Move audio frame memory handling inside AudioMixer.

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 361 matching lines @@
                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::AudioFrameWithInfo GetAudioFrameWithInfo(
-      int sample_rate_hz);
+  AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
+      int sample_rate_hz,
+      AudioFrame* audio_frame);
 
   // From FileCallback
   void PlayNotification(int32_t id, uint32_t durationMs) override;
   void RecordNotification(int32_t id, uint32_t durationMs) override;
   void PlayFileEnded(int32_t id) override;
   void RecordFileEnded(int32_t id) override;
 
   uint32_t InstanceId() const { return _instanceId; }
   int32_t ChannelId() const { return _channelId; }
   bool Playing() const { return channel_state_.Get().playing; }
@@ skipping 74 matching lines @@
   std::unique_ptr<RtpReceiver> rtp_receiver_;
   TelephoneEventHandler* telephone_event_handler_;
   std::unique_ptr<RtpRtcp> _rtpRtcpModule;
   std::unique_ptr<AudioCodingModule> audio_coding_;
   acm2::CodecManager codec_manager_;
   acm2::RentACodec rent_a_codec_;
   std::unique_ptr<AudioSinkInterface> audio_sink_;
   AudioLevel _outputAudioLevel;
   bool _externalTransport;
   AudioFrame _audioFrame;
-  AudioFrame mix_audio_frame_;
   // Downsamples to the codec rate if necessary.
   PushResampler<int16_t> input_resampler_;
   std::unique_ptr<FilePlayer> input_file_player_;
   std::unique_ptr<FilePlayer> output_file_player_;
   std::unique_ptr<FileRecorder> output_file_recorder_;
   int _inputFilePlayerId;
   int _outputFilePlayerId;
   int _outputFileRecorderId;
   bool _outputFileRecording;
   bool _outputExternalMedia;
@@ skipping 65 matching lines @@
   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_