Commit Graph

26 Commits

Author SHA1 Message Date
pwestin@webrtc.org
c450a19669 Removed Version function from all modules.
TBR=henrik_a
Review URL: http://webrtc-codereview.appspot.com/329023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1330 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 15:00:12 +00:00
henrika@webrtc.org
af71f0e5d9 Fixes two minor issues reported by the Coverty Integration Manager.
BUG=none
TEST=voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/302002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1098 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-05 07:02:22 +00:00
perkj@webrtc.org
68f2168978 Remove global voe::Channel::numSocketThreads.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1067 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 18:11:23 +00:00
henrik.lundin@webrtc.org
524eb48081 Removing deprecated NetEQ APIs
Removing WebRtcNetEQ_GetPreferredBufferSize and
WebRtcNetEQ_GetCurrentDelay and all dependent APIs.

Review URL: http://webrtc-codereview.appspot.com/289006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1063 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 16:21:22 +00:00
xians@webrtc.org
e07247af8d Valgrind reports a racing condition on _sending because it is accessed by
both TransmitMixer::PrepareDemux() and StartSend()/StopSend().
Put a lock to resolve it.
Review URL: http://webrtc-codereview.appspot.com/293005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1038 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-28 16:31:28 +00:00
xians@webrtc.org
83661f534e fixing the racing conditions
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1025 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 10:58:15 +00:00
henrik.lundin@webrtc.org
df10de4b27 Removing statistics API from NetEQ
Removing WebRtcNetEQ_GetJitterStatistics(),
WebRtcNetEQ_ResetJitterStatistics(), and the associated
struct WebRtcNetEQ_JitterStatistics. The change ripples
through all the way to the VoiceEngine API.

Review URL: http://webrtc-codereview.appspot.com/285002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@998 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 09:36:23 +00:00
henrike@webrtc.org
31d30700d6 Addressed review comments from http://webrtc-codereview.appspot.com/256004/
Review URL: http://webrtc-codereview.appspot.com/256007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@979 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 19:59:32 +00:00
niklas.enbom@webrtc.org
af26f64616 Inband DTMF stereo support
Review URL: http://webrtc-codereview.appspot.com/267011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@956 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-16 12:41:36 +00:00
andrew@webrtc.org
755b04a06e Add RMS computation for the RTP level indicator.
- Compute RMS over a packet's worth of audio to be sent in Channel, rather than the captured audio in TransmitMixer.
- We now use the entire packet rather than the last 10 ms frame.
- Restore functionality to LevelEstimator.
- Fix a bug in the splitting filter.
- Fix a number of bugs in process_test related to a poorly named
  AudioFrame member.
- Update the unittest protobuf and float reference output.
- Add audioproc unittests.
- Reenable voe_extended_tests, and add a real function test.
- Use correct minimum level of 127.

TEST=audioproc_unittest, audioproc, voe_extended_test, voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/279003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@950 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 16:57:56 +00:00
stefan@webrtc.org
b351d6a8d8 Reverting rev 929 due to failing assert on Linux.
Failing at: audio_buffer.cc:159

TBR=mflodman
Review URL: http://webrtc-codereview.appspot.com/270008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@935 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-11 13:26:05 +00:00
niklas.enbom@webrtc.org
50b3cbe979 First pass. You can now enable a stereo codec and send and receive. This does not include more advances use cases (DTMF etc), but I'd rather keep the CLs manageable.
Review URL: http://webrtc-codereview.appspot.com/269007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@929 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-11 08:31:32 +00:00
henrike@webrtc.org
e2a34f8275 Removes the API for setting RX VAD since the RX vad should always be on anyways.
Review URL: http://webrtc-codereview.appspot.com/264001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@897 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-07 21:33:24 +00:00
andrew@webrtc.org
a4b9660372 Add mistakenly removed VAD enabling function.
This resolves the unknown VAD status warnings introduced in r845.

BUG=
TEST=voe_cmd_test

Review URL: http://webrtc-codereview.appspot.com/252004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@879 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 01:36:27 +00:00
henrike@webrtc.org
b37c628ae4 Fixes crash due to r841.
Review URL: http://webrtc-codereview.appspot.com/256004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@853 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 23:53:04 +00:00
andrew@webrtc.org
2c74bab8b9 Remove unneeded assert and tracing.
This is related to r840.

BUG=
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/239019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@845 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 19:54:20 +00:00
henrike@webrtc.org
066f9e5a2f Ray, please verify that this cl fixes the issue. Once the verification has been made, please review:
Henrik A: VoE
Andrew: audio_conference_mixer

Thanks!
Review URL: http://webrtc-codereview.appspot.com/241010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@841 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 23:15:47 +00:00
pwestin@webrtc.org
1da1ce0da5 First implementation of simulcast, adds VP8 simulcast to video engine.
Changed API to RTP module
Expanded Auto test with a test for simulcast
Made the video codec tests compile
Added the vp8_simulcast files to this cl
Added missing auto test file
Review URL: http://webrtc-codereview.appspot.com/188001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@736 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 15:19:55 +00:00
wu@webrtc.org
fcd12b3b7d Add necessary spaces to log.
Review URL: http://webrtc-codereview.appspot.com/148002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@602 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-15 20:49:50 +00:00
zakkhoyt@google.com
b448ae229c Permanently adding additional logs
Review URL: http://webrtc-codereview.appspot.com/137024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@577 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-09 17:41:49 +00:00
andrew@webrtc.org
ceb148ce59 Fix compile warnings in Release configuration.
Review URL: http://webrtc-codereview.appspot.com/119003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@424 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-23 17:53:54 +00:00
andrew@webrtc.org
f81f9f8c2a Add -Werror and -Wextra to the Linux build.
Includes all fixes required for -Wextra.
Review URL: http://webrtc-codereview.appspot.com/117006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@410 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-19 22:56:22 +00:00
xians@google.com
4257b175f3 The 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.

More detail about the fix please refer to 
http://code.google.com/p/webrtc/issues/detail?id=36
Review URL: http://webrtc-codereview.appspot.com/93020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@322 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-08 12:02:36 +00:00
xians@google.com
0b0665acc1 This 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.
BR,
/SX
Review URL: http://webrtc-codereview.appspot.com/89014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@320 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-08 08:18:44 +00:00
xians@google.com
22963abffe Removing 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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@294 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-03 12:40:23 +00:00
niklase@google.com
470e71d364 git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00