Remove the type parameter to NetEq::GetAudio

webrtc/modules/audio_coding/neteq/neteq_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 39 matching lines @@
 class RandomVector;
 class SyncBuffer;
 class TimestampScaler;
 struct AccelerateFactory;
 struct DtmfEvent;
 struct ExpandFactory;
 struct PreemptiveExpandFactory;
 
 class NetEqImpl : public webrtc::NetEq {
  public:
+  enum class OutputType {
+    kNormalSpeech,
+    kPLC,
+    kCNG,
+    kPLCCNG,
+    kVadPassive
+  };
+
   // Creates a new NetEqImpl object. The object will assume ownership of all
   // injected dependencies, and will delete them when done.
   NetEqImpl(const NetEq::Config& config,
             BufferLevelFilter* buffer_level_filter,
             DecoderDatabase* decoder_database,
             DelayManager* delay_manager,
             DelayPeakDetector* delay_peak_detector,
             DtmfBuffer* dtmf_buffer,
             DtmfToneGenerator* dtmf_tone_generator,
             PacketBuffer* packet_buffer,
@@ skipping 19 matching lines @@
   // losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq
   // might insert sync-packet when they observe that buffer level of NetEq is
   // 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.
   int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
                        uint32_t receive_timestamp) override;
 
-  int GetAudio(AudioFrame* audio_frame, NetEqOutputType* type) override;
+  int GetAudio(AudioFrame* audio_frame) override;
 
   int RegisterPayloadType(NetEqDecoder codec,
                           const std::string& codec_name,
                           uint8_t rtp_payload_type) override;
 
   int RegisterExternalDecoder(AudioDecoder* decoder,
                               NetEqDecoder codec,
                               const std::string& codec_name,
                               uint8_t rtp_payload_type,
                               int sample_rate_hz) override;
@@ skipping 193 matching lines @@
   int ExtractPackets(size_t required_samples, PacketList* packet_list)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Resets various variables and objects to new values based on the sample rate
   // |fs_hz| and |channels| number audio channels.
   void SetSampleRateAndChannels(int fs_hz, size_t channels)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Returns the output type for the audio produced by the latest call to
   // GetAudio().
-  NetEqOutputType LastOutputType() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+  OutputType LastOutputType() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Updates Expand and Merge.
   virtual void UpdatePlcComponents(int fs_hz, size_t channels)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Creates DecisionLogic object with the mode given by |playout_mode_|.
   virtual void CreateDecisionLogic() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   rtc::CriticalSection crit_sect_;
   const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
@@ skipping 57 matching lines @@
   bool nack_enabled_ GUARDED_BY(crit_sect_);
   AudioFrame::VADActivity last_vad_activity_ GUARDED_BY(crit_sect_) =
       AudioFrame::kVadPassive;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_