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 ae2518b73af9d74feed8a847d51c4f2f9b60ab9e..4ca83570c4eb71ba809567df5d3f696421c5074e 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -28,6 +28,7 @@
 #include "webrtc/base/common.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/optional_ios.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/trace_event.h"
@@ -1422,7 +1423,7 @@ bool WebRtcVoiceMediaChannel::SetSendParameters(
     }
   }
 
-  if (!SetMaxSendBandwidth(params.max_bandwidth_bps)) {
+  if (!SetMaxSendBitrate(params.max_bitrate_bps)) {
     return false;
   }
   return SetOptions(params.options);
@@ -1746,7 +1747,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(int channel) {
     }
   }
 
-  if (send_bitrate_setting_) {
+  if (send_bitrate_bps_) {
     SetSendBitrateInternal(send_bitrate_bps_);
   }
 
@@ -2368,17 +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) {
   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()) {
@@ -2390,7 +2388,7 @@ 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;
@@ -2398,7 +2396,7 @@ bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
 
   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
@@ -2411,7 +2409,7 @@ 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.";