Remove voe::OutputMixer and AudioConferenceMixer.

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.
  */
 
 #ifndef VOICE_ENGINE_CHANNEL_H_
 #define VOICE_ENGINE_CHANNEL_H_
 
 #include <memory>
 
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/call/audio_sink.h"
 #include "api/call/transport.h"
 #include "api/optional.h"
 #include "common_audio/resampler/include/push_resampler.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/acm2/codec_manager.h"
 #include "modules/audio_coding/acm2/rent_a_codec.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
-#include "modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "modules/audio_processing/rms_level.h"
 #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "modules/rtp_rtcp/include/rtp_receiver.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "rtc_base/criticalsection.h"
 #include "rtc_base/event.h"
 #include "rtc_base/thread_checker.h"
 #include "voice_engine/audio_level.h"
 #include "voice_engine/include/voe_base.h"
@@ skipping 45 matching lines @@
   uint8_t fraction_lost;
   uint32_t cumulative_num_packets_lost;
   uint32_t extended_highest_sequence_number;
   uint32_t interarrival_jitter;
   uint32_t last_SR_timestamp;
   uint32_t delay_since_last_SR;
 };
 
 namespace voe {
 
-class OutputMixer;
 class RtcEventLogProxy;
 class RtcpRttStatsProxy;
 class RtpPacketSenderProxy;
 class Statistics;
 class TransportFeedbackProxy;
 class TransportSequenceNumberProxy;
 class VoERtcpObserver;
 
 // Helper class to simplify locking scheme for members that are accessed from
 // multiple threads.
@@ skipping 34 matching lines @@
   rtc::CriticalSection lock_;
   State state_;
 };
 
 class Channel
     : public RtpData,
       public RtpFeedback,
       public Transport,
       public AudioPacketizationCallback,  // receive encoded packets 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,
                                int32_t channelId,
                                uint32_t instanceId,
                                const VoEBase::ChannelConfig& config);
   Channel(int32_t channelId,
           uint32_t instanceId,
           const VoEBase::ChannelConfig& config);
   int32_t Init();
   void Terminate();
   int32_t SetEngineInformation(Statistics& engineStatistics,
-                               OutputMixer& outputMixer,
                                ProcessThread& moduleProcessThread,
                                AudioDeviceModule& audioDeviceModule,
                                VoiceEngineObserver* voiceEngineObserver,
                                rtc::CriticalSection* callbackCritSect,
                                rtc::TaskQueue* encoder_queue);
 
   void SetSink(std::unique_ptr<AudioSinkInterface> sink);
 
   // TODO(ossu): Don't use! It's only here to confirm that the decoder factory
   // passed into AudioReceiveStream is the same as the one set when creating the
@@ skipping 100 matching lines @@
                               uint32_t rate) override;
   void OnIncomingSSRCChanged(uint32_t ssrc) override;
   void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override;
 
   // From Transport (called by the RTP/RTCP module)
   bool SendRtp(const uint8_t* data,
                size_t len,
                const PacketOptions& packet_options) override;
   bool SendRtcp(const uint8_t* data, size_t len) override;
 
-  // From MixerParticipant
-  MixerParticipant::AudioFrameInfo GetAudioFrameWithMuted(
-      int32_t id,
-      AudioFrame* audioFrame) override;
-  int32_t NeededFrequency(int32_t id) const override;
-
   // From AudioMixer::Source.
   AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
       int sample_rate_hz,
       AudioFrame* audio_frame);
 
+  int PreferredSampleRate() const;
+
   uint32_t InstanceId() const { return _instanceId; }
   int32_t ChannelId() const { return _channelId; }
   bool Playing() const { return channel_state_.Get().playing; }
   bool Sending() const { return channel_state_.Get().sending; }
   bool ExternalTransport() const {
     rtc::CritScope cs(&_callbackCritSect);
     return _externalTransport;
   }
   RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); }
   int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); }
@@ skipping 119 matching lines @@
 
   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_ RTC_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_ RTC_ACCESS_ON(encoder_queue_);
   bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_);
   bool previous_frame_muted_ RTC_ACCESS_ON(encoder_queue_);
   float _outputGain RTC_GUARDED_BY(volume_settings_critsect_);
   // VoeRTP_RTCP
@@ skipping 32 matching lines @@
 
   bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false;
 
   rtc::TaskQueue* encoder_queue_ = nullptr;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // VOICE_ENGINE_CHANNEL_H_