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

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 6fe3d86fd98f086fe0622711c7769573d5276a1b..7282616900a6bafe38549853bd0b3b73ea925f8b 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1418,7 +1418,7 @@ bool WebRtcVoiceMediaChannel::SetSendParameters(
     }
   }
 
-  if (!SetMaxSendBandwidth(params.max_bandwidth_bps)) {
+  if (!SetMaxSendBitrate(params.max_bitrate_bps)) {
     return false;
   }
   return SetOptions(params.options);
@@ -1742,7 +1742,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(int channel) {
     }
   }
 
-  if (send_bitrate_setting_) {
+  if (send_bitrate_bps_) {
     SetSendBitrateInternal(send_bitrate_bps_);
   }
 
@@ -2369,17 +2369,14 @@ bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
   return true;
 }
 
-// TODO(minyue): SetMaxSendBandwidth() is subject to be renamed to
-// SetMaxSendBitrate() in future.
-bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) {
+bool WebRtcVoiceMediaChannel::SetMaxSendBitrate(rtc::Optional<int> bps) {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBandwidth.";
   return SetSendBitrateInternal(bps);
 }
 
-bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
+bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(rtc::Optional<int> bps) {
   LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBitrateInternal.";
-
-  send_bitrate_setting_ = true;
   send_bitrate_bps_ = bps;
 
   if (!HasSendCodec()) {
@@ -2391,15 +2388,16 @@ bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
   // Bitrate is auto by default.
   // TODO(bemasc): Fix this so that if SetMaxSendBandwidth(50) is followed by
   // SetMaxSendBandwith(0), the second call removes the previous limit.
-  if (bps <= 0)
+  if (!bps) {
     return true;
+  }
 
   webrtc::CodecInst codec = send_codec_spec_.codec_inst;
   bool is_multi_rate = WebRtcVoiceCodecs::IsCodecMultiRate(codec);
 
   if (is_multi_rate) {
     // If codec is multi-rate then just set the bitrate.
-    codec.rate = bps;
+    codec.rate = *bps;
     for (const auto& ch : send_streams_) {
       if (!SetSendCodec(ch.second->channel(), codec)) {
         LOG(LS_INFO) << "Failed to set codec " << codec.plname
@@ -2412,7 +2410,7 @@ bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
     // If codec is not multi-rate and |bps| is less than the fixed bitrate
     // then fail. If codec is not multi-rate and |bps| exceeds or equal the
     // fixed bitrate then ignore.
-    if (bps < codec.rate) {
+    if (*bps < codec.rate) {
       LOG(LS_INFO) << "Failed to set codec " << codec.plname
                    << " to bitrate " << bps << " bps"
                    << ", requires at least " << codec.rate << " bps.";