[NOT FOR REVIEW] Convert channel counts to size_t.

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 346 matching lines @@
                                   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(int32_t id,
                                 int8_t payloadType,
                                 const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                                 int frequency,
-                                uint8_t channels,
+                                size_t channels,
                                 uint32_t rate) override;
     void OnIncomingSSRCChanged(int32_t id, uint32_t ssrc) override;
     void OnIncomingCSRCChanged(int32_t id, uint32_t CSRC, bool added) override;
 
     // From RtpAudioFeedback in the RTP/RTCP module
     void OnPlayTelephoneEvent(int32_t id,
                               uint8_t event,
                               uint16_t lengthMs,
                               uint8_t volume) override;
 
@@ skipping 48 matching lines @@
     {
         return _outputAudioLevel.Level();
     }
     uint32_t Demultiplex(const AudioFrame& audioFrame);
     // Demultiplex the data to the channel's |_audioFrame|. The difference
     // between this method and the overloaded method above is that |audio_data|
     // does not go through transmit_mixer and APM.
     void Demultiplex(const int16_t* audio_data,
                      int sample_rate,
                      size_t number_of_frames,
-                     int number_of_channels);
+                     size_t number_of_channels);
     uint32_t PrepareEncodeAndSend(int mixingFrequency);
     uint32_t EncodeAndSend();
 
     // Associate to a send channel.
     // Used for obtaining RTT for a receive-only channel.
     void set_associate_send_channel(const ChannelOwner& channel) {
       assert(_channelId != channel.channel()->ChannelId());
       CriticalSectionScoped lock(assoc_send_channel_lock_.get());
       associate_send_channel_ = channel;
     }
@@ skipping 131 matching lines @@
     rtc::scoped_ptr<NetworkPredictor> network_predictor_;
     // An associated send channel.
     rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
     ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_