Convert channel counts to size_t.

webrtc/modules/audio_device/test/func_test_manager.cc

 /*
  *  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 176 matching lines @@
          iter != _audioList.end(); ++iter) {
             delete *iter;
     }
     _audioList.clear();
 }
 
 int32_t AudioTransportImpl::RecordedDataIsAvailable(
     const void* audioSamples,
     const size_t nSamples,
     const size_t nBytesPerSample,
-    const uint8_t nChannels,
+    const size_t nChannels,
     const uint32_t samplesPerSec,
     const uint32_t totalDelayMS,
     const int32_t clockDrift,
     const uint32_t currentMicLevel,
     const bool keyPressed,
     uint32_t& newMicLevel)
 {
     if (_fullDuplex && _audioList.size() < 15)
     {
         AudioPacket* packet = new AudioPacket();
@@ skipping 124 matching lines @@
         }
     }
 
     return 0;
 }
 
 
 int32_t AudioTransportImpl::NeedMorePlayData(
     const size_t nSamples,
     const size_t nBytesPerSample,
-    const uint8_t nChannels,
+    const size_t nChannels,
     const uint32_t samplesPerSec,
     void* audioSamples,
     size_t& nSamplesOut,
     int64_t* elapsed_time_ms,
     int64_t* ntp_time_ms)
 {
     if (_fullDuplex)
     {
         if (_audioList.empty())
         {
             // use zero stuffing when not enough data
             memset(audioSamples, 0, nBytesPerSample * nSamples);
         } else
         {
             AudioPacket* packet = _audioList.front();
             _audioList.pop_front();
             if (packet)
             {
                 int ret(0);
                 size_t lenOut(0);
                 int16_t tmpBuf_96kHz[80 * 12];
                 int16_t* ptr16In = NULL;
                 int16_t* ptr16Out = NULL;
 
                 const size_t nSamplesIn = packet->nSamples;
-                const uint8_t nChannelsIn = packet->nChannels;
+                const size_t nChannelsIn = packet->nChannels;
                 const uint32_t samplesPerSecIn = packet->samplesPerSec;
                 const size_t nBytesPerSampleIn = packet->nBytesPerSample;
 
                 int32_t fsInHz(samplesPerSecIn);
                 int32_t fsOutHz(samplesPerSec);
 
                 if (fsInHz == 44100)
                     fsInHz = 44000;
 
                 if (fsOutHz == 44100)
@@ skipping 2359 matching lines @@
 
     TEST_LOG("\n");
     PRINT_TEST_RESULTS;
 
     return 0;
 }
 
 }  // namespace webrtc
 
 // EOF