Updated structures and functions for setting the max bitrate limit to take rtc::Optional<int>

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 0ccc64908dc77539a2f7139253e78d8adccfe9a7..9d93b159c8f7382a0e23c85c6ee20659584f907d 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -217,7 +217,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   bool SetSendCodecs(int channel);
   void SetNack(int channel, bool nack_enabled);
   bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
-  bool SetMaxSendBandwidth(int bps);
+  bool SetMaxSendBitrate(rtc::Optional<int> bps);
   bool SetLocalSource(uint32_t ssrc, AudioSource* source);
   bool MuteStream(uint32_t ssrc, bool mute);
 
@@ -231,7 +231,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   bool IsDefaultRecvStream(uint32_t ssrc) {
     return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
   }
-  bool SetSendBitrateInternal(int bps);
+  bool SetSendBitrateInternal(rtc::Optional<int> bps);
   bool HasSendCodec() const {
     return send_codec_spec_.codec_inst.pltype != -1;
   }
@@ -240,8 +240,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
 
   WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> recv_codecs_;
-  bool send_bitrate_setting_ = false;
-  int send_bitrate_bps_ = 0;
+  rtc::Optional<int> send_bitrate_bps_;
   AudioOptions options_;
   rtc::Optional<int> dtmf_payload_type_;
   bool desired_playout_ = false;