Enable setting the maximum bitrate limit in RtpSender.

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 2cd19e7435e884ceefe2c801cd8070b12977948c..236d25bc497ace9098119b5afa42c568b6668b19 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1138,13 +1138,16 @@ int WebRtcVoiceEngine::CreateVoEChannel() {
 class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     : public AudioRenderer::Sink {
  public:
-  WebRtcAudioSendStream(int ch, webrtc::AudioTransport* voe_audio_transport,
-                        uint32_t ssrc, const std::string& c_name,
+  WebRtcAudioSendStream(int ch,
+                        webrtc::AudioTransport* voe_audio_transport,
+                        uint32_t ssrc,
+                        const std::string& c_name,
                         const std::vector<webrtc::RtpExtension>& extensions,
                         webrtc::Call* call)
       : voe_audio_transport_(voe_audio_transport),
         call_(call),
-        config_(nullptr) {
+        config_(nullptr),
+        rtp_parameters_(webrtc::RTCRtpParameters::CreateDefault()) {
     RTC_DCHECK_GE(ch, 0);
     // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
     // RTC_DCHECK(voe_audio_transport);
@@ -1247,6 +1250,12 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     return config_.voe_channel_id;
   }
 
+  webrtc::RTCRtpParameters rtp_parameters() const { return rtp_parameters_; }
+
+  void set_rtp_parameters(const webrtc::RTCRtpParameters& parameters) {
+    rtp_parameters_ = parameters;
+  }
+
  private:
   rtc::ThreadChecker worker_thread_checker_;
   rtc::ThreadChecker audio_capture_thread_checker_;
@@ -1261,6 +1270,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
   // PeerConnection will make sure invalidating the pointer before the object
   // goes away.
   AudioRenderer* renderer_ = nullptr;
+  webrtc::RTCRtpParameters rtp_parameters_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
 };
@@ -1429,6 +1439,40 @@ bool WebRtcVoiceMediaChannel::SetRecvParameters(
   return true;
 }
 
+webrtc::RTCRtpParameters WebRtcVoiceMediaChannel::GetRtpParameters(

[pthatcher1, 2016/03/12 01:21:04]

Can you split this into a CL for video and a second CL for audio?  I think we'll
be able to land the video one faster that way, and it's more important.

+    uint32_t ssrc) {
+  auto it = send_streams_.find(ssrc);
+  if (it == send_streams_.end()) {
+    LOG(LS_WARNING) << "Attempting to get RTP parameters for stream with ssrc "
+                    << ssrc << " which doesn't exist.";
+    return webrtc::RTCRtpParameters();
+  }
+  return it->second->rtp_parameters();
+}
+
+bool WebRtcVoiceMediaChannel::SetRtpParameters(
+    uint32_t ssrc,
+    const webrtc::RTCRtpParameters& parameters) {
+  auto it = send_streams_.find(ssrc);
+  if (it == send_streams_.end()) {
+    LOG(LS_WARNING) << "Trying to set RTP parameters for stream with ssrc "
+                    << ssrc << " which doesn't exist.";
+    return false;
+  }
+
+  if (parameters.encodings.size() != 1) {
+    LOG(LS_WARNING)
+        << "Attempting to set RTP parameters without exactly 1 encoding";
+    return false;
+  }
+  if (ApplyBitrateLimits(it->second->channel(), send_bitrate_bps_,
+                         parameters.encodings[0].max_bitrate_bps)) {
+    it->second->set_rtp_parameters(parameters);
+    return true;
+  }

[pthatcher1, 2016/03/12 01:21:04]

It feels like we should move the ApplyBitrateLimits to the send stream, perhaps
calls it SetBitrate, and pass in the already calculated min.  Actually, we
should probably mirror the video send stream and have SetParameters that does
the min calculation internal to the stream.

+  return false;
+}
+
 bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "Setting voice channel options: "
@@ -2356,15 +2400,39 @@ bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) {
 bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
   LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBitrateInternal.";
 
+  for (const auto& ch : send_streams_) {
+    if (!ApplyBitrateLimits(
+            ch.second->channel(), bps,
+            ch.second->rtp_parameters().encodings[0].max_bitrate_bps)) {
+      LOG(LS_INFO) << "Failed to limit max bitrate of a stream";
+      return false;
+    }
+  }
+
+  if (send_streams_.size() == 0) {
+    LOG(LS_ERROR) << "send_streams_ empty!";
+  }
+
   send_bitrate_setting_ = true;
   send_bitrate_bps_ = bps;
+  return true;
+}
 
+bool WebRtcVoiceMediaChannel::ApplyBitrateLimits(int channel,
+                                                 int global_limit,
+                                                 int local_limit) {

[pthatcher1, 2016/03/12 01:21:04]

Why not just pass in one value and do the min() before?

   if (!send_codec_) {
     LOG(LS_INFO) << "The send codec has not been set up yet. "
                  << "The send bitrate setting will be applied later.";
     return true;
   }
 
+  int bps = MinPositive(global_limit, local_limit);
+
+  LOG(LS_INFO) << "Applying bandwidth limit to channel " << channel
+               << ": global_limit=" << global_limit
+               << " local_limit=" << local_limit << " result=" << 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.
@@ -2377,12 +2445,10 @@ bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
   if (is_multi_rate) {
     // If codec is multi-rate then just set the bitrate.
     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;
-      }
+    if (!SetSendCodec(channel, codec)) {
+      LOG(LS_INFO) << "Failed to set codec " << codec.plname << " to bitrate "
+                   << bps << " bps.";
+      return false;
     }
     return true;
   } else {