Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/voice_engine/transmit_mixer.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.
  */
 
 #include "webrtc/voice_engine/transmit_mixer.h"
 
+#include "webrtc/base/format_macros.h"
 #include "webrtc/modules/utility/interface/audio_frame_operations.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/channel_manager.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"
 #include "webrtc/voice_engine/statistics.h"
 #include "webrtc/voice_engine/utility.h"
@@ skipping 281 matching lines @@
       CodecInst codec;
       channel->GetSendCodec(codec);
       *max_sample_rate = std::max(*max_sample_rate, codec.plfreq);
       *max_channels = std::max(*max_channels, codec.channels);
     }
   }
 }
 
 int32_t
 TransmitMixer::PrepareDemux(const void* audioSamples,
-                            uint32_t nSamples,
+                            size_t nSamples,
                             uint8_t nChannels,
                             uint32_t samplesPerSec,
                             uint16_t totalDelayMS,
                             int32_t clockDrift,
                             uint16_t currentMicLevel,
                             bool keyPressed)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
-                 "TransmitMixer::PrepareDemux(nSamples=%u, nChannels=%u,"
-                 "samplesPerSec=%u, totalDelayMS=%u, clockDrift=%d,"
-                 "currentMicLevel=%u)", nSamples, nChannels, samplesPerSec,
-                 totalDelayMS, clockDrift, currentMicLevel);
+                 "TransmitMixer::PrepareDemux(nSamples=%" PRIuS ", "
+                 "nChannels=%u, samplesPerSec=%u, totalDelayMS=%u, "
+                 "clockDrift=%d, currentMicLevel=%u)",
+                 nSamples, nChannels, samplesPerSec, totalDelayMS, clockDrift,
+                 currentMicLevel);
 
     // --- Resample input audio and create/store the initial audio frame
     GenerateAudioFrame(static_cast<const int16_t*>(audioSamples),
                        nSamples,
                        nChannels,
                        samplesPerSec);
 
     {
       CriticalSectionScoped cs(&_callbackCritSect);
       if (external_preproc_ptr_) {
@@ skipping 784 matching lines @@
     return _fileCallRecording;
 }
 
 bool TransmitMixer::IsRecordingMic()
 {
     CriticalSectionScoped cs(&_critSect);
     return _fileRecording;
 }
 
 void TransmitMixer::GenerateAudioFrame(const int16_t* audio,
-                                       int samples_per_channel,
+                                       size_t samples_per_channel,
                                        int num_channels,
                                        int sample_rate_hz) {
   int codec_rate;
   int 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
   // When 48 kHz is supported natively by AudioProcessing, this will have
   // to be changed to handle 44.1 kHz.
   int max_sample_rate_hz = kAudioProcMaxNativeSampleRateHz;
@@ skipping 40 matching lines @@
     }
 
     return 0;
 }
 
 int32_t TransmitMixer::MixOrReplaceAudioWithFile(
     int mixingFrequency)
 {
   rtc::scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]);
 
-    int fileSamples(0);
+    size_t fileSamples(0);
     {
         CriticalSectionScoped cs(&_critSect);
         if (_filePlayerPtr == NULL)
         {
             WEBRTC_TRACE(kTraceWarning, kTraceVoice,
                          VoEId(_instanceId, -1),
                          "TransmitMixer::MixOrReplaceAudioWithFile()"
                          "fileplayer doesnot exist");
             return -1;
         }
@@ skipping 132 matching lines @@
 void TransmitMixer::EnableStereoChannelSwapping(bool enable) {
   swap_stereo_channels_ = enable;
 }
 
 bool TransmitMixer::IsStereoChannelSwappingEnabled() {
   return swap_stereo_channels_;
 }
 
 }  // namespace voe
 }  // namespace webrtc