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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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5ec8feeb2b3bebe7d5e06262e4d3efd76b63d356
/
voice_engine
5ec8fee
Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.
by andrew@webrtc.org
· 10 years ago
11fa357
Add webrtc field trials API.
by andresp@webrtc.org
· 10 years ago
7ef6fff
Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
a2d1993
Removes parts of the webrtc::VoEVolumeControl sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
4ee3791
VoEVolumeTest: Adds error return tests.
by bjornv@webrtc.org
· 10 years ago
9754a5d
Make vie/voe_auto_test accept non-supported flags without error.
by kjellander@webrtc.org
· 10 years ago
7ea3607
Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255
by bjornv@webrtc.org
· 10 years ago
bb1e3ff
Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
8773fa6
Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
b0295bf
Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
3a87cff
Removes parts of the webrtc::VoEHardware sub API (relanding)
by henrika@webrtc.org
· 10 years ago
28e9b66
Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..."
by henrika@webrtc.org
· 10 years ago
5c6f3fd
Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
8ec46c6
Allow the RTP level indicator computation to work at any sample rate.
by andrew@webrtc.org
· 10 years ago
ba9daa7
Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupported case and we should not maintain such a test.
by henrika@webrtc.org
· 10 years ago
616cbcd
Only clamp to 16 kHz when AECM is enabled.
by andrew@webrtc.org
· 10 years ago
b0079ed
Replace scoped_array<T> with scoped_ptr<T[]>.
by andrew@webrtc.org
· 10 years ago
0061d86
* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.
by wu@webrtc.org
· 10 years ago
110a2d2
Remove ACM1/ACM2 switching from VoiceEngine tests
by henrik.lundin@webrtc.org
· 10 years ago
2e24460
Support arbitrary input/output rates and downmixing in AudioProcessing.
by andrew@webrtc.org
· 10 years ago
8b4811b
Reland "Stop using ACM factory in VoiceEngine"
by henrik.lundin@webrtc.org
· 10 years ago
a19bee3
Revert "Stop using ACM factory in VoiceEngine"
by henrik.lundin@webrtc.org
· 10 years ago
a61127d
Stop using ACM factory in VoiceEngine
by henrik.lundin@webrtc.org
· 10 years ago
13f9d37
Reland "Make VoiceEngine choose ACM2 by default""
by henrik.lundin@webrtc.org
· 10 years ago
cf526f7
Resampler modifications in preparation for arbitrary audioproc rates.
by andrew@webrtc.org
· 10 years ago
514abde
Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
3ea24b2
Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
14c5e8a
Revert "Make VoiceEngine choose ACM2 by default"
by henrik.lundin@webrtc.org
· 10 years ago
4f9c08f
Make VoiceEngine choose ACM2 by default
by henrik.lundin@webrtc.org
· 10 years ago
1857d7e
Re-enable AGC tests:
by aluebs@webrtc.org
· 10 years ago
a738ae3
iOS: baby steps to being able to include_tests=1
by fischman@webrtc.org
· 10 years ago
0b559b6
Moved voe_neteq_stats_unittest to audio_coding_module_unittest
by henrik.lundin@webrtc.org
· 10 years ago
6b1114a
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
by fischman@webrtc.org
· 10 years ago
290c5a5
Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
e338cc2
Added a new OnMoreData() interface which will not feed the playout data to APM.
by xians@webrtc.org
· 10 years ago
7a06daa
(landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds
by henrika@webrtc.org
· 10 years ago
eb90479
Move output_mixer_unittest.cc to utility_unittest.cc.
by andrew@webrtc.org
· 10 years ago
4b0cd7f
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
by andresp@webrtc.org
· 10 years ago
bf4f232
Consolidate audio conversion from Channel and TransmitMixer.
by andrew@webrtc.org
· 10 years ago
c55faad
Remove AudioDevice::{Microphone,Speaker}IsAvailable.
by andrew@webrtc.org
· 10 years ago
0725df6
Extends max sample rate from 96kHz to 192kHz on the input side.
by henrika@webrtc.org
· 10 years ago
5ae01bf
Make ACM2 the default in voe_cmd_test.
by andrew@webrtc.org
· 10 years ago
8f5ab19
VoiceEngine(iOS & Android): removed NOT_SUPPORTED
by fischman@webrtc.org
· 10 years ago
7c3f468
VoE Channel: Don't register codecs when stopping receiver
by henrik.lundin@webrtc.org
· 10 years ago
0027f0a
Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out.
by solenberg@webrtc.org
· 10 years ago
5f804f8
VoE changes to allow forwarding of packets from VoE to ViE BWE.
by solenberg@webrtc.org
· 10 years ago
d327be4
Fix "unreachable code" warnings (MSVC warning 4702) in webrtc.
by pbos@webrtc.org
· 10 years ago
40fee00
Adding operator== and != methods for CodecInst and VideoCodec structures.
by mallinath@webrtc.org
· 10 years ago
55f4fe8
Prevent playout delay wrap-around in VoiceEngine
by henrik.lundin@webrtc.org
· 10 years ago
4d9df07
Removes error printout in voe_cmd_test which was caused by attempts to transmit RTCP packets even if a transport object was not registered.
by henrika@webrtc.org
· 10 years ago
48bbc5a
Resolves TSan v2 warnings in voe_auto_test.
by henrika@webrtc.org
· 10 years ago
5ddb6fe
Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instead of _rtpRtcpModule now.
by braveyao@webrtc.org
· 10 years ago
8a4a39c
Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005.
by wu@webrtc.org
· 10 years ago
fa28e37
Removes VoERTP_RTCP::InsertExtraRTPPacket.
by henrika@webrtc.org
· 10 years ago
8513671
Move the volume quantization workaround from VoE to AGC.
by andrew@webrtc.org
· 10 years ago
c8529ab
Remove obsolete voe_unit_test.
by solenberg@webrtc.org
· 10 years ago
ae50521
Remove external encryption API for VoE.
by solenberg@webrtc.org
· 10 years ago
0fd5775
Remove unused and not working voe_extended_test.
by solenberg@webrtc.org
· 10 years ago
48a5cdb
Reduce mixing threshold in test to avoid flakiness.
by andrew@webrtc.org
· 10 years ago
247df83
Add an interface for accepting keypress signals to AudioProcessing.
by andrew@webrtc.org
· 10 years ago
dd1d6ce
Restore mixing integration tests.
by andrew@webrtc.org
· 10 years ago
1eba384
Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents"
by pbos@webrtc.org
· 10 years ago
4f41016
Fix locking in LoopBackTransport::StorePacket.
by pbos@webrtc.org
· 10 years ago
0a7d406
Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents
by marpan@webrtc.org
· 10 years ago
910910a
Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base
by xians@webrtc.org
· 10 years ago
25bec2a
Move out typing detection to its own class.
by henrikg@webrtc.org
· 10 years ago
4f23307
Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc.
by xians@webrtc.org
· 10 years ago
ccee3c3
Output logs to stderr from voe_cmd_test by default.
by andrew@webrtc.org
· 11 years ago
90c6679
Temporarily disabling some more audio processing tests.
by aluebs@webrtc.org
· 11 years ago
457e101
Minor voice engine improvements around AGC.
by andrew@webrtc.org
· 11 years ago
842d07a
Android: Fixes crash when exiting WebRTCDemo.
by henrike@webrtc.org
· 11 years ago
926e88a
Remove the requirement to call set_sample_rate_hz and friends.
by andrew@webrtc.org
· 11 years ago
9a7cb02
Fix the include guard in transmit_mixer.h
by braveyao@webrtc.org
· 11 years ago
a3ae4d1
Fix the include guard in transmit_mixer.h
by braveyao@webrtc.org
· 11 years ago
acc2e43
Add callbacks for receive channel RTCP statistics.
by sprang@webrtc.org
· 11 years ago
b8dc2e2
Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing.
by turaj@webrtc.org
· 11 years ago
b50a841
Fix jitter buffer delay estimate.
by turaj@webrtc.org
· 11 years ago
7f0519e
Update talk to 58174641 together with http://review.webrtc.org/4319005/.
by wu@webrtc.org
· 11 years ago
2f70422
Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).
by henrike@webrtc.org
· 11 years ago
894dab9
Roll chromium_revision 232627:238260
by kjellander@webrtc.org
· 11 years ago
c8bd975
Allow opening an AEC dump from an existing file handle.
by henrikg@webrtc.org
· 11 years ago
5459e0b
Remove the long disabled WEBRTC_SVNREVISION define.
by andrew@webrtc.org
· 11 years ago
36fb531
Fixes a crash in VoE when unregistering JNI hooks.
by henrike@webrtc.org
· 11 years ago
4bfa866
Add experimental noise suppression dummy API.
by aluebs@webrtc.org
· 11 years ago
8dda8d2
Inject config when creating channels to override the existing one.
by turaj@webrtc.org
· 11 years ago
e9274ae
Fix for making sure that the packet in order checks are done prior to updating the last received packet state.
by stefan@webrtc.org
· 11 years ago
7606f43
Fixing broken tests in voe_auto_test extended
by tina.legrand@webrtc.org
· 11 years ago
2ba95be
Upgrade scoped_ptr to Chromium's latest version.
by andrew@webrtc.org
· 11 years ago
6c0739e
Protect _transportPtr, which can be accessed by different threads in the case of external transport. This change avoid the potential use-after-free, e.g. the case in the reported bug.
by wu@webrtc.org
· 11 years ago
224c0f5
Fix tsan failures in channel.cc regarding to the volume settings.
by wu@webrtc.org
· 11 years ago
1e6493d
Check the number of playout channels instead of the send channels in StopPlayout()
by xians@webrtc.org
· 11 years ago
0f281aa
Roll chromium_revision 226126:228675 and fix clang warnings
by kjellander@webrtc.org
· 11 years ago
9670be6
Add APK and isolate target for video_engine_tests
by kjellander@webrtc.org
· 11 years ago
28ea6f8
Clean up AudioProcessing defaults and errors.
by andrew@webrtc.org
· 11 years ago
3de1b22
Fix include of isolate.gypi
by kjellander@webrtc.org
· 11 years ago
1364cf1
1. adding request of ACM version in the manual mode of voe_auto_test
by minyue@webrtc.org
· 11 years ago
a064105
Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket
by henrika@webrtc.org
· 11 years ago
d24ce00
Remove deprecated AudioCodingModule::Destroy.
by andrew@webrtc.org
· 11 years ago
0580c2c
Enable SetInitialPlayoutDelay on Android.
by dwkang@webrtc.org
· 11 years ago
bee99b1
Small refactoring of AudioProcessing use in channel.cc.
by andrew@webrtc.org
· 11 years ago
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