Variable audio bitrate.

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 469098a32274973cb38189eb8094336920ae113c..dcc9273002394fee2b71f889b91c639a4f5dea44 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1136,6 +1136,13 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
       stream_ = nullptr;
     }
     config_.rtp.extensions = extensions;
+    if (webrtc::field_trial::FindFullName("WebRTC-AdaptAudioBitrate") ==
+        "Enabled") {
+      // TODO(mflodman):Set this properly.
+      config_.min_bitrate_kbps = kOpusMinBitrate;
+      config_.max_bitrate_kbps = kOpusBitrateFb;

[stefan-webrtc, 2016/07/20 10:15:00]

kOpusBitrateFb seems like a bad variable name. Could we rename it or in some way
clarify here what it means?

[minyue-webrtc, 2016/07/21 10:59:19]

is it possible to use kOpusMaxBitrate (although we may not go that high in
practice). or does max_bitrate_kbps make audio compete with video?

also add a comment "WebRTC-AdaptAudioBitrate is currently only available to
Opus"

[mflodman, 2016/07/22 13:50:29]

@minyue:
I didn't want to increase the bitrate more than used today at this stage, rather
focus on the the lower bitrate use case. And also like you said, keep the audio
bitrate and the video bitrate non-overlapping for now. 

Sounds ok for now and then we'll discuss and try this out together?
Done.

[mflodman, 2016/07/22 13:50:29]

@stefan:

There are three different Opus bitrate constants with: NB = Narrow Band, WB =
Wide Band and FB = Full Band. I'm not sure what to name it otherwise and I don't
want to go change that convention in this CL without approval from the audio
team that has chosen this.

OK?

[stefan-webrtc, 2016/07/25 15:06:20]

I see. Probably good enough then, although I would have preferred
kOpusBitrateFullBand.

[mflodman, 2016/07/26 09:50:52]

I agree and I'll bring that up for discussion.

+    }
+
     RTC_DCHECK(!stream_);
     stream_ = call_->CreateAudioSendStream(config_);
     RTC_CHECK(stream_);