generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
1,701 Commits 1 Branch 0 Tags 108 MiB
07c68b9c9d
Commit Graph

137 Commits

Author SHA1 Message Date
leozwang@webrtc.org
07c68b9c9d Correct wrong usage of WebRtc_Word8 in rtp and udp module 
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/418001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1798 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-29 16:09:51 +00:00
phoglund@webrtc.org
2d124f3d88 Enabled the volume tests we believe are nonflaky and the vie_auto_test extended tests. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/422002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-29 14:34:06 +00:00
phoglund@webrtc.org
b45ceed9ef Rewrote the call report test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/399006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-20 08:55:04 +00:00
leozwang@webrtc.org
a52838b684 Update Android.mk and add test app 
Review URL: https://webrtc-codereview.appspot.com/388010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-17 01:16:43 +00:00
xians@webrtc.org
3ab6dda5cb Truncated the volume to 255 when the users set the volume above 100%. 
Allowed the users to set the volume above 100% when AGC is enabled, in this case AGC can gradually scale down the volume instead of jumping to 100% immediately.
Reduced the flakiness of the volume tests in linux.
Review URL: https://webrtc-codereview.appspot.com/387011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 18:15:54 +00:00
braveyao@webrtc.org
59d6cec291 Fix the crash at playing 48kHz stereo wav file. 
http://code.google.com/p/webrtc/issues/detail?id=208
Review URL: https://webrtc-codereview.appspot.com/396001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1681 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-14 18:17:16 +00:00
phoglund@webrtc.org
292da24166 New attempt. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/391006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1672 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-10 15:21:33 +00:00
phoglund@webrtc.org
dbe1e13b53 Fixed compilation error on Windows. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/389007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1670 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-10 14:03:44 +00:00
phoglund@webrtc.org
6b3bb89f12 Rewrote file test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/391004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-10 12:14:54 +00:00
phoglund@webrtc.org
aaa76f3ba8 Rewrote network test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/383003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-09 16:41:30 +00:00
phoglund@webrtc.org
78088c2f36 Removed warnings on Windows and enabled warnings-as-errors on Windows. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/377004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1624 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-07 14:56:45 +00:00
niklas.enbom@webrtc.org
87885e8409 This CL will look a bit strange. Essentially I've removed sanity checks for > 1 channel and then fixed the bugs that remained. Will add testing in a separate CL. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/390001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-07 14:48:59 +00:00
wu@webrtc.org
06c7dbae14 Disable flaky test AudioProcessingTest.TestVoiceActivityDetectionWithObserver. 
BUG=263
TBR=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/380009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1615 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 23:13:21 +00:00
phoglund@webrtc.org
56b85c6ba8 Reduced potential for flakiness in voice detection tests. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/367024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1612 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 18:48:33 +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
kma@webrtc.org
de66b91274 In voice engine, added member audioFrame to classes AudioCodingModuleImpl and VoEBaseImpl, 
and touched VoEBaseImpl::NeedMorePlayData and AudioCodingModuleImpl::PlayoutData10Ms(), for
performance reasons in Android platforms.
The two functions used about 6% of VoE originally. After the change, the percentage reduced
to about 0.2%.
Review URL: https://webrtc-codereview.appspot.com/379001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1589 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-01 18:39:44 +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
phoglund@webrtc.org
048eb7cda6 Finished rewriting the audio processing test. 
Partial rewrite of audio processing tests.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/367008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1561 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-27 11:58:41 +00:00
andrew@webrtc.org
b9d7d934de Rename interface/ to include/ in audio_processing. 
BUG=none
TEST=build on Linux.

Review URL: https://webrtc-codereview.appspot.com/367007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1552 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-25 19:21:13 +00:00
andrew@webrtc.org
24bd58e689 Properly count anonymous mixing participants. 
When _amountOfMixableParticipants == 1, we skip mixing and saturation
protection. Without this fix, an anonymous participant would only be
properly counted if it was the last added.

For example, if an anonymous participant was added first, followed by
a regular participant, _amoutOfMixableParticipants would == 1 and the
regular participant would not be mixed.

BUG=issue209
TEST=New test added to voe_auto_test to verify, and used voe_cmd_test.

Review URL: https://webrtc-codereview.appspot.com/367006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1551 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-25 18:57:44 +00:00
andrew@webrtc.org
eeaf3d1fc1 Merge /branches/3.2:r1380 to /trunk 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1523 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 06:30:02 +00:00
leozwang@webrtc.org
f5cacdce8c Fix line aligement 
Review URL: https://webrtc-codereview.appspot.com/373002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1516 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 23:14:13 +00:00
phoglund@webrtc.org
12dbc23851 Rewrote volume test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/367004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1506 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 16:03:04 +00:00
phoglund@webrtc.org
3b57ee0238 Rewrote DTMF test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/368001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1502 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 09:22:33 +00:00
leozwang@webrtc.org
2638577f03 Add an argument in ANDROID_NOT_SUPPORT macro 
Review URL: https://webrtc-codereview.appspot.com/363003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1499 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-20 18:45:45 +00:00
tommi@webrtc.org
9ff87db5c0 Remove the diamond inheritance pattern from VoEVideoSyncImpl in attempt to see if this fixes coverity reports. 
CID=10446,10445,10444,10443
Review URL: https://webrtc-codereview.appspot.com/343018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 15:05:36 +00:00
punyabrata@webrtc.org
ad1927d368 Changing the typing detection sensitivity as the current 
setting does not work well in some scenarios especially
using webcams with built-in microphones.
Review URL: https://webrtc-codereview.appspot.com/349009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-18 18:53:04 +00:00
phoglund@webrtc.org
5badc7e969 Put system cpu tests back in, improved documentation. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/350011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1452 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-18 10:46:07 +00:00
phoglund@webrtc.org
c12f815de6 Rewrote hardware test and fixed broken tests on Windows. 
Fixed broken tests on Windows, including old tests.

Rewrote hardware test.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/347008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1434 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-16 12:40:18 +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
braveyao@webrtc.org
f5c6573725 fix defect http://code.google.com/p/webrtc/issues/detail?id=215, audio device is not stopped appropriately. 
Review URL: http://webrtc-codereview.appspot.com/350008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-16 03:04:46 +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
henrik.lundin@webrtc.org
053c7991e3 Add minimum waiting time to NetEQ metrics 
Adding minWaitingTimeMs to ACMNetworkStatistics and to
NetworkStatistics. Also adding unittest.

TEST=audio_coding_unittests

Review URL: http://webrtc-codereview.appspot.com/350006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 14:16:44 +00:00
kjellander@webrtc.org
7f3c724e12 Renaming 47 files from .cpp to .cc 
In addition to our naming guidelines, this will cause these files to get parsed by Sonar, and to make searching/grepping the source using file extensions easier in the future.

BUG=
TEST=Compiling on Linux.

Review URL: http://webrtc-codereview.appspot.com/348005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1405 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 10:23:41 +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
phoglund@webrtc.org
01530a2ac2 Rewrote the rcp_rtcp test. 
Finished rewriting the rtp_rtcp test.

Rewrote first RTP RTCP test

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/342007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-11 12:26:34 +00:00
phoglund@webrtc.org
0aa7b32652 Finished rewriting the codec test. 
Rewrote more tests.

Rewrote most of the codec test and removed it from the regular test.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/329008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1355 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-09 11:15:46 +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
henrik.lundin@webrtc.org
d439870473 Adding two new network metrics to NetEQ 
Clock-drift (in parts-per-million) and peaky-jitter mode status.
Both metrics are propagated to the VoE API. Tests are added
in the NetEQ unittests, and to some extent in ACM unittests
and VoE tests.

Introducing a proper translation between structs NetworkStatistics
and ACMNetworkStatistics.

Note: The file neteq_network_stats.dat in resources must be updated
for the unittests to pass.

Review URL: http://webrtc-codereview.appspot.com/337005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 13:09:55 +00:00
andrew@webrtc.org
3192d655bd Fix for devices lacking stereo support. 
The number of capture channels can only be determined upon receiving the
first captured frame. We now assume stereo capture by default and set the
number of AudioProcessing input channels based on captured frames.

TEST=Windows mono-only device now runs AudioProcessing correctly (NS etc.), voe_auto_test (though some new, seemingly unrelated, tests are failing)

Review URL: http://webrtc-codereview.appspot.com/330013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1273 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 18:00:59 +00:00
henrik.lundin@webrtc.org
dbba1f969f Packet waiting-time statistics 
Adding new statistics API to NetEQ, reporting the waiting time
for each frame. The output is raw waiting time for the frames
that have been decoded since the last statistics report (or
maximum 100 frames). The statistics are reset on each query.

Implemented functionality in ACM to query NetEQ for the raw
waiting times, and process it to produce max, average and
median.

Updating common_types.h and VoiceEngine tests to include the
new metrics.

Unit tests are also added for NetEQ and AcmNetEq.

Review URL: http://webrtc-codereview.appspot.com/328011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 15:45:05 +00:00
phoglund@webrtc.org
f3cea2336b Added an empty voice engine unit test binary in order to get correct coverage measurements. This will make the voice engine show up in the coverage measurements. The empty test is necessary to get the coverage tool to pick it up (and it will be easier to start writing unit tests for the voice engine later). 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/334003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1245 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 10:14:53 +00:00
phoglund@webrtc.org
fda17c2b00 Rewrote NetEQ test, made standard suite run googletestified tests too. 
The standard suite will now also run the googletestified tests.

Removed NetEQ tests from the standard test.

Initial version of new neteq test. Moved fixtures to own folder.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/328010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1242 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 09:07:37 +00:00
phoglund@webrtc.org
86a9f9b946 Fixed build error. 
Merge branch 'master' into voe_rewrites

Conflicts:
	src/voice_engine/main/test/auto_test/standard/after_streaming_fixture.cc

Revert "Rewrote network-before-streaming."

This reverts commit f1a07b813a90e4feef0a0737ebde9fb15acfd459.

Rewrote network-before-streaming.

Fixed strange build error.

Merge branch 'master' into voe_rewrites

Revert "Rewrote network-before-streaming."

This reverts commit f1a07b813a90e4feef0a0737ebde9fb15acfd459.

Rewrote network-before-streaming.

Nit fixes

Clarified some comments and method names.

Style fixes.

Removed tab characters.

Merge branch 'master' into voe_rewrites

Conflicts:
	src/voice_engine/main/test/auto_test/voe_standard_test.cc

Revert "Rewrote network-before-streaming."

This reverts commit f1a07b813a90e4feef0a0737ebde9fb15acfd459.

Rewrote network-before-streaming.

Rewrote the hold test.

Abstracted out resource handling and created a new fixture for starting and stopping playing.

Rewrote network-before-streaming.

Revert "Rewrote network-before-streaming."

This reverts commit f1a07b813a90e4feef0a0737ebde9fb15acfd459.

Rewrote network-before-streaming.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/333007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1230 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 10:24:46 +00:00
phoglund@webrtc.org
188fc35e07 Rewrote the hold and netw-before-streaming tests. 
Rewrote the hold test.

Abstracted out resource handling and created a new fixture for starting and stopping playing.

Rewrote network-before-streaming.

BUG=
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/331001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1228 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 09:36:03 +00:00
phoglund@webrtc.org
610e90e910 Completed rewrite of codec test. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/324011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1203 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 10:40:19 +00:00
leozwang@webrtc.org
eda2da796e Fix compilation errors 
Review URL: http://webrtc-codereview.appspot.com/322014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1195 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 20:03:09 +00:00
phoglund@webrtc.org
667eca6290 Rewrote the hardware-before-streaming test. 
Restructured the test hierarchy somewhat - there is now a fixture for before-voe-init time and one for after-voe-init time.

Rewrote the hardware-before-streaming test.
Separated unrelated tests out from the rtp_rtcp tests.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/323009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 13:55:34 +00:00
phoglund@webrtc.org
fe61bc3607 Merge branch 'master' into voe_create_test 
Fixed broken build.

Nit fix.

Fixed style issues.

Removed accidental comment-out.

Removed test that no longer makes sense.

Rewrote hardware-before-init and rtp-rtcp-before-streaming test code to gtest.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/320009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-12 17:02:16 +00:00