats for send/receive FEC bitrate:
- "WebRTC.Video.FecBitrateReceivedInKbps"
- "WebRTC.Video.FecBitrateSentInKbps"

Correct media payload bytes in StreamDataCounter to not include FEC bytes.

Fix stats for rtcp packets sent/received per minute (regression from r7910).

BUG=crbug/419657
R=holmer@google.com, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 12:17:29 +00:00
asapersson@webrtc.org
cfd82dfc11 Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. 
Prepares for adding FEC bytes to the StreamDataCounter.

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

Review URL: https://webrtc-codereview.appspot.com/37579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 09:39:59 +00:00
asapersson@webrtc.org
df7b65ba01 Change CreateOrGetReportBlockInformation to have one return path. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8113 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-21 13:07:04 +00:00
asapersson@webrtc.org
9ffd8fe96b Indentation changes. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-21 08:22:50 +00:00
asapersson@webrtc.org
0800db74b9 Add percentage of fec packets and recovered media packets to histogram stats: 
- "WebRTC.Video.ReceivedFecPacketsInPercent"
- "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec"

BUG=crbug/419657
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 07:40:20 +00:00
kwiberg@webrtc.org
2ebfac5649 Remove COMPILE_ASSERT and use static_assert everywhere 
COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.

R=aluebs@webrtc.org, andrew@webrtc.org, hellner@chromium.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 10:51:54 +00:00
andresp@webrtc.org
86e1e487e7 Move system_wrappers.gyp files to the proper directory. 
Build targets should not refer to non-subpaths as was happening before when
 source/system_wrappers.gyp refers to ../interface/ files.

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

Review URL: https://webrtc-codereview.appspot.com/37609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 09:30:52 +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
kwiberg@webrtc.org
3df38b442f Unify the two copies of compile_assert.h 
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.

R=aluebs@webrtc.org, andrew@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-13 11:37:48 +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
stefan@webrtc.org
11d8176cb3 Move updating nack bitrate inside UpdateNACKBitRate. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7960 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 09:52:24 +00:00
asapersson@webrtc.org
cb79141eab Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc. 
When using rtx, receiver reports with two report blocks are received. The report blocks have the same remote ssrc and therefore the first report block was overwritten by the second report block when stored in the ReportBlockInfoMap.

Removed unused function ResetRTT.

BUG=4114
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33659005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7952 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 14:30:32 +00:00
pbos@webrtc.org
ce4e9a3562 Refactor some receive-side stats. 
Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.

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

Review URL: https://webrtc-codereview.appspot.com/28259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 13:50:16 +00:00
asapersson@webrtc.org
d08d389ce8 Add field to counters for when first rtp/rtcp packet is sent/received. 
Use this time for histogram statistics (send/receive bitrates, sent/received rtcp fir/nack packets/min).

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7910 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 12:03:11 +00:00
pkasting@chromium.org
0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. 
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 22:09:40 +00:00
asapersson@webrtc.org
97d0489058 Add video send bitrates to histogram stats: 
- total bitrate ("WebRTC.Video.BitrateSentInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateSentInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateSentInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateSentInKbps")
- retransmitted bitrate ("WebRTC.Video.RetransmittedBitrateInKbps")

Add retransmitted bytes to StreamDataCounters.

Change in UpdateRtpStats to also update counters for retransmitted packet.

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7838 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 09:47:53 +00:00
asapersson@webrtc.org
ba8138ba38 Change type of nack_last_time_sent_full_ from uint32_t to int64_t. 
Could cause nack requests to be sent too frequently.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7825 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 13:29:02 +00:00
andresp@webrtc.org
7f722492f1 Set simulcastIdx field to zero even if it has no meaning. 
Helps to be able to memcmp between 2 parses of the same packet.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7773 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 15:29:29 +00:00
asapersson@webrtc.org
d952c40c7e Add receive bitrates to histogram stats: 
- total bitrate ("WebRTC.Video.BitrateReceivedInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateReceivedInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateReceivedInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateReceivedInKbps")

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27189005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7756 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-27 07:38:56 +00:00
henrik.lundin@webrtc.org
aff1751c96 Add new test for VP8 packetizer to test tight partitions 
It was discovered that if remaining_bytes is an exact multiple of
max_payload_len in RtpPacketizerVp8::CalcNextSize, then the packetizer
will produce too many packets (i.e., split the payload into more
packets than needed).

This CL adds a test to trigger the problem.

BUG=4019
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 12:36:58 +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
pbos@webrtc.org
49ff40e32e Make SetREMBData accept vector of SSRCs. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7697 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-13 14:42:37 +00:00
pbos@webrtc.org
d42a3adf42 Remove partially defined WebRtcRTPHeader from Parse(). 
It' bit ugly that RtpDepacketizer::ParsedPayload partially defines WebRtcRTPHeader, and then sent to Parse() function for internal change.
To make it clearer, the CL gets rid of using partially-defined WebRtcRTPHeader.

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

Review URL: https://webrtc-codereview.appspot.com/28919004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 11:02:12 +00:00
stefan@webrtc.org
0bae1fab4a Wire up bandwidth stats to the new API and webrtcvideoengine2. 
Adds stats to verify bandwidth and pacer stats.

BUG=1788
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-05 14:05:29 +00:00
sprang@webrtc.org
dcebf2daa7 Reworked paced sender queue 
Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage.

Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these.

Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long.

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

Review URL: https://webrtc-codereview.appspot.com/27869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 16:27:16 +00:00
asapersson@webrtc.org
2dd3134e50 Add stats for duplicate sent and received NACK requests. 
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7559 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 12:42:30 +00:00
stefan@webrtc.org
76960d5f74 For FIR packet, payload length is zero, so SendToNetwork function is failing. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7490 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 09:47:14 +00:00
stefan@webrtc.org
eb24b04f16 Add periodic logging of received RTP headers and estimated clock offsets for e2e delay. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7444 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 11:40:13 +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
minyue@webrtc.org
2c0cdbce22 Estimating NTP time with a given RTT. 
RemoteNtpTimeEstimator needed user to give a remote SSRC and it intended to call RtpRtcp module to obtain RTT, to be able to calculate Ntp time.

When RTT cannot be directly obtained from the RtpRtcp module with the specified SSRC, RemoteNtpTimeEstimator would fail.

This change allows RemoteNtpTimeEstimator to calculate NTP with an external RTT estimate.

An immediate benefit is that capture_start_ntp_time_ms_ can be obtained in a Google hangout call.

BUG=

TEST=chromium + hangout call
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 10:52:43 +00:00
pbos@webrtc.org
730d270771 Remove callback from RtpDepacketizer::Parse(). 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30489004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 08:00:22 +00:00
pbos@webrtc.org
315669939a Fix typo from RtpPacketizerH264. 
BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27609004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7295 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-25 07:31:42 +00:00
pbos@webrtc.org
38344ed280 Move thread_annotations.h to webrtc/base/. 
R=andresp@webrtc.org, mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 06:05:00 +00:00
stefan@webrtc.org
5a098c51ea Refactor VP8 de-packetizer. 
It's duplicated to parse VP8 RTP packet at the moment. We firstly call
RTPPayloadParser functions to save parsed information in RTPPayload
structure, then copy them to RTP header.

This CL removes RTPPayloadParser class and directly saves parsed data in
RTP header.

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

Review URL: https://webrtc-codereview.appspot.com/27449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 11:58:20 +00:00
pbos@webrtc.org
1fb5d1204b Initialize restored_packet in nack_rtx_unittest.cc. 
This is to get the DrMemory Full bots to go green, this was previously
suppressed. This fix is likely hiding a real bug that should be
investigated, but it's not a regression from before. The issue should
not be closed before we figure out why this is the case and revert this
"fix".

TBR=stefan@webrtc.org
BUG=3183

Review URL: https://webrtc-codereview.appspot.com/30369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7169 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 16:16:00 +00:00
pbos@webrtc.org
b5e6bfc76a Make RTPSender/RTPReceiver generic. 
Changes include,
1) Introduce class RtpPacketizerGeneric & RtpDePacketizerGeneric.
2) Introduce class RtpDepacketizerVp8.
3) Make RTPSenderVideo::SendH264 generic and used by all packetizers.
4) Move codec specific functions from RTPSenderVideo/RTPReceiverVideo to
RtpPacketizer/RtpDePacketizer sub-classes.

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

Review URL: https://webrtc-codereview.appspot.com/26399004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7163 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 11:05:55 +00:00
stefan@webrtc.org
6071b0636d Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such. 
This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.

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

-- This is was reviewed in https://webrtc-codereview.appspot.com/19309004/, but
-- a new cl was needed to resolve a small conflict before committing.

BUG=none
TEST=none
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 07:42:33 +00:00