Handle padded audio packets correctly

webrtc/modules/audio_coding/neteq/include/neteq.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 127 matching lines @@
   virtual ~NetEq() {}
 
   // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
   // of the time when the packet was received, and should be measured with
   // the same tick rate as the RTP timestamp of the current payload.
   // Returns 0 on success, -1 on failure.
   virtual int InsertPacket(const RTPHeader& rtp_header,
                            rtc::ArrayView<const uint8_t> payload,
                            uint32_t receive_timestamp) = 0;
 
+  // Lets NetEq know that a packet arrived with an empty payload. This typically
+  // happens when empty packets are used for probing the network channel, and
+  // these packets use RTP sequence numbers from the same series as the actual
+  // audio packets.
+  virtual void InsertEmptyPacket(const RTPHeader& rtp_header) = 0;
+
   // Instructs NetEq to deliver 10 ms of audio data. The data is written to
   // |audio_frame|. All data in |audio_frame| is wiped; |data_|, |speech_type_|,
   // |num_channels_|, |sample_rate_hz_|, |samples_per_channel_|, and
   // |vad_activity_| are updated upon success. If an error is returned, some
   // fields may not have been updated, or may contain inconsistent values.
   // If muted state is enabled (through Config::enable_muted_state), |muted|
   // may be set to true after a prolonged expand period. When this happens, the
   // |data_| in |audio_frame| is not written, but should be interpreted as being
   // all zeros.
   // Returns kOK on success, or kFail in case of an error.
@@ skipping 158 matching lines @@
 
  protected:
   NetEq() {}
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_