generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
2,171 Commits 1 Branch 0 Tags 108 MiB
899baa821b
Commit Graph

47 Commits

Author SHA1 Message Date
tina.legrand@webrtc.org
4517585db5 Adding separate payload types for stereo modes 
BUG=Issue 452
TEST=audio_coding_test, voe_auto_test, voe_cmd_test

Edit: adding Patrik to review:
src/modules/rtp_rtcp/source/rtp_receiver.cc
...and Shijing to review:
src/voice_engine/main/source/channel.cc
src/voice_engine/main/test/cmd_test/voe_cmd_test.cc

Review URL: https://webrtc-codereview.appspot.com/540004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2340 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-01 09:27:35 +00:00
pwestin@webrtc.org
2853dde520 Refactor the internal API to the rtp/rtcp module. 
Combination of previous CLs in revisions 2211, 2212, 2214, 2215, 2216.
Review URL: https://webrtc-codereview.appspot.com/570008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-11 11:08:54 +00:00
turaj@webrtc.org
3c383abd27 Revert 2211 - Refactor the internal API to the rtp/rtcp module. 
Review URL: https://webrtc-codereview.appspot.com/568004

A series of CL:s by Patrik W. is breaking the auto-test. It started with CL 2211, but the later CL:s seems dependent on another. So I decided to go in reverse order and revert all of them.

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/563011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 23:01:04 +00:00
pwestin@webrtc.org
0774838f3d Refactor the internal API to the rtp/rtcp module. 
Review URL: https://webrtc-codereview.appspot.com/568004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 12:33:50 +00:00
andrew@webrtc.org
e59a0aca6a Fix AudioFrame types. 
volume_ is not set anywhere so I'm removing it.

BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/556004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2196 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-08 17:12:40 +00:00
andrew@webrtc.org
63a509858d Rename AudioFrame members. 
BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/542005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 23:56:37 +00:00
pwestin@webrtc.org
49888ce428 Breaking out send side bitrate controll cont. 
Review URL: https://webrtc-codereview.appspot.com/475004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 05:25:53 +00:00
tina.legrand@webrtc.org
16b6b90a82 Split of stereo packets moved 
In this CL I have rewritten the way we handle stereo packets in VoE.
Before this change we split the packets in the RTP module and added two packets to ACM, one for the left channel and one for the right. This lead to timing problems caused when a different thread called RecOut in between the two calls to add stereo packet to ACM. (RecOut is called to pull audio data, decode packets, on the receiving side).

While doing the change I also took the opportunity to changed some functions so that the data stream is uint8 everywhere.

The list of files in this CL is long, but should be fairly easy to review. It is difficult to see what has been changed  in some of the tests, but I can explain offline.

Reviewers:
Björn - /src/modules/audio_coding
Patrik - /src/modules/rtp_rtcp
Patrik -/src/modules/utility
Henrik A - /src/voice_engine

BUG=410
TEST=voe_cmd_test

Review URL: https://webrtc-codereview.appspot.com/473003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2012 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-12 11:02:38 +00:00
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