Remove VoEAudioProcessing interface.

webrtc/voice_engine/channel.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 12 matching lines @@
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/voice_engine/audio_level.h"
 #include "webrtc/voice_engine/file_player.h"
 #include "webrtc/voice_engine/file_recorder.h"
-#include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/shared_data.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace rtc {
 class TimestampWrapAroundHandler;
 }
 
 namespace webrtc {
@@ skipping 83 matching lines @@
 };
 
 class Channel
     : public RtpData,
       public RtpFeedback,
       public FileCallback,  // receiving notification from file player &
                             // recorder
       public Transport,
       public AudioPacketizationCallback,  // receive encoded packets from the
                                           // ACM
-      public ACMVADCallback,              // receive voice activity from the ACM
       public MixerParticipant,  // supplies output mixer with audio frames
       public OverheadObserver {
  public:
   friend class VoERtcpObserver;
 
   enum { KNumSocketThreads = 1 };
   enum { KNumberOfSocketBuffers = 8 };
   virtual ~Channel();
   static int32_t CreateChannel(
       Channel*& channel,
@@ skipping 110 matching lines @@
   // Audio+Video Sync
   uint32_t GetDelayEstimate() const;
   int SetMinimumPlayoutDelay(int delayMs);
   int GetPlayoutTimestamp(unsigned int& timestamp);
   int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
 
   // DTMF
   int SendTelephoneEventOutband(int event, int duration_ms);
   int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency);
 
-  // VoEAudioProcessingImpl
-  int VoiceActivityIndicator(int& activity);
-
   // VoERTP_RTCP
   int SetLocalSSRC(unsigned int ssrc);
   int GetLocalSSRC(unsigned int& ssrc);
   int GetRemoteSSRC(unsigned int& ssrc);
   int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
   int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
   void EnableSendTransportSequenceNumber(int id);
   void EnableReceiveTransportSequenceNumber(int id);
 
   void RegisterSenderCongestionControlObjects(
@@ skipping 19 matching lines @@
   void SetNACKStatus(bool enable, int maxNumberOfPackets);
 
   // From AudioPacketizationCallback in the ACM
   int32_t SendData(FrameType frameType,
                    uint8_t payloadType,
                    uint32_t timeStamp,
                    const uint8_t* payloadData,
                    size_t payloadSize,
                    const RTPFragmentationHeader* fragmentation) override;
 
-  // From ACMVADCallback in the ACM
-  int32_t InFrameType(FrameType frame_type) override;
-
   // From RtpData in the RTP/RTCP module
   int32_t OnReceivedPayloadData(const uint8_t* payloadData,
                                 size_t payloadSize,
                                 const WebRtcRTPHeader* rtpHeader) override;
   bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
 
   // From RtpFeedback in the RTP/RTCP module
   int32_t OnInitializeDecoder(int8_t payloadType,
                               const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                               int frequency,
@@ skipping 126 matching lines @@
   int _inputFilePlayerId;
   int _outputFilePlayerId;
   int _outputFileRecorderId;
   bool _outputFileRecording;
   uint32_t _timeStamp;
 
   RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
 
   // Timestamp of the audio pulled from NetEq.
   rtc::Optional<uint32_t> jitter_buffer_playout_timestamp_;
+
+  rtc::CriticalSection video_sync_lock_;
   uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_);
   uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_);
   uint16_t send_sequence_number_;
   uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes];
 
   rtc::CriticalSection ts_stats_lock_;
 
   std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
   // The rtp timestamp of the first played out audio frame.
   int64_t capture_start_rtp_time_stamp_;
   // The capture ntp time (in local timebase) of the first played out audio
   // frame.
   int64_t capture_start_ntp_time_ms_ GUARDED_BY(ts_stats_lock_);
 
   // uses
   Statistics* _engineStatisticsPtr;
   OutputMixer* _outputMixerPtr;
   ProcessThread* _moduleProcessThreadPtr;
   AudioDeviceModule* _audioDeviceModulePtr;
   VoiceEngineObserver* _voiceEngineObserverPtr;  // owned by base
   rtc::CriticalSection* _callbackCritSectPtr;    // owned by base
   Transport* _transportPtr;  // WebRtc socket or external transport
   RmsLevel rms_level_;
-  int32_t _sendFrameType;  // Send data is voice, 1-voice, 0-otherwise
   bool input_mute_ GUARDED_BY(volume_settings_critsect_);
   bool previous_frame_muted_;  // Only accessed from PrepareEncodeAndSend().
   float _outputGain GUARDED_BY(volume_settings_critsect_);
   // VoEBase
   bool _mixFileWithMicrophone;
   // VoeRTP_RTCP
   uint32_t _lastLocalTimeStamp;
   int8_t _lastPayloadType;
   bool _includeAudioLevelIndication;
   size_t transport_overhead_per_packet_ GUARDED_BY(overhead_per_packet_lock_);
   size_t rtp_overhead_per_packet_ GUARDED_BY(overhead_per_packet_lock_);
   rtc::CriticalSection overhead_per_packet_lock_;
   // VoENetwork
   AudioFrame::SpeechType _outputSpeechType;
-  // VoEVideoSync
-  rtc::CriticalSection video_sync_lock_;
-  // VoEAudioProcessing
+  // DTX.
   bool restored_packet_in_use_;
   // RtcpBandwidthObserver
   std::unique_ptr<VoERtcpObserver> rtcp_observer_;
   // An associated send channel.
   rtc::CriticalSection assoc_send_channel_lock_;
   ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 
   bool pacing_enabled_;
   PacketRouter* packet_router_ = nullptr;
   std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
   std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
   std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_