Enable audio streams to send padding.

webrtc/modules/rtp_rtcp/source/rtp_sender.h

 /*
  *  Copyright (c) 2012 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 102 matching lines @@
                         uint32_t timestamp,
                         int64_t capture_time_ms,
                         const uint8_t* payload_data,
                         size_t payload_size,
                         const RTPFragmentationHeader* fragmentation,
                         const RTPVideoHeader* rtp_header,
                         uint32_t* transport_frame_id_out);
 
   // RTP header extension
   int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
-  bool IsRtpHeaderExtensionRegistered(RTPExtensionType type);
+  bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) const;
   int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
 
   bool TimeToSendPacket(uint32_t ssrc,
                         uint16_t sequence_number,
                         int64_t capture_time_ms,
                         bool retransmission,
                         int probe_cluster_id);
   size_t TimeToSendPadding(size_t bytes, int probe_cluster_id);
 
   // NACK.
@@ skipping 194 matching lines @@
   OverheadObserver* overhead_observer_;
 
   const bool send_side_bwe_with_overhead_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_