Convert channel counts to size_t.

webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc

 /*
  *  Copyright (c) 2015 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 355 matching lines @@
         play_count_(0),
         rec_count_(0),
         audio_stream_(nullptr) {}
 
   virtual ~MockAudioTransport() {}
 
   MOCK_METHOD10(RecordedDataIsAvailable,
                 int32_t(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));
   MOCK_METHOD8(NeedMorePlayData,
                int32_t(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));
 
   // Set default actions of the mock object. We are delegating to fake
   // implementations (of AudioStreamInterface) here.
   void HandleCallbacks(EventWrapper* test_is_done,
                        AudioStreamInterface* audio_stream,
                        size_t num_callbacks) {
     test_is_done_ = test_is_done;
     audio_stream_ = audio_stream;
     num_callbacks_ = num_callbacks;
     if (play_mode()) {
       ON_CALL(*this, NeedMorePlayData(_, _, _, _, _, _, _, _))
           .WillByDefault(
               Invoke(this, &MockAudioTransport::RealNeedMorePlayData));
     }
     if (rec_mode()) {
       ON_CALL(*this, RecordedDataIsAvailable(_, _, _, _, _, _, _, _, _, _))
           .WillByDefault(
               Invoke(this, &MockAudioTransport::RealRecordedDataIsAvailable));
     }
   }
 
   int32_t RealRecordedDataIsAvailable(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) {
     EXPECT_TRUE(rec_mode()) << "No test is expecting these callbacks.";
     rec_count_++;
     // Process the recorded audio stream if an AudioStreamInterface
     // implementation exists.
     if (audio_stream_) {
       audio_stream_->Write(audioSamples, nSamples);
     }
     if (ReceivedEnoughCallbacks()) {
       if (test_is_done_) {
         test_is_done_->Set();
       }
     }
     return 0;
   }
 
   int32_t RealNeedMorePlayData(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) {
     EXPECT_TRUE(play_mode()) << "No test is expecting these callbacks.";
     play_count_++;
     nSamplesOut = nSamples;
     // Read (possibly processed) audio stream samples to be played out if an
     // AudioStreamInterface implementation exists.
@@ skipping 387 matching lines @@
   StopPlayout();
   StopRecording();
   // Verify that the correct number of transmitted impulses are detected.
   EXPECT_EQ(latency_audio_stream->num_latency_values(),
             static_cast<size_t>(
                 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
   latency_audio_stream->PrintResults();
 }
 
 }  // namespace webrtc