e66a0c0ce7Correcting one bug and changing sleep time while waiting for packets.
mflodman@webrtc.org
2011-08-16 08:11:15 +00:00
bf5d2ba8fbThis CL will exclude the built-in ADM when building with chromium. When building with chromium, we use dummy ADM for all the platforms. Review URL: http://webrtc-codereview.appspot.com/108001
xians@google.com
2011-08-16 07:44:19 +00:00
74a49a833eTwo changes: 1) Libjingle and webrtc use different jsoncpp, so we need to distinguish them. 2) Update the webrtcsession_unittest: * Use the Thread::SleepMs * Remove main, which has been defined in talk/base/unittest_main. Review URL: http://webrtc-codereview.appspot.com/107003
wu@webrtc.org
2011-08-12 00:20:24 +00:00
60873adc3ertp_sender_video: Modify behavior on send video packet error. This issue was already updated in CL r217, and accidentally reverted in CL r231. Review URL: http://webrtc-codereview.appspot.com/106004
mikhal@google.com
2011-08-11 22:30:00 +00:00
b0d7a87bb0Mock implementation for the UI of the linux version of the peerconnection client. At this point, there's not a lot too it as it only shows what the UI will look like and basically mimics what the Windows version does presently. Review URL: http://webrtc-codereview.appspot.com/92018
tommi@google.com
2011-08-10 09:03:29 +00:00
c3976c8c92exclude the video_capture_module_test target from building with chromium.
xians@google.com
2011-08-10 08:02:41 +00:00
14acdbc14dUpdate fixed-point profile output due to r313.
andrew@webrtc.org
2011-08-09 01:54:03 +00:00
492dbc258eUse the full path instead of the current directory. In chromium build this libjingle.gyp will be included by third_party/libjingle/libjingle.gyp. In that case the "." will mean the third_party/libjingle/ instead of what we want - third_party_mods/libjingle. Review URL: http://webrtc-codereview.appspot.com/100004
ronghuawu@google.com
2011-08-09 00:36:01 +00:00
8deee041b8Enable full chromium_code warnings for all platforms.
ajm@google.com
2011-08-08 20:47:41 +00:00
4a4d7c6693Fix a compile error on MSVC.
ajm@google.com
2011-08-08 20:40:31 +00:00
4257b175f3The Cl is to support mixing output file in a stereo stream. Previously, an assert will be triggered in case it is not a mono stream. With the CL, the mono file stream will be copied into a strereo stream and mixed with the channel stream.
xians@google.com
2011-08-08 12:02:36 +00:00
0b0665acc1This CL changes all the freq relevant variables to be int type. So it will take away the VoE "comparison between signed and unsigned integer expressions" warnings.
xians@google.com
2011-08-08 08:18:44 +00:00
35f534529b* Point the webrtc libjingle dependency to third_party_mods. * For unchanged files, change the third_party_mods libjingle.gyp to point to the original version of libjingle. Review URL: http://webrtc-codereview.appspot.com/89015
ronghuawu@google.com
2011-08-05 22:08:29 +00:00
a1ec48dec2Updates to VP8 wrapper: 1) Added VP8E_SET_STATIC_THRESHOLD which sets threshold for forcing static macroblocks to skipped mode (useful for encoding of static background). 2) Removed some code in decoding to allow for decoding of incomplete frames. 3) Minor change to MaxIntraTarget() 4) Added control setting for maximum (target) size of intra-frame. 5) Increased kf_max_dist(reduce/remove key-frame periodicity) since we use kf_auto_mode. Review URL: http://webrtc-codereview.appspot.com/101001
marpan@google.com
2011-08-05 20:51:32 +00:00
7fa20c3880exclude both audio_device_test_api and audio_device_test_func test targets from building with chromium. Review URL: http://webrtc-codereview.appspot.com/97006
xians@google.com
2011-08-05 12:04:30 +00:00
15ad55b03cThis CL fixes the gclient runhooks problem with building with chromium. The problem is src/third_party/webrtc/third_party/protobuf/protobuf.gyp not found Review URL: http://webrtc-codereview.appspot.com/92008
xians@google.com
2011-08-05 08:21:25 +00:00
e256187f8b* Push the //depotGoogle/chrome/third_party/libjingle/...@38654 to svn third_party_mods\libjingle. * Update the peerconnection sample client accordingly. Review URL: http://webrtc-codereview.appspot.com/60008
ronghuawu@google.com
2011-08-04 17:44:30 +00:00
88bd440ef6Removing the "initialized after" warnings. This CL tweat the order of the initialization in the constructor to adapt to the order of declaration of the members. Review URL: http://webrtc-codereview.appspot.com/92007
xians@google.com
2011-08-04 15:33:30 +00:00
55ce2d8a25This changelist is basically a code cleanup and restructuring. Main things is that we now have a function for time to frequency transformation and the delay estimator is separated into its own struct and files. Review URL: http://webrtc-codereview.appspot.com/90004
bjornv@google.com
2011-08-04 07:16:54 +00:00
ce7c2a231eAdd SSE2 support for Windows.
ajm@google.com
2011-08-04 01:50:00 +00:00
22963abffeRemoving the "initialized after" warnings. This CL tweat the order of the initialization in the constructor to adapt to the order of declaration of the members. Review URL: http://webrtc-codereview.appspot.com/99002
xians@google.com
2011-08-03 12:40:23 +00:00
c4df42be6bAdded option for enabling the video protection settings and testing packet loss/delay in auto_test_loopback.
marpan@google.com
2011-08-02 18:53:33 +00:00
e91655f8e2Incorrect parameters being passed to trace function.
frkoenig@google.com
2011-08-01 23:14:32 +00:00
679450f4a6media_opt_util: Update robustness settings for Hybrid mode. Updated table for the computation of the adjustment factor. Review URL: http://webrtc-codereview.appspot.com/93013
mikhal@google.com
2011-08-01 22:14:58 +00:00
5fc2dcd64aChange to make the VP8-RTP Fragmentation (FI bits) setting (in the payload header) agree with "draft-westin-payload-vp8-02" document.
marpan@google.com
2011-08-01 21:47:46 +00:00
191b780741Added a correction factor to FEC overhead in media_opt_util. This is too handle cases of rate-mismatch (at low rates/low packet number) between estimate in mediaOpt and actual FEC generated in RTP. Review URL: http://webrtc-codereview.appspot.com/93012
marpan@google.com
2011-08-01 19:59:57 +00:00
b5c49ff8afRename all files/classes shared by Linux and Mac to Posix.
ajm@google.com
2011-08-01 17:04:04 +00:00
b29d940db7VCM: Updating Media Opt: 1. Removed protection method specific code from SetTargetRates 2. Updated encoding rate following protection settings 3. Removing RTT max threshold from NACK, as it is not used in the receiver side. 4. Two bug fixes: FEC conversion function fix (line #133) and residual loss calculation (line #94) 5. Removing compiler warnings 6.. Removed unused code and general clean-up. Review URL: http://webrtc-codereview.appspot.com/96002
mikhal@google.com
2011-08-01 16:39:20 +00:00