Cap the send bitrate for opus and iSAC before passing down to VoE.

webrtc/media/engine/webrtcvoiceengine.h

Index: webrtc/media/engine/webrtcvoiceengine.h
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index 4c53ad0222a41c4fee11190fe170cff23cc24406..87a9af4ae3bb697934f03c6ae7a635b14faf29c1 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -227,10 +227,10 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   bool IsDefaultRecvStream(uint32_t ssrc) {
     return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
   }
-  bool SetSendBitrate(int bps);
+  bool SetMaxSendBitrate(int bps);
   bool SetChannelParameters(int channel,
                             const webrtc::RtpParameters& parameters);
-  bool SetSendBitrate(int channel, int bps);
+  bool SetMaxSendBitrate(int channel, int bps);
   bool HasSendCodec() const {
     return send_codec_spec_.codec_inst.pltype != -1;
   }
@@ -242,7 +242,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> send_codecs_;
   std::vector<AudioCodec> recv_codecs_;
-  int send_bitrate_bps_ = 0;
+  int max_send_bitrate_bps_ = 0;
   AudioOptions options_;
   rtc::Optional<int> dtmf_payload_type_;
   bool desired_playout_ = false;