admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc/voice_engine
History
minyue@webrtc.org 4521e2d0bd Adding online bitrate change to voe_cmd_test 
This is to verify a way of changing the bitrate on-the-fly under current WebRTC implementation.

TEST=changing bit rate for different codecs. sound quality changed when bit rate was set successful. catched error when bit rate is invalid for a running codec.

BUG=
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6901 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-14 12:15:27 +00:00
..
include
Adding SetOpusMaxBandwidth in VoE and ACM
2014-08-12 08:13:33 +00:00
test
Adding online bitrate change to voe_cmd_test
2014-08-14 12:15:27 +00:00
Android.mk
…
BUILD.gn
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
channel_manager.cc
…
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
…
channel.cc
Adding SetOpusMaxBandwidth in VoE and ACM
2014-08-12 08:13:33 +00:00
channel.h
Adding SetOpusMaxBandwidth in VoE and ACM
2014-08-12 08:13:33 +00:00
dtmf_inband_queue.cc
…
dtmf_inband_queue.h
…
dtmf_inband.cc
…
dtmf_inband.h
…
level_indicator.cc
…
level_indicator.h
…
monitor_module.cc
…
monitor_module.h
…
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
…
shared_data.h
…
statistics.cc
…
statistics.h
…
transmit_mixer_unittest.cc
…
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
…
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
…
voe_audio_processing_impl.h
…
voe_audio_processing_unittest.cc
…
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
…
voe_codec_impl.cc
Adding SetOpusMaxBandwidth in VoE and ACM
2014-08-12 08:13:33 +00:00
voe_codec_impl.h
Adding SetOpusMaxBandwidth in VoE and ACM
2014-08-12 08:13:33 +00:00
voe_codec_unittest.cc
…
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
…
voe_external_media_impl.h
…
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
…
voe_hardware_impl.h
…
voe_neteq_stats_impl.cc
…
voe_neteq_stats_impl.h
…
voe_network_impl.cc
…
voe_network_impl.h
…
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
…
voe_video_sync_impl.h
…
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
…
voice_engine_impl.cc
…
voice_engine_impl.h
…
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