[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h

Index: webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index 4a09bc2b0851c03eb42c7b2d2f9440ea86c55a22..6ab1807f80ddee05e8470c5ac7b7d6d2ecc76a73 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -244,7 +244,7 @@ class AudioCodingModuleImpl : public AudioCodingModule {
     uint32_t input_timestamp;
     const int16_t* audio;
     size_t length_per_channel;
-    uint8_t audio_channel;
+    size_t audio_channel;
     // If a re-mix is required (up or down), this buffer will store a re-mixed
     // version of the input.
     int16_t buffer[WEBRTC_10MS_PCM_AUDIO];
@@ -395,7 +395,7 @@ class AudioCodingImpl : public AudioCoding {
   static bool MapCodecTypeToParameters(int codec_type,
                                        std::string* codec_name,
                                        int* sample_rate_hz,
-                                       int* channels);
+                                       size_t* channels);
 
   int playout_frequency_hz_;
   // TODO(henrik.lundin): All members below this line are temporary and should