Remove voe::OutputMixer and AudioConferenceMixer.

test/mock_voe_channel_proxy.h

 /*
  *  Copyright (c) 2015 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 59 matching lines @@
   MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived& packet));
   MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
   MOCK_CONST_METHOD0(GetAudioDecoderFactory,
                      const rtc::scoped_refptr<AudioDecoderFactory>&());
   MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
   MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log));
   MOCK_METHOD1(SetRtcpRttStats, void(RtcpRttStats* rtcp_rtt_stats));
   MOCK_METHOD2(GetAudioFrameWithInfo,
       AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
                                          AudioFrame* audio_frame));
-  MOCK_CONST_METHOD0(NeededFrequency, int());
+  MOCK_CONST_METHOD0(PreferredSampleRate, int());
   MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet));
   MOCK_METHOD1(AssociateSendChannel,
                void(const ChannelProxy& send_channel_proxy));
   MOCK_METHOD0(DisassociateSendChannel, void());
   MOCK_CONST_METHOD2(GetRtpRtcp, void(RtpRtcp** rtp_rtcp,
                                       RtpReceiver** rtp_receiver));
   MOCK_CONST_METHOD0(GetPlayoutTimestamp, uint32_t());
   MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms));
   MOCK_CONST_METHOD1(GetRecCodec, bool(CodecInst* codec_inst));
   MOCK_METHOD1(SetReceiveCodecs,
                void(const std::map<int, SdpAudioFormat>& codecs));
   MOCK_METHOD1(OnTwccBasedUplinkPacketLossRate, void(float packet_loss_rate));
   MOCK_METHOD1(OnRecoverableUplinkPacketLossRate,
                void(float recoverable_packet_loss_rate));
   MOCK_CONST_METHOD0(GetSources, std::vector<RtpSource>());
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // TEST_MOCK_VOE_CHANNEL_PROXY_H_