Enable audio streams to send padding.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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 47 matching lines @@
   int32_t DeRegisterSendPayload(int8_t payload_type) override;
 
   int8_t SendPayloadType() const;
 
   // Register RTP header extension.
   int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
                                          uint8_t id) override;
 
   int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) override;
 
+  bool HasBweExtensions() const override;
+
   // Get start timestamp.
   uint32_t StartTimestamp() const override;
 
   // Configure start timestamp, default is a random number.
   void SetStartTimestamp(uint32_t timestamp) override;
 
   uint16_t SequenceNumber() const override;
 
   // Set SequenceNumber, default is a random number.
   void SetSequenceNumber(uint16_t seq) override;
@@ skipping 273 matching lines @@
   PacketLossStats receive_loss_stats_;
 
   // The processed RTT from RtcpRttStats.
   rtc::CriticalSection critical_section_rtt_;
   int64_t rtt_ms_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_