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gerrit-public.fairphone.software
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fp2-dev
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platform
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external
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chromium_org
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third_party
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webrtc
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24f07023ade0cf0058bd3546999068c902a94dea
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voice_engine
0de0049
1. adding request of ACM version in the manual mode of voe_auto_test
by minyue@webrtc.org
· 11 years ago
8da2f65
Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket
by henrika@webrtc.org
· 11 years ago
510ee1b
Remove deprecated AudioCodingModule::Destroy.
by andrew@webrtc.org
· 11 years ago
2529558
Enable SetInitialPlayoutDelay on Android.
by dwkang@webrtc.org
· 11 years ago
80142aa
Small refactoring of AudioProcessing use in channel.cc.
by andrew@webrtc.org
· 11 years ago
39e22a1
Adds a new voice engine warning for the typing noise off state.
by jiayl@webrtc.org
· 11 years ago
4489c51
This issue is related to https://chromereviews.googleplex.com/9908014/
by minyue@webrtc.org
· 11 years ago
f46fff6
OpenSL (not default): Enables low latency audio on Android.
by henrike@webrtc.org
· 11 years ago
7b30ce3
Remove include_dirs from voice_engine.gyp.
by pbos@webrtc.org
· 11 years ago
5cf83f4
Remove redundant STR_CASE_CMP macro definitions.
by andrew@webrtc.org
· 11 years ago
db74c61
Adds support for combining RTX and FEC/RED.
by stefan@webrtc.org
· 11 years ago
06eaa54
Restore severity precondition to logging.h.
by andrew@webrtc.org
· 11 years ago
0f62690
Revert 4671 "Enable SetInitialPlayoutDelay on Android."
by mflodman@webrtc.org
· 11 years ago
0fe8944
Enable SetInitialPlayoutDelay on Android.
by dwkang@webrtc.org
· 11 years ago
c766a74
Update SSRC in RtpRtcp for audio channel so that it can have NTP values for further AV sync.
by dwkang@webrtc.org
· 11 years ago
8c6633c
Add isolate configuration for Android for all tests.
by kjellander@webrtc.org
· 11 years ago
e21b64b
Disables RtpRtcpTest.CanTransmitExtraRtpPacketsWithoutError as it flakily breaks the waterfall. See http://chromegw.corp.google.com/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/99/steps/voe_auto_test/logs/stdio the cl triggering it was a no-change (disabled some other broken tests).
by henrike@webrtc.org
· 11 years ago
3540c82
Isolate GYP target and .isolate files for tests
by kjellander@webrtc.org
· 11 years ago
a20e2d4
Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ..."""
by stefan@webrtc.org
· 11 years ago
c0976d2
Revert 4582 "Reverts a second set of reverts caused by a bug in ..."
by henrike@webrtc.org
· 11 years ago
efe1f0f
Reverts a second set of reverts caused by a bug in a dependency.
by stefan@webrtc.org
· 11 years ago
7fc75bb
Update talk to 50918584.
by wu@webrtc.org
· 11 years ago
1e817c3
Roll chromium_revision 214260:217707 and gflags 45:84
by fischman@webrtc.org
· 11 years ago
e155918
Revert 4547 "Isolate GYP target and .isolate files for tests"
by kjellander@webrtc.org
· 11 years ago
298bbdb
Isolate GYP target and .isolate files for tests
by kjellander@webrtc.org
· 11 years ago
d171544
Disable CanTransmitExtraRtpPacketsWithoutError on Windows.
by pbos@webrtc.org
· 11 years ago
7d82c9d
Hand over loopback packets to a network thread.
by pbos@webrtc.org
· 11 years ago
a4a1afa
Delete Channels without ChannelManager lock.
by pbos@webrtc.org
· 11 years ago
b3ada15
Ref-counted rewrite of ChannelManager.
by pbos@webrtc.org
· 11 years ago
f3bae63
Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer.
by phoglund@webrtc.org
· 11 years ago
44634a6
Disabled SsrcPropagatesCorrectly on Linux.
by phoglund@webrtc.org
· 11 years ago
3f45c2e
Switch C++-style C headers with their C equivalents.
by pbos@webrtc.org
· 11 years ago
acb00f5
Adds all unittests to android NDK-APK framework.
by henrike@webrtc.org
· 11 years ago
5ce8723
Merge r4374 from stable to trunk.
by xians@webrtc.org
· 11 years ago
0e6fa8c
Merge r4394 from stable to trunk.
by xians@webrtc.org
· 11 years ago
44f1239
Merge r4326 from stable to trunk.
by xians@webrtc.org
· 11 years ago
6349e17
Default constructor for RtcpAppHandler.
by pbos@webrtc.org
· 11 years ago
1c8d5a0
clean up incomplete revert in r4357 Also revert r4319, will follow up with pbos
by tnakamura@webrtc.org
· 11 years ago
0ba496b
Revert r4301
by tnakamura@webrtc.org
· 11 years ago
9d788a1
Revert r4321 "Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered"
by elham@webrtc.org
· 11 years ago
b89eed3
Revert r4322 "Support sending multiple report blocks and keeping track of statistics on
by elham@webrtc.org
· 11 years ago
6a4acb9
Fix some voe_auto_test uninitialised-value errors.
by pbos@webrtc.org
· 11 years ago
46088d2
Support sending multiple report blocks and keeping track of statistics on several SSRCs.
by stefan@webrtc.org
· 11 years ago
446ea2e
Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered.
by stefan@webrtc.org
· 11 years ago
d5e5863
Initialize payload-type frequency in channel.cc.
by pbos@webrtc.org
· 11 years ago
a32d18f
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
by stefan@webrtc.org
· 11 years ago
3b89e10
Proper spacing for end-of-namespace comments.
by pbos@webrtc.org
· 11 years ago
f47d0f8
Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing.
by andrew@webrtc.org
· 11 years ago
46cec2a
Reorganize test targets in WebRTC
by kjellander@webrtc.org
· 11 years ago
0e7cd85
Fixed Rtp/Rtcp tests
by pwestin@webrtc.org
· 11 years ago
9aeef32
Fix size_t to int conversion error on Win64.
by andrew@webrtc.org
· 11 years ago
5f545ff
Fix for STL vector function data not available.
by pwestin@webrtc.org
· 11 years ago
4aa9f1a
Connect ACM with RTP module for audio NACK.
by pwestin@webrtc.org
· 11 years ago
b8171ff
Wire up Nack for Voe
by pwestin@webrtc.org
· 11 years ago
915ca75
Fix error in mixing test for supported sample rates.
by andrew@webrtc.org
· 11 years ago
a80d94b
Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling.
by wu@webrtc.org
· 11 years ago
92bfbbd
Replace the old resampler with SincResampler in the voice engine signal path.
by andrew@webrtc.org
· 11 years ago
2753b76
Add dummy audio NACK APIs
by niklas.enbom@webrtc.org
· 11 years ago
50a4d9f
Remove #pragma once
by pbos@webrtc.org
· 11 years ago
6696fba
Breaking out RTP header parsing from the RTP module.
by stefan@webrtc.org
· 11 years ago
5221d1c
Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp.
by andrew@webrtc.org
· 11 years ago
d557734
API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay.
by turaj@webrtc.org
· 11 years ago
471ae72
Include files from webrtc/.. paths in voice_engine/
by pbos@webrtc.org
· 11 years ago
8510750
Make sure VoiceEngine tests only include one test framework.
by pbos@webrtc.org
· 11 years ago
ca7a9a2
Remove const for plain data types in voice_engine/
by pbos@webrtc.org
· 11 years ago
28832e1
Refactoring for typing detection
by niklas.enbom@webrtc.org
· 11 years ago
c0fc487
Allow voe_cmd_test to select Opus mono (now the default).
by andrew@webrtc.org
· 11 years ago
ad9cee8
Relax VoE's max packet length threshold.
by andrew@webrtc.org
· 11 years ago
9e0d9a6
Disabled flaky test.
by phoglund@webrtc.org
· 11 years ago
4a68e95
Replace Resampler with PushResampler in transmit_mixer.
by andrew@webrtc.org
· 11 years ago
166153e
Consolidate common_audio into a single target.
by andrew@webrtc.org
· 11 years ago
b6fadb1
Add a wrapper around PushSincResampler and the old Resampler.
by andrew@webrtc.org
· 11 years ago
570c4a5
Fix for "RTP dynamic payload type 100 is reserved"
by henrika@webrtc.org
· 11 years ago
237fe4f
Remove vim/emacs modelines from .gypi files
by pbos@webrtc.org
· 11 years ago
f272497
Adding playout buffer status to the voe video sync
by pwestin@webrtc.org
· 11 years ago
54f03bc
WebRtc_Word32 -> int32_t in voice_engine/
by pbos@webrtc.org
· 11 years ago
8ec8955
Remove the old unused udp_transport
by pwestin@webrtc.org
· 11 years ago
1d25eac
Resolves TSan v2 reports data races in voe_auto_test.
by henrika@webrtc.org
· 11 years ago
ef91cbf
Remove WEBRTC_*_ENGINE_NETWORK_API use
by pwestin@webrtc.org
· 11 years ago
c39749a
Fix no received audio in tests.
by pwestin@webrtc.org
· 11 years ago
84423e9
Disabling MixingTests due to race conditions.
by henrika@webrtc.org
· 11 years ago
45ce6a8
TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC
by henrika@webrtc.org
· 11 years ago
e493218
Remove UDP transport API from VoE
by pwestin@webrtc.org
· 11 years ago
9e8a401
Fixes memory leak in AudioLevel class reported by memory try bots.
by henrika@webrtc.org
· 11 years ago
c4efe71
Fixes data race in WebRTCAudioDeviceTest.StartRecording reported by ThreadSanitizer
by henrika@webrtc.org
· 11 years ago
3b6f728
Removed CPU APIs from VoEHardware. Code is now only used by test applications.
by henrike@webrtc.org
· 11 years ago
fa2dd22
Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested.
by solenberg@webrtc.org
· 11 years ago
2ffc8bf
Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..."
by wu@webrtc.org
· 11 years ago
365ca40
Removed CPU APIs from VoEHardware. Code is now only used by test applications.
by henrike@webrtc.org
· 11 years ago
f386e2b
Remove VoE's default call in Trace::SetLevelFilter.
by andrew@webrtc.org
· 11 years ago
31b4448
Alphabetize include order in fake_voe_external_media.h.
by andrew@webrtc.org
· 11 years ago
13f66d1
Add some VoE and AudioProcessing mocks.
by andrew@webrtc.org
· 11 years ago
0c1f10b
Enable the below APIs for iOS.
by sjlee@webrtc.org
· 11 years ago
aa922de
Move the VoE tests to use external transport instead of the built in udp transport
by pwestin@webrtc.org
· 11 years ago
912b7f7
Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004
by pwestin@webrtc.org
· 11 years ago
2daec4c
Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work.
by pwestin@webrtc.org
· 11 years ago
15a03fd
Remove DTMF detection. Talk team has been in the loop and there is no need for
by turaj@webrtc.org
· 12 years ago
0f919be
Remove the error return on SetAGC failure introduced by r3605.
by andrew@webrtc.org
· 12 years ago
8665399
None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise.
by turaj@webrtc.org
· 12 years ago
b79627b
Expose the capture-side AudioProcessing object and allow it to be injected.
by andrew@webrtc.org
· 12 years ago
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