Wire up x-google-{min,start,max}-bitrate to WebRtcVoiceMediaChannel.

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..eec9340640493e4b90afa6dac79108267062563f 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1616,6 +1616,14 @@ bool WebRtcVoiceMediaChannel::SetSendParameters(
     return false;
   }
 
+  if (params.max_bandwidth_bps >= 0) {
+    // Note that max_bandwidth_bps intentionally takes priority over the
+    // bitrate config for the codec.
+    bitrate_config_.max_bitrate_bps =
+        params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps;
+  }
+  call_->SetBitrateConfig(bitrate_config_);

[minyue-webrtc, 2016/11/30 14:25:25]

can we determine bitrate_config in one util function, assuming
SetCallBitrateConfig, we can repeat some lines like

const AudioCodec* codec = WebRtcVoiceCodecs::GetPreferredCodec(...)

but it may be easier to maintain.

[stefan-webrtc, 2016/11/30 14:52:23]

I don't know. It's probably doable, but it doesn't seem unreasonable to me that
we get the BitrateConfig inside SetSendCodec if a new codec is actually
determined.

Here we use the already stored BitrateConfig and update it if necessary, so it
shouldn't be responsible for figuring out if the codec has changed.

+
   if (!ValidateRtpExtensions(params.extensions)) {
     return false;
   }
@@ -1921,6 +1929,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
 
     send_codec_spec.transport_cc_enabled = HasTransportCc(*codec);
     send_codec_spec.nack_enabled = HasNack(*codec);
+    bitrate_config_ = GetBitrateConfigForCodec(*codec);
 
     // For Opus as the send codec, we are to determine inband FEC, maximum
     // playback rate, and opus internal dtx.
@@ -1987,12 +1996,16 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
     }
   }
 
-  // Apply new settings to all streams.
   if (send_codec_spec_ != send_codec_spec) {
     send_codec_spec_ = std::move(send_codec_spec);
+    // Apply new settings to all streams.
     for (const auto& kv : send_streams_) {
       kv.second->RecreateAudioSendStream(send_codec_spec_);
     }
+  } else {
+    // If the codec isn't changing, set the start bitrate to -1 which means
+    // "unchanged" so that BWE isn't affected.
+    bitrate_config_.start_bitrate_bps = -1;
   }

[the sun, 2016/11/30 13:55:51]

If you move call_->SetBitrateConfig() here instead, do you even need to store
the config in WVoMC?

[stefan-webrtc, 2016/11/30 14:15:30]

It isn't that simple, since params.max_bandwidth_bps should override this, see
line 1622. Maybe if I have SetSendCodec() return a bitrate config that I can use
there? WDYT?

 
   // Check if the transport cc feedback or NACK status has changed on the