[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/voice_engine/transmit_mixer.cc

Index: webrtc/voice_engine/transmit_mixer.cc
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index a02f298509fc521845f8ae602c0886dedaefb620..fd1200c7360c19517aab75b7c41d7e958955d57e 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -295,7 +295,8 @@ TransmitMixer::SetAudioProcessingModule(AudioProcessing* audioProcessingModule)
     return 0;
 }
 
-void TransmitMixer::GetSendCodecInfo(int* max_sample_rate, int* max_channels) {
+void TransmitMixer::GetSendCodecInfo(int* max_sample_rate,
+                                     size_t* max_channels) {
   *max_sample_rate = 8000;
   *max_channels = 1;
   for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid();
@@ -313,7 +314,7 @@ void TransmitMixer::GetSendCodecInfo(int* max_sample_rate, int* max_channels) {
 int32_t
 TransmitMixer::PrepareDemux(const void* audioSamples,
                             size_t nSamples,
-                            uint8_t nChannels,
+                            size_t nChannels,
                             uint32_t samplesPerSec,
                             uint16_t totalDelayMS,
                             int32_t clockDrift,
@@ -322,7 +323,7 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::PrepareDemux(nSamples=%" PRIuS ", "
-                 "nChannels=%u, samplesPerSec=%u, totalDelayMS=%u, "
+                 "nChannels=%" PRIuS ", samplesPerSec=%u, totalDelayMS=%u, "
                  "clockDrift=%d, currentMicLevel=%u)",
                  nSamples, nChannels, samplesPerSec, totalDelayMS, clockDrift,
                  currentMicLevel);
@@ -427,8 +428,8 @@ TransmitMixer::DemuxAndMix()
 }
 
 void TransmitMixer::DemuxAndMix(const int voe_channels[],
-                                int number_of_voe_channels) {
-  for (int i = 0; i < number_of_voe_channels; ++i) {
+                                size_t number_of_voe_channels) {
+  for (size_t i = 0; i < number_of_voe_channels; ++i) {
     voe::ChannelOwner ch = _channelManagerPtr->GetChannel(voe_channels[i]);
     voe::Channel* channel_ptr = ch.channel();
     if (channel_ptr) {
@@ -460,8 +461,8 @@ TransmitMixer::EncodeAndSend()
 }
 
 void TransmitMixer::EncodeAndSend(const int voe_channels[],
-                                  int number_of_voe_channels) {
-  for (int i = 0; i < number_of_voe_channels; ++i) {
+                                  size_t number_of_voe_channels) {
+  for (size_t i = 0; i < number_of_voe_channels; ++i) {
     voe::ChannelOwner ch = _channelManagerPtr->GetChannel(voe_channels[i]);
     voe::Channel* channel_ptr = ch.channel();
     if (channel_ptr && channel_ptr->Sending())
@@ -1131,10 +1132,10 @@ bool TransmitMixer::IsRecordingMic()
 
 void TransmitMixer::GenerateAudioFrame(const int16_t* audio,
                                        size_t samples_per_channel,
-                                       int num_channels,
+                                       size_t num_channels,
                                        int sample_rate_hz) {
   int codec_rate;
-  int num_codec_channels;
+  size_t num_codec_channels;
   GetSendCodecInfo(&codec_rate, &num_codec_channels);
   // TODO(ajm): This currently restricts the sample rate to 32 kHz.
   // See: https://code.google.com/p/webrtc/issues/detail?id=3146