This is a first CL wiring up AudioSendStream to BitrateAllocator. This
is still experimental and there is a test added for the audio only case,
combined audio video variable bitrate test cases will be added as a
follow up.
BUG=5079
Review-Url: https://codereview.webrtc.org/2165743003
Cr-Commit-Position: refs/heads/master@{#13527}
Removes the need to use VoEVolume::SetInputMute()/GetInputMute().
BUG=webrtc:4690
NOTRY=true
Review-Url: https://codereview.webrtc.org/2066973002
Cr-Commit-Position: refs/heads/master@{#13172}
Any file that uses the RTC_DISALLOW_* macros should #include
"webrtc/base/constructormagic.h", but a shocking number of them don't.
This causes trouble when we try to wean files off of #including
scoped_ptr.h, since a bunch of files get their constructormagic macros
only from there.
Rather than fixing these errors one by one as they turn up, this CL
simply ensures that every file in the WebRTC tree that uses the
RTC_DISALLOW_* macros #includes "webrtc/base/constructormagic.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1917043005
Cr-Commit-Position: refs/heads/master@{#12509}
- Use better types in AudioSendStream::SendTelephoneEvent() and related methods.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1782053002
Cr-Commit-Position: refs/heads/master@{#11953}
- Use better types in AudioSendStream::SendTelephoneEvent() and related methods.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1722253002
Cr-Commit-Position: refs/heads/master@{#11927}
This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.
BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1479023002 .
Cr-Commit-Position: refs/heads/master@{#10909}
- Flatten logic and make the relevant calls on VoE::Channel from AudioSendStream::SendTelephoneEvent().
- Store current payload type for telephone events in WVoMC, instead of setting it on the Channel. This should be refactored to be an AudioSendStream::Config parameter when we redo WVoMC::SetSendCodecs().
BUG=webrtc:4690
R=pthatcher@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1491743004 .
Cr-Commit-Position: refs/heads/master@{#10895}
This will allow Audio[Send|Receive]Stream to bypass the VoE interfaces in many cases and talk directly to the channel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1459083007
Cr-Commit-Position: refs/heads/master@{#10788}
AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1397123003
Cr-Commit-Position: refs/heads/master@{#10307}
