Remove the type parameter to NetEq::GetAudio

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 36 matching lines @@
                            // (positive or negative).
   size_t added_zero_samples;  // Number of zero samples added in "off" mode.
   // Statistics for packet waiting times, i.e., the time between a packet
   // arrives until it is decoded.
   int mean_waiting_time_ms;
   int median_waiting_time_ms;
   int min_waiting_time_ms;
   int max_waiting_time_ms;
 };
 
-enum NetEqOutputType {
-  kOutputNormal,
-  kOutputPLC,
-  kOutputCNG,
-  kOutputPLCtoCNG,
-  kOutputVADPassive
-};
-
 enum NetEqPlayoutMode {
   kPlayoutOn,
   kPlayoutOff,
   kPlayoutFax,
   kPlayoutStreaming
 };
 
 // This is the interface class for NetEq.
 class NetEq {
  public:
@@ skipping 83 matching lines @@
   // decreasing below a certain threshold, defined by the application.
   // Sync-packets should have the same payload type as the last audio payload
   // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
   // can be implied by inserting a sync-packet.
   // Returns kOk on success, kFail on failure.
   virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
                                uint32_t receive_timestamp) = 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_|, |interleaved_|,
-  // |num_channels_|, and |samples_per_channel_| are updated upon success. If
-  // an error is returned, some fields may not have been updated.
-  // The speech type is written to |type|, if |type| is not NULL.
+  // |num_channels_|, |samples_per_channel_|, |speech_type_|, and
+  // |vad_activity_| are updated upon success. If an error is returned, some
+  // fields may not have been updated.
   // Returns kOK on success, or kFail in case of an error.
-  virtual int GetAudio(AudioFrame* audio_frame, NetEqOutputType* type) = 0;
+  virtual int GetAudio(AudioFrame* audio_frame) = 0;
 
   // Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the
   // information in the codec database. Returns 0 on success, -1 on failure.
   // The name is only used to provide information back to the caller about the
   // decoders. Hence, the name is arbitrary, and may be empty.
   virtual int RegisterPayloadType(NetEqDecoder codec,
                                   const std::string& codec_name,
                                   uint8_t rtp_payload_type) = 0;
 
   // Provides an externally created decoder object |decoder| to insert in the
@@ skipping 113 matching lines @@
 
  protected:
   NetEq() {}
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_