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d713143d99
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39 Commits

Author SHA1 Message Date
braveyao@webrtc.org
d713143d99 To support playing mono file with stereo codec as mixing with microphone capture 
BUG=413
TEST=Manual test.
Review URL: https://webrtc-codereview.appspot.com/460004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1953 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-29 10:39:44 +00:00
niklas.enbom@webrtc.org
40197d7b3b Fixing build issus on non-Win 
TBR: bjornv
Review URL: https://webrtc-codereview.appspot.com/460005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1940 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-26 08:45:47 +00:00
niklas.enbom@webrtc.org
5398d9583b Force commit of 449006' 
Review URL: https://webrtc-codereview.appspot.com/455006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1939 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-26 08:11:25 +00:00
mflodman@webrtc.org
3e820e5109 Remove RTP Keep-alive from VoE and ViE. The RTP module functionality will be removed in a follow-up CL shortly. 
TEST=VoE autotest and ViE autotest

Review URL: https://webrtc-codereview.appspot.com/458002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1929 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-23 09:41:44 +00:00
mflodman@webrtc.org
9a065d1eae VoiceEngine now uses pointer constructor of CriticalSectionScoped, instead of reference. 
BUG=184
TEST=Compiles on all platforms.

Review URL: https://webrtc-codereview.appspot.com/436001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1853 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-07 08:12:21 +00:00
leozwang@webrtc.org
0975d2158c Cleanup messy data type of unknown_payload_type 
BUG=322
TEST=build on all platforms
Review URL: https://webrtc-codereview.appspot.com/430002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-06 20:59:13 +00:00
leozwang@webrtc.org
813e4b0af0 Correct WebRtc_Word8 in voice engine 
BUG=http://code.google.com/p/webrtc/issues/detail?id=311&sort=-id
TEST=build on all platforms

Review URL: https://webrtc-codereview.appspot.com/425002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1816 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-01 18:34:25 +00:00
mflodman@webrtc.org
c80d9d9361 Removed default cases causing clang errors, -Wcovered-switch-default. 
BUG=
TEST=Bulid with clang version 3.1 (trunk 148911)

Review URL: https://webrtc-codereview.appspot.com/379008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 10:11:25 +00:00
xians@webrtc.org
79af734807 This patch fixes the converity warnings in voice engine. 
Review URL: https://webrtc-codereview.appspot.com/373017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1579 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-31 12:22:14 +00:00
henrika@webrtc.org
2919e95c2a Resolves Coverty issue #10347. 
Uninitialized member (UNINIT_CTOR).
Review URL: https://webrtc-codereview.appspot.com/369023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-31 08:45:03 +00:00
henrika@webrtc.org
f75901fa4c Resolves CID 10540: Copy into fixed size buffer (STRING_OVERFLOW). 
You might overrun the 32 byte fixed-size string "receiveCodec.plname" by copying "payloadName" without checking the length.
Note: This defect has an elevated risk because the source argument is a parameter of the current function.
Review URL: http://webrtc-codereview.appspot.com/352009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-16 08:45:42 +00:00
andrew@webrtc.org
7859e10985 Propagate decoding errors to the mixer module. 
Review URL: http://webrtc-codereview.appspot.com/348001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1417 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-13 00:30:11 +00:00
perkj@webrtc.org
ce5990cb0b Fix defect http://code.google.com/p/webrtc/issues/detail?id=222 
"ViE GetSentRTCPStatistics fails on a sending channel if it don't receive rtp video packets.

BUG=222
TEST= tested in loopback. No new test added yet.

Review URL: http://webrtc-codereview.appspot.com/343003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-11 13:00:08 +00:00
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