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webrtc
/
src
/
webrtc
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3f685f111a87b5f977815c44b7df69925e6a1b19
/
call
/
audio_send_stream.h
3aa28f0
Add new ANA stats to the old GetStats() to count the number of actions taken by each controller.
by ivoc
· 7 years ago
5b18967
Move optional.h to webrtc/api/
by kwiberg
· 7 years ago
36189cd
Move RtpExtension to api/ directory and config.h/.cc to call/.
by Stefan Holmer
· 7 years ago
ea04cc0
Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
by eladalon
· 7 years ago
d1701d0
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
by zstein
· 7 years ago
76de83e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 8 years ago
bc32410
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 8 years ago
60154fd
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 8 years ago
f26202b
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.
by ossu
· 8 years ago
fe5f71c
Injectable audio encoders: WebRtcVoiceEngine and company
by ossu
· 8 years ago
ee5a316
Added a new echo likelihood stat that reports the maximum value from a previous time period.
by ivoc
· 8 years ago
31212ba
Moved call.h and most of api/call/* into call/
by ossu
· 8 years ago
[Renamed (96%) from api/call/audio_send_stream.h]
017ebe5
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
by aleloi
· 8 years ago
9a1d49f
Expose RtpCodecParameters to VoiceMediaInfo stats.
by hbos
· 8 years ago
ff9d77c
Support multiple timestamp rates for sending DTMF.
by solenberg
· 8 years ago
a1912ef
Fixing config for Audio BWE.
by minyue
· 8 years ago
5206602
Using AudioOption to enable audio network adaptor.
by minyue
· 8 years ago
91c6f34
Clean up logging in AudioSendStream::SetupSendCodec().
by solenberg
· 8 years ago
77b3ea6
Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ )
by terelius
· 8 years ago
b8bff80
Clean up logging in AudioSendStream::SetupSendCodec().
by solenberg
· 8 years ago
dfb640f
Add a placeholder stat for logging the estimated residual echo likelihood.
by ivoc
· 8 years ago
bffc190
Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.
by minyue
· 8 years ago
cf82062
Move webrtc/audio_*.h to webrtc/api/call
by kjellander
· 8 years ago
[Renamed (95%) from audio_send_stream.h]
f29ccf6
Variable audio bitrate.
by mflodman
· 8 years ago
d9cd888
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
by solenberg
· 9 years ago
5a37e3e
Configure VoE NACK through AudioSendStream::Config, for send streams.
by solenberg
· 9 years ago
8f99654
Remove RED support from WebRtcVoiceEngine/MediaChannel
by kwiberg
· 9 years ago
bcf3191
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 9 years ago
97aa5c2
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
by kwiberg
· 9 years ago
8b348aa
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 9 years ago
47a40a3
Remove webrtc/stream.h and unutilized inheritance.
by pbos
· 9 years ago
b670f85
Replace scoped_ptr with unique_ptr everywhere
by kwiberg
· 9 years ago
f2e3315
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
by solenberg
· 9 years ago
b9a65af
Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ )
by solenberg
· 9 years ago
9ef75db
- Clean up unused voice engine DTMF code following removal of VoEDtmf APIs.
by solenberg
· 9 years ago
f95302f
Prepare the AudioSendStream to be hooked up to send-side BWE.
by Stefan Holmer
· 9 years ago
ae4b1f0
- Remove calls to VoEDtmf from WVoE/MC.
by Fredrik Solenberg
· 9 years ago
f1f7cbb
Add comments about the Audio parts of the public Call API being WIP.
by Fredrik Solenberg
· 9 years ago
ffe1ce0
Move some send stream configuration into webrtc::AudioSendStream.
by solenberg
· 9 years ago
1c28f5c
Implement AudioSendStream::GetStats().
by solenberg
· 9 years ago
f707c68
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
by solenberg
· 9 years ago
0c670ff
Align new VoE API with design.
by solenberg
· 9 years ago
d5bdda3
Unify Transport and newapi::Transport interfaces.
by pbos
· 9 years ago
dce9bf3
Add send transports to individual webrtc::Call streams.
by solenberg
· 9 years ago
2fb88e4
Define Stream base classes
by Jelena Marusic
· 9 years ago
ee9b72e
VoE2 API draft
by Fredrik Solenberg
· 10 years ago