822fbd8b68
Update talk to 50918584.
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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
aa4d96a134
Revert r4301
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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
a4407329d4
Include files from webrtc/.. paths in video_coding/.
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BUG=1662
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1783006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4348 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 12:32:05 +00:00
66b2e5c05a
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
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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
a5cb98cbbd
Breaking out RTP header parsing from the RTP module.
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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
77f6b2175e
Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..."
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> Revert 3933 "Remove traces of deprecated WebRtc_Word types."
>
> > Remove traces of deprecated WebRtc_Word types.
> >
> > BUG=314
> > R=tommi@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/1385004
>
> TBR=pbos@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1386004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1397004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3948 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 12:02:11 +00:00
68e5a68f07
Revert 3933 "Remove traces of deprecated WebRtc_Word types."
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> Remove traces of deprecated WebRtc_Word types.
>
> BUG=314
> R=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1385004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1386004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3934 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 09:30:12 +00:00
265a5d298a
Remove traces of deprecated WebRtc_Word types.
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BUG=314
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1385004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3933 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 09:11:20 +00:00
56b5f77a2b
Add support for multiple streams to RtpPlayer:
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- Tests video_rtp_play.cc, video_rtp_play_mt.cc, decode_from_storage.cc rewritten
- rtp_player.cc/.h rewritten; added interfaces for externally setting up sinks
- Support for reading .rtp files pulled out into rtp_file_reader namespace
- Added support for reading .pcap (libpcap/wireshark/tcpdump) files, see pcap_file_reader
BUG=
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1201009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3856 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-16 10:31:56 +00:00
7b859cc1e9
Webrtc_Word32 => int32_t in video_coding/main/
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BUG=
Review URL: https://webrtc-codereview.appspot.com/1279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3753 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-02 15:54:38 +00:00
2baf5f5fa0
Refactor webrtc specific Event implementation to an EventFactory.
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Review URL: https://webrtc-codereview.appspot.com/1187005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3664 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 08:46:25 +00:00
becf9c897c
Fix mismatch between different NACK list lengths and packet buffers.
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This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors.
BUG=1289
Review URL: https://webrtc-codereview.appspot.com/1065007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 15:09:57 +00:00
e8ef807a2d
Added possibility to run quality modes test. Added possibility to input arguments to the test. The test will (for each frame) log the values in contentMetrics to a txt-file. The txt-file can optionally be saved in a specific place. Fixed an issue where video_coding_test crashed if there weren't any parameter submitted to an input argument.
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BUG=
Review URL: https://webrtc-codereview.appspot.com/772005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3068 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-09 16:16:41 +00:00
14b43beb7c
Move src/ -> webrtc/
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TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00