Update audio code to use size_t more correctly, talk/ portion.

talk/media/webrtc/webrtcvoiceengine.cc

Index: talk/media/webrtc/webrtcvoiceengine.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 284afed4d119e8b11dce8aa8d0cd0153d5d4e0e2..ab54deb80152c743775c5aee19cbd0c2e2d616eb 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1569,7 +1569,7 @@ bool WebRtcVoiceEngine::UnregisterProcessor(
 void WebRtcVoiceEngine::Process(int channel,
                                 webrtc::ProcessingTypes type,
                                 int16_t audio10ms[],
-                                int length,
+                                size_t length,
                                 int sampling_freq,
                                 bool is_stereo) {
     rtc::CritScope cs(&signal_media_critical_);
@@ -1666,7 +1666,7 @@ class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
               int bits_per_sample,
               int sample_rate,
               int number_of_channels,
-              int number_of_frames) override {
+              size_t number_of_frames) override {
     voe_audio_transport_->OnData(channel_,
                                  audio_data,
                                  bits_per_sample,