[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.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 81 matching lines @@
   for (int i = 0; i < num_csrcs_; ++i) {
     cb_rtp_feedback_->OnIncomingCSRCChanged(id_, current_remote_csrc_[i],
                                             false);
   }
 }
 
 int32_t RtpReceiverImpl::RegisterReceivePayload(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const int8_t payload_type,
     const uint32_t frequency,
-    const uint8_t channels,
+    const size_t channels,
     const uint32_t rate) {
   CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
 
   // TODO(phoglund): Try to streamline handling of the RED codec and some other
   // cases which makes it necessary to keep track of whether we created a
   // payload or not.
   bool created_new_payload = false;
   int32_t result = rtp_payload_registry_->RegisterReceivePayload(
       payload_name, payload_type, frequency, channels, rate,
       &created_new_payload);
@@ skipping 135 matching lines @@
 
 bool RtpReceiverImpl::HaveReceivedFrame() const {
   return last_received_frame_time_ms_ >= 0;
 }
 
 // Implementation note: must not hold critsect when called.
 void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
   bool new_ssrc = false;
   bool re_initialize_decoder = false;
   char payload_name[RTP_PAYLOAD_NAME_SIZE];
-  uint8_t channels = 1;
+  size_t channels = 1;
   uint32_t rate = 0;
 
   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
 
     int8_t last_received_payload_type =
         rtp_payload_registry_->last_received_payload_type();
     if (ssrc_ != rtp_header.ssrc ||
         (last_received_payload_type == -1 && ssrc_ == 0)) {
       // We need the payload_type_ to make the call if the remote SSRC is 0.
@@ skipping 213 matching lines @@
     // implementations might have CSRC 0 as a valid value.
     if (num_csrcs_diff > 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(id_, 0, true);
     } else if (num_csrcs_diff < 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(id_, 0, false);
     }
   }
 }
 
 }  // namespace webrtc