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

webrtc/voice_engine/channel.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 3318 matching lines @@
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::Demultiplex()");
     _audioFrame.CopyFrom(audioFrame);
     _audioFrame.id_ = _channelId;
     return 0;
 }
 
 void Channel::Demultiplex(const int16_t* audio_data,
                           int sample_rate,
-                          int number_of_frames,
+                          size_t number_of_frames,
                           int number_of_channels) {
   CodecInst codec;
   GetSendCodec(codec);
 
   if (!mono_recording_audio_.get()) {
     // Temporary space for DownConvertToCodecFormat.
     mono_recording_audio_.reset(new int16_t[kMaxMonoDataSizeSamples]);
   }
   DownConvertToCodecFormat(audio_data,
                            number_of_frames,
@@ skipping 41 matching lines @@
                (int16_t*)_audioFrame.data_,
                 _audioFrame.samples_per_channel_,
                 _audioFrame.sample_rate_hz_,
                 isStereo);
         }
     }
 
     InsertInbandDtmfTone();
 
     if (_includeAudioLevelIndication) {
-      int length = _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
+      size_t length =
+          _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
       if (is_muted) {
         rms_level_.ProcessMuted(length);
       } else {
         rms_level_.Process(_audioFrame.data_, length);
       }
     }
 
     return 0;
 }
 
@@ skipping 267 matching lines @@
     *rtp_receiver = rtp_receiver_.get();
     return 0;
 }
 
 // TODO(andrew): refactor Mix functions here and in transmit_mixer.cc to use
 // a shared helper.
 int32_t
 Channel::MixOrReplaceAudioWithFile(int mixingFrequency)
 {
   rtc::scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]);
-    int fileSamples(0);
+    size_t fileSamples(0);
 
     {
         CriticalSectionScoped cs(&_fileCritSect);
 
         if (_inputFilePlayerPtr == NULL)
         {
             WEBRTC_TRACE(kTraceWarning, kTraceVoice,
                          VoEId(_instanceId, _channelId),
                          "Channel::MixOrReplaceAudioWithFile() fileplayer"
                              " doesnt exist");
@@ skipping 49 matching lines @@
     return 0;
 }
 
 int32_t
 Channel::MixAudioWithFile(AudioFrame& audioFrame,
                           int mixingFrequency)
 {
     assert(mixingFrequency <= 48000);
 
     rtc::scoped_ptr<int16_t[]> fileBuffer(new int16_t[960]);
-    int fileSamples(0);
+    size_t fileSamples(0);
 
     {
         CriticalSectionScoped cs(&_fileCritSect);
 
         if (_outputFilePlayerPtr == NULL)
         {
             WEBRTC_TRACE(kTraceWarning, kTraceVoice,
                          VoEId(_instanceId, _channelId),
                          "Channel::MixAudioWithFile() file mixing failed");
             return -1;
@@ skipping 17 matching lines @@
         // TODO(xians): Change the code when FilePlayer supports real stereo.
         MixWithSat(audioFrame.data_,
                    audioFrame.num_channels_,
                    fileBuffer.get(),
                    1,
                    fileSamples);
     }
     else
     {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
-            "Channel::MixAudioWithFile() samples_per_channel_(%d) != "
-            "fileSamples(%d)",
+            "Channel::MixAudioWithFile() samples_per_channel_(%" PRIuS ") != "
+            "fileSamples(%" PRIuS ")",
             audioFrame.samples_per_channel_, fileSamples);
         return -1;
     }
 
     return 0;
 }
 
 int
 Channel::InsertInbandDtmfTone()
 {
@@ skipping 39 matching lines @@
         // Get 10ms tone segment and set time since last tone to zero
         if (_inbandDtmfGenerator.Get10msTone(toneBuffer, toneSamples) == -1)
         {
             WEBRTC_TRACE(kTraceWarning, kTraceVoice,
                        VoEId(_instanceId, _channelId),
                        "Channel::EncodeAndSend() inserting Dtmf failed");
             return -1;
         }
 
         // Replace mixed audio with DTMF tone.
-        for (int sample = 0;
+        for (size_t sample = 0;
             sample < _audioFrame.samples_per_channel_;
             sample++)
         {
             for (int channel = 0;
                 channel < _audioFrame.num_channels_;
                 channel++)
             {
-                const int index = sample * _audioFrame.num_channels_ + channel;
+                const size_t index =
+                    sample * _audioFrame.num_channels_ + channel;
                 _audioFrame.data_[index] = toneBuffer[sample];
             }
         }
 
         assert(_audioFrame.samples_per_channel_ == toneSamples);
     } else
     {
         // Add 10ms to "delay-since-last-tone" counter
         _inbandDtmfGenerator.UpdateDelaySinceLastTone();
     }
@@ skipping 268 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt)
       != 0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc