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

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1404 matching lines @@
   std::vector<webrtc::RtpExtension> filtered_extensions =
       FilterRtpExtensions(params.extensions,
                           webrtc::RtpExtension::IsSupportedForAudio, true);
   if (send_rtp_extensions_ != filtered_extensions) {
     send_rtp_extensions_.swap(filtered_extensions);
     for (auto& it : send_streams_) {
       it.second->RecreateAudioSendStream(send_rtp_extensions_);
     }
   }
 
-  if (!SetMaxSendBandwidth(params.max_bandwidth_bps)) {
+  if (!SetMaxSendBitrate(params.max_bitrate_bps)) {
     return false;
   }
   return SetOptions(params.options);
 }
 
 bool WebRtcVoiceMediaChannel::SetRecvParameters(
     const AudioRecvParameters& params) {
   TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetRecvParameters");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetRecvParameters: "
@@ skipping 303 matching lines @@
                    << channel;
       if (engine()->voe()->codec()->SetOpusMaxPlaybackRate(
           channel, send_codec_spec_.opus_max_playback_rate) == -1) {
         LOG_RTCERR2(SetOpusMaxPlaybackRate, channel,
                     send_codec_spec_.opus_max_playback_rate);
         return false;
       }
     }
   }
 
-  if (send_bitrate_setting_) {
+  if (send_bitrate_bps_) {
     SetSendBitrateInternal(send_bitrate_bps_);
   }
 
   // Set the CN payloadtype and the VAD status.
   if (send_codec_spec_.cng_payload_type != -1) {
     // The CN payload type for 8000 Hz clockrate is fixed at 13.
     if (send_codec_spec_.cng_plfreq != 8000) {
       webrtc::PayloadFrequencies cn_freq;
       switch (send_codec_spec_.cng_plfreq) {
         case 16000:
@@ skipping 601 matching lines @@
     }
   }
 
   webrtc::AudioProcessing* ap = engine()->voe()->base()->audio_processing();
   if (ap) {
     ap->set_output_will_be_muted(all_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.";

[the sun, 2016/03/29 11:20:34]

Please add:
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());

[skvlad, 2016/03/30 19:40:45]

Done.

   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()) {
     LOG(LS_INFO) << "The send codec has not been set up yet. "
                  << "The send bitrate setting will be applied later.";
     return true;
   }
 
   // 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)

[Taylor Brandstetter, 2016/03/29 02:26:58]

nit: {}s

[skvlad, 2016/03/30 19:40:44]

Done.

     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
                      << " to bitrate " << bps << " bps.";
         return false;
       }
     }
     return true;
   } else {
     // 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.";
       return false;
     }
     return true;
   }
 }
 
 bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
@@ skipping 117 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE