Add overhead to audio bwe min, max.

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 96f9ed70d403c6c464cc9b015f83190630a0b331..17175c4e14cb741c543a3829aa40b8543add6af6 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1420,8 +1420,20 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
         "Enabled") {
       // TODO(mflodman): Keep testing this and set proper values.
       // Note: This is an early experiment currently only supported by Opus.
-      config_.min_bitrate_bps = kOpusMinBitrateBps;
-      config_.max_bitrate_bps = kOpusBitrateFbBps;
+      if (webrtc::field_trial::FindFullName(
+              "WebRTC-SendSideBwe-WithOverhead") == "Enabled") {
+        // OverheadPerPacket(58) = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
+        //     + RTP_ext(8);
+        // kMaxOverheadBps = OverheadPerPacket * 8 * 1000ms / frame_length(20ms)

[minyue-webrtc, 2016/11/24 10:42:10]

use 10ms I think due to
https://cs.chromium.org/chromium/src/third_party/webrtc/media/engine/webrtcvo...

[michaelt, 2016/11/24 10:51:24]

I think we should not us 10ms frames since they are not used in ANA. Setting
here 10ms as min frame length would lead to a higher max encoding rate which
would be a waste of bandwidth. 

[minyue-webrtc, 2016/11/28 15:27:06]

It does not sound convincing to me. Are we only defining a lower/upper bounds
for the overall bitrate here. Or are you saying that the limit is a function of
frame length?

+        // kMinOverheadBps = OverheadPerPacket * 8 * 1000ms / frame_length(60ms)
+        constexpr int kMaxOverheadBps = 23200;
+        constexpr int kMinOverheadBps = 7733;

[stefan-webrtc, 2016/11/24 10:48:27]

This will be hard to maintain. can't we compute them instead based on
https://cs.chromium.org/chromium/src/third_party/webrtc/media/engine/webrtcvo...

[minyue-webrtc, 2016/11/28 15:27:06]

It looks like that the problem is that the min frame length in
webrtcvoiceengine.cc is one thing and the min frame length that can ever happen
in audio network adaptation is another.

But since we are defining lower/upper bound now, it is supposed to work if go
choose global min and max (Opus min and max sounds good to me, too), unless we
made error somewhere.

[stefan-webrtc, 2016/12/22 09:06:25]

I still don't really follow. How are they different? I think that must be made
clear here, otherwise it will be hard for others to change this code.

Can the adaptive configuration override the frame lengths configured here:
https://cs.chromium.org/chromium/src/third_party/webrtc/media/engine/webrtcvo...
?

Seems like we should be restricted by what frame lengths are negotiated?

+        config_.min_bitrate_bps = kOpusMinBitrateBps + kMinOverheadBps;
+        config_.max_bitrate_bps = kOpusBitrateFbBps + kMaxOverheadBps;
+      } else {
+        config_.min_bitrate_bps = kOpusMinBitrateBps;
+        config_.max_bitrate_bps = kOpusBitrateFbBps;
+      }
     }
     stream_ = call_->CreateAudioSendStream(config_);
     RTC_CHECK(stream_);