Convert channel counts to size_t.

webrtc/modules/audio_coding/acm2/acm_receiver.h

 /*
  *  Copyright (c) 2013 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 26 matching lines @@
 
 namespace acm2 {
 
 class AcmReceiver {
  public:
   struct Decoder {
     int acm_codec_id;
     uint8_t payload_type;
     // This field is meaningful for codecs where both mono and
     // stereo versions are registered under the same ID.
-    int channels;
+    size_t channels;
     int sample_rate_hz;
   };
 
   // Constructor of the class
   explicit AcmReceiver(const AudioCodingModule::Config& config);
 
   // Destructor of the class.
   ~AcmReceiver();
 
   //
@@ skipping 51 matching lines @@
   //                           used for built-in decoders if NetEq doesn't have
   //                           all the info it needs to construct them properly
   //                           (e.g. iSAC, where the decoder needs to be paired
   //                           with an encoder).
   //
   // Return value             : 0 if OK.
   //                           <0 if NetEq returned an error.
   //
   int AddCodec(int acm_codec_id,
                uint8_t payload_type,
-               int channels,
+               size_t channels,
                int sample_rate_hz,
                AudioDecoder* audio_decoder,
                const std::string& name);
 
   //
   // Sets a minimum delay for packet buffer. The given delay is maintained,
   // unless channel condition dictates a higher delay.
   //
   // Input:
   //   - delay_ms             : minimum delay in milliseconds.
@@ skipping 168 matching lines @@
   Clock* clock_;  // TODO(henrik.lundin) Make const if possible.
   bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
   rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);
 };
 
 }  // namespace acm2
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_