turaj@webrtc.org 1ebd2e96df Remove timestamp retreival warning/error. ...

An error reported while retreiving playout timestamp if no RTP packet received, yet. This causes an overflow of errors/warnings in applications where few channel are created but only one is actively engaged in a conversation. Therefore, we don't find such logging informative (there is no check upon correctness of timestamp computaion only if a packet already received).

BUG=3545
TEST=manual with voe_cmd_test,try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6781 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-07-25 17:50:10 +00:00

..

include

Remove the send-side cname getter APIs from voice and video engine.

2014-07-11 09:55:30 +00:00

test

This is related to an earlier CL of enabling Opus 48 kHz.

2014-07-18 12:28:28 +00:00

Android.mk

Remove external encryption API for VoE.

2014-02-18 11:27:22 +00:00

BUILD.gn

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

channel_manager.cc

Inject config when creating channels to override the existing one.

2013-11-13 00:02:48 +00:00

channel_manager.h

Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.

2014-05-21 21:18:46 +00:00

channel_unittest.cc

Include files from webrtc/.. paths in voice_engine/

2013-05-21 13:52:32 +00:00

channel.cc

Remove timestamp retreival warning/error.

2014-07-25 17:50:10 +00:00

channel.h

Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.

2014-07-16 21:28:26 +00:00

dtmf_inband_queue.cc

Resolves TSan v2 warnings in voe_auto_test.

2014-03-18 10:32:33 +00:00

dtmf_inband_queue.h

Proper spacing for end-of-namespace comments.

2013-07-03 15:12:26 +00:00

dtmf_inband.cc

Switch C++-style C headers with their C equivalents.

2013-08-05 16:22:53 +00:00

dtmf_inband.h

Proper spacing for end-of-namespace comments.

2013-07-03 15:12:26 +00:00

level_indicator.cc

Proper spacing for end-of-namespace comments.

2013-07-03 15:12:26 +00:00

level_indicator.h

Include files from webrtc/.. paths in voice_engine/

2013-05-21 13:52:32 +00:00

monitor_module.cc

Proper spacing for end-of-namespace comments.

2013-07-03 15:12:26 +00:00

monitor_module.h

Proper spacing for end-of-namespace comments.

2013-07-03 15:12:26 +00:00

network_predictor_unittest.cc

Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.

2014-07-16 21:28:26 +00:00

network_predictor.cc

Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.

2014-07-16 21:28:26 +00:00

network_predictor.h

Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.

2014-07-16 21:28:26 +00:00

output_mixer.cc

Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-14 12:04:29 +00:00

output_mixer.h

Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-14 12:04:29 +00:00

OWNERS

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

shared_data.cc

Upgrade scoped_ptr to Chromium's latest version.

2013-10-22 12:50:00 +00:00

shared_data.h

Check the number of playout channels instead of the send channels in StopPlayout()

2013-10-17 16:15:34 +00:00

statistics.cc

Switch C++-style C headers with their C equivalents.

2013-08-05 16:22:53 +00:00

statistics.h

Proper spacing for end-of-namespace comments.

2013-07-03 15:12:26 +00:00

transmit_mixer_unittest.cc

Include files from webrtc/.. paths in voice_engine/

2013-05-21 13:52:32 +00:00

transmit_mixer.cc

Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.

2014-07-11 19:09:59 +00:00

transmit_mixer.h

Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-12 12:24:10 +00:00

utility_unittest.cc

Resampler modifications in preparation for arbitrary audioproc rates.

2014-04-19 00:32:07 +00:00

utility.cc

Fix the chain that propagates the audio frame's rtp and ntp timestamp including:

2014-06-05 20:34:08 +00:00

utility.h

Add a Reset() method to AudioFrame.

2014-05-30 17:28:50 +00:00

voe_audio_processing_impl.cc

Allow opening an AEC dump from an existing file handle.

2013-12-06 16:05:17 +00:00

voe_audio_processing_impl.h

Allow opening an AEC dump from an existing file handle.

2013-12-06 16:05:17 +00:00

voe_audio_processing_unittest.cc

Include files from webrtc/.. paths in voice_engine/

2013-05-21 13:52:32 +00:00

voe_auto_test.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

voe_base_impl.cc

Fix the chain that propagates the audio frame's rtp and ntp timestamp including:

2014-06-05 20:34:08 +00:00

voe_base_impl.h

Fix the chain that propagates the audio frame's rtp and ntp timestamp including:

2014-06-05 20:34:08 +00:00

voe_base_unittest.cc

Include files from webrtc/.. paths in voice_engine/

2013-05-21 13:52:32 +00:00

voe_codec_impl.cc

This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate.

2014-05-28 09:52:06 +00:00

voe_codec_impl.h

This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate.

2014-05-28 09:52:06 +00:00

voe_codec_unittest.cc

Adding operator== and != methods for CodecInst and VideoCodec structures.

2014-03-21 00:41:28 +00:00

voe_dtmf_impl.cc

Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-14 12:04:29 +00:00

voe_dtmf_impl.h

Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-14 12:04:29 +00:00

voe_external_media_impl.cc

Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-12 12:19:19 +00:00

voe_external_media_impl.h

Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-12 12:19:19 +00:00

voe_file_impl.cc

Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-12 12:24:10 +00:00

voe_file_impl.h

Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-12 12:24:10 +00:00

voe_hardware_impl.cc

Removes parts of the webrtc::VoEHardware sub API (relanding)

2014-05-09 11:43:00 +00:00

voe_hardware_impl.h

Removes parts of the webrtc::VoEHardware sub API (relanding)

2014-05-09 11:43:00 +00:00

voe_neteq_stats_impl.cc

VoiceEngine(iOS & Android): removed NOT_SUPPORTED

2014-04-01 00:16:35 +00:00

voe_neteq_stats_impl.h

Update talk to 58174641 together with http://review.webrtc.org/4319005/.

2013-12-13 19:17:43 +00:00

voe_network_impl.cc

VoE changes to allow forwarding of packets from VoE to ViE BWE.

2014-03-24 10:38:25 +00:00

voe_network_impl.h

VoE changes to allow forwarding of packets from VoE to ViE BWE.

2014-03-24 10:38:25 +00:00

voe_rtp_rtcp_impl.cc

Remove the send-side cname getter APIs from voice and video engine.

2014-07-11 09:55:30 +00:00

voe_rtp_rtcp_impl.h

Remove the send-side cname getter APIs from voice and video engine.

2014-07-11 09:55:30 +00:00

voe_video_sync_impl.cc

VoiceEngine(iOS & Android): removed NOT_SUPPORTED

2014-04-01 00:16:35 +00:00

voe_video_sync_impl.h

Update talk to 50918584.

2013-08-15 23:38:54 +00:00

voe_volume_control_impl.cc

Removes parts of the webrtc::VoEVolumeControl sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-14 11:51:45 +00:00

voe_volume_control_impl.h

Removes parts of the webrtc::VoEVolumeControl sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-14 11:51:45 +00:00

voice_engine_defines.h

Consolidate audio conversion from Channel and TransmitMixer.

2014-04-03 21:56:01 +00:00

voice_engine_impl.cc

Reland "Stop using ACM factory in VoiceEngine"

2014-04-22 19:04:34 +00:00

voice_engine_impl.h

Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-04-14 14:12:50 +00:00

voice_engine_unittests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

voice_engine.gyp

Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.

2014-07-16 21:28:26 +00:00