Enable setting the maximum bitrate limit in RtpSender.

webrtc/media/base/mediachannel.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
+#include "webrtc/api/rtpparameters.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/base/window.h"
 #include "webrtc/media/base/codec.h"
 #include "webrtc/media/base/constants.h"
@@ skipping 44 matching lines @@
     for (size_t i = 0; i < vals.size(); ++i) {
       if (i > 0) {
         ost << ", ";
       }
       ost << vals[i].ToString();
     }
     ost << "]";
     return ost.str();
 }
 
+template <typename T>
+static T MinPositive(T a, T b) {

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

This would be more clear as the min of rtc::Optionals. 

[skvlad, 2016/03/15 21:18:18]

I'm going to send a separate code review for the change to replace the magic
constants (such as -1) with rtc::Optional in bitrate limiting code everywhere.
Until then, can we keep it like this?

[pthatcher1, 2016/03/16 00:48:42]

sure

+  if (a <= 0)

[Taylor Brandstetter, 2016/03/12 01:57:06]

nit: {}s

[skvlad, 2016/03/15 21:18:18]

Done.

+    return b;
+  if (b <= 0)
+    return a;
+  return std::min(a, b);
+}
+
 // Construction-time settings, passed to
 // MediaControllerInterface::Create, and passed on when creating
 // MediaChannels.
 struct MediaConfig {
   // Set DSCP value on packets. This flag comes from the
   // PeerConnection constraint 'googDscp'.
   bool enable_dscp = false;
 
   // Video-specific config
 
@@ skipping 793 matching lines @@
     ERROR_PLAY_SRTP_ERROR,                // Generic SRTP failure.
     ERROR_PLAY_SRTP_AUTH_FAILED,          // Failed to authenticate packets.
     ERROR_PLAY_SRTP_REPLAY,               // Packet replay detected.
   };
 
   VoiceMediaChannel() {}
   VoiceMediaChannel(const MediaConfig& config) : MediaChannel(config) {}
   virtual ~VoiceMediaChannel() {}
   virtual bool SetSendParameters(const AudioSendParameters& params) = 0;
   virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;
+  virtual webrtc::RTCRtpParameters GetRtpParameters(uint32_t ssrc) = 0;

[Taylor Brandstetter, 2016/03/12 01:57:06]

Again, can probably be const.

[skvlad, 2016/03/15 21:18:18]

Done.

+  virtual bool SetRtpParameters(uint32_t ssrc,
+                                const webrtc::RTCRtpParameters& parameters) = 0;
   // Starts or stops playout of received audio.
   virtual bool SetPlayout(bool playout) = 0;
   // Starts or stops sending (and potentially capture) of local audio.
   virtual bool SetSend(SendFlags flag) = 0;
   // Configure stream for sending.
   virtual bool SetAudioSend(uint32_t ssrc,
                             bool enable,
                             const AudioOptions* options,
                             AudioRenderer* renderer) = 0;
   // Gets current energy levels for all incoming streams.
@@ skipping 52 matching lines @@
     ERROR_PLAY_SRTP_AUTH_FAILED,          // Failed to authenticate packets.
     ERROR_PLAY_SRTP_REPLAY,               // Packet replay detected.
   };
 
   VideoMediaChannel() {}
   VideoMediaChannel(const MediaConfig& config) : MediaChannel(config) {}
   virtual ~VideoMediaChannel() {}
 
   virtual bool SetSendParameters(const VideoSendParameters& params) = 0;
   virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0;
+  virtual webrtc::RTCRtpParameters GetRtpParameters(uint32_t ssrc) = 0;
+  virtual bool SetRtpParameters(uint32_t ssrc,
+                                const webrtc::RTCRtpParameters& parameters) = 0;
   // Gets the currently set codecs/payload types to be used for outgoing media.
   virtual bool GetSendCodec(VideoCodec* send_codec) = 0;
   // Starts or stops transmission (and potentially capture) of local video.
   virtual bool SetSend(bool send) = 0;
   // Configure stream for sending.
   virtual bool SetVideoSend(uint32_t ssrc,
                             bool enable,
                             const VideoOptions* options) = 0;
   // Sets the sink object to be used for the specified stream.
   // If SSRC is 0, the renderer is used for the 'default' stream.
@@ skipping 127 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_