| ad856229a796a8efa1126ef8aa8d238f2b0a2b21 |
|
27-Nov-2015 |
pbos <pbos@webrtc.org> |
Use webrtc/base/logging.h for voice_engine.
BUG=webrtc:5118
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1474363002
Cr-Commit-Position: refs/heads/master@{#10827}
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
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}
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
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| f353dd59b59aea3c3a1c64aa20a66c2a8c32225e |
|
06-May-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
VoE: cleanup VoENetwork implementation
Changes:
1. Documented return values of VoENetwork methods.
2. In VoENetworkImpl: replaced calls to SetLastError() with LOG_F(). SetLastError() is not used anymore because no one is calling LastError() to check for last error. Also, its usage is being removed in Video Engine and we want to be consistent.
3. In VoENetworkImpl: removed WEBRTC_TRACE() usage.
4. In VoENetworkImpl: replaced some defensive code with assert(). We are now assuming that the caller has called VoEBase::Init() where needed. We are also assuming that it is invalid to pass nullptr where data is expected.
5. Updated unit tests accordingly.
R=henrika@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53369004
Cr-Commit-Position: refs/heads/master@{#9145}
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| 0d266054acece70259fc1e85026194154f41e5a0 |
|
04-May-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
VoE: apply new style guide on VoE interfaces and their implementations
Changes:
1. Ran clang-format on VoE interfaces and their implementations.
2. Replaced virtual with override in derived classes.
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49239004
Cr-Commit-Position: refs/heads/master@{#9130}
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
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| 4591fbd09f9cb6e83433c49a12dd8524c2806502 |
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20-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
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
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| b1f50100757036cf475072c26f5f374eee9588ca |
|
24-Mar-2014 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
VoE changes to allow forwarding of packets from VoE to ViE BWE.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5757 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| 9b6eefcedf3603c9c7b1bd424653ae0a08c1a9f4 |
|
24-Sep-2013 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket
Ensures that we always call DeRegisterExternalTransport() even if a fuzz test calls DeRegisterExternalTransport in an unintialized state.
TBR=tommi
BUG=296804 in crbug.com
Review URL: https://webrtc-codereview.appspot.com/2275008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4827 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
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| 822fbd8b68ffdb481b9557e2950ae8d6657c8ce6 |
|
16-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
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| 676ff1ed893d6ae59a7ce29a4428e0d7c9f855d9 |
|
07-Aug-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Ref-counted rewrite of ChannelManager.
The complexity of the last ChannelManager and potentially usage of it as well caused race conditions and deadlocks in loopback voe_auto_test. This ref-counted solution takes no long-term locks, uses less locks overall and is significantly easier to understand.
ScopedChannel has been split up into a ChannelOwner with a reference to a channel and an Iterator over ChannelManager. Previous code was really used for both things. ChannelOwner is used as a shared pointer to a channel object, while an Iterator should work as expected.
BUG=2081
R=tommi@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1802004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4502 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| aa4d96a134a03f998d52fb9699845d9c644eb24b |
|
16-Jul-2013 |
tnakamura@webrtc.org <tnakamura@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| 66b2e5c05a3f2a93d634d1dbbcbb283fb218ca4f |
|
05-Jul-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| d900e8bea84c474696bf0219aed1353ce65ffd8e |
|
03-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Proper spacing for end-of-namespace comments.
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1760006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| 8c845cb6233e5d9c3a0e510d8553057470be3ee1 |
|
02-May-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Relax VoE's max packet length threshold.
The earlier threshold would cause packets from a currently available
codec (L16, 32 kHz, stereo) to be discarded.
TESTED=voe_cmd_test using L16, 32 kHz, stereo now works properly.
R=henrika@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1305008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3936 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| 6141e13873d0fdea626de08dfec2efa2c9171c76 |
|
09-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 -> int32_t in voice_engine/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1305004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3792 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| b9e402d99f25d879fd62777e6646e734be07348b |
|
04-Apr-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove WEBRTC_*_ENGINE_NETWORK_API use
Review URL: https://webrtc-codereview.appspot.com/1203009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3767 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| 0c45957e3a6963e1460c0b5b62a6adf43cf44314 |
|
03-Apr-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove UDP transport API from VoE
Review URL: https://webrtc-codereview.appspot.com/1236004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3757 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| 684f0577fbe4ea393fef1dddf2ca7d02e3205b49 |
|
14-Mar-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r3667 and r3665
Review URL: https://webrtc-codereview.appspot.com/1199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| 361bac7a4f30a81e58c53ba86c58ffec085306d7 |
|
13-Mar-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work.
Review URL: https://webrtc-codereview.appspot.com/1029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| 0989fb7bfa482074e0161ea177653a44174ac492 |
|
15-Feb-2013 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make VoiceEngineImpl inherit from VoiceEngine.
This associates the two types instead of incorrectly reinterpret casting
VoiceEngineImpl* to VoiceEngine* (since these types were previously unrelated).
Please see more details in the bug for how this is currently causing problems
with security tools.
BUG=38612
Review URL: https://webrtc-codereview.appspot.com/1099013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3520 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
|
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/
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
/external/webrtc/webrtc/voice_engine/voe_network_impl.cc
|