Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h

 /*
  *  Copyright (c) 2014 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 13 matching lines @@
 class AudioEncoderOpus final : public AudioEncoder {
  public:
   enum ApplicationMode {
     kVoip = 0,
     kAudio = 1,
   };
 
   struct Config {
     bool IsOk() const;
     int frame_size_ms = 20;
-    int num_channels = 1;
+    size_t num_channels = 1;
     int payload_type = 120;
     ApplicationMode application = kVoip;
     int bitrate_bps = 64000;
     bool fec_enabled = false;
     int max_playback_rate_hz = 48000;
     int complexity = kDefaultComplexity;
     bool dtx_enabled = false;
 
    private:
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
     // If we are on Android, iOS and/or ARM, use a lower complexity setting as
     // default, to save encoder complexity.
     static const int kDefaultComplexity = 5;
 #else
     static const int kDefaultComplexity = 9;
 #endif
   };
 
   explicit AudioEncoderOpus(const Config& config);
   explicit AudioEncoderOpus(const CodecInst& codec_inst);
   ~AudioEncoderOpus() override;
 
   size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
-  int NumChannels() const override;
+  size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
 
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
                              rtc::ArrayView<const int16_t> audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
 
   void Reset() override;
@@ skipping 23 matching lines @@
   double packet_loss_rate_;
   std::vector<int16_t> input_buffer_;
   OpusEncInst* inst_;
   uint32_t first_timestamp_in_buffer_;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_