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

webrtc/voice_engine/channel.h

 /*
  *  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 414 matching lines @@
     int8_t OutputEnergyLevel() const
     {
         return _outputAudioLevel.Level();
     }
     uint32_t Demultiplex(const AudioFrame& audioFrame);
     // Demultiplex the data to the channel's |_audioFrame|. The difference
     // between this method and the overloaded method above is that |audio_data|
     // does not go through transmit_mixer and APM.
     void Demultiplex(const int16_t* audio_data,
                      int sample_rate,
-                     int number_of_frames,
+                     size_t number_of_frames,

[henrika_webrtc, 2015/07/13 07:38:08]

It is not clear to me why size_t is required here. Can you provide some more
general background on why this sort of changes are done. I am sure that all is
OK; just want to understand and learn. Thanks.

[Peter Kasting, 2015/07/13 22:12:28]

Sure.  A fairly complete argument/set of tradeoffs can be found in the design
doc I wrote for a previous "use size_t more" change; see
https://docs.google.com/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZy...
.

If you want a short summary, basically the rule is "use size_t for object sizes
or lengths of containers in memory".  This matches how the STL works, is
approved by the Google style guide, promotes consistency of type usage, and can
sometimes simplify or clarify code by eliminating the possibility of certain
variables taking on negative values.

In this example, the actual frames are stored in memory, so we use a size_t to
count them.

[henrika_webrtc, 2015/07/14 07:54:09]

Thanks!

                      int number_of_channels);
     uint32_t PrepareEncodeAndSend(int mixingFrequency);
     uint32_t EncodeAndSend();
 
     // Associate to a send channel.
     // Used for obtaining RTT for a receive-only channel.
     void set_associate_send_channel(const ChannelOwner& channel) {
       assert(_channelId != channel.channel()->ChannelId());
       CriticalSectionScoped lock(assoc_send_channel_lock_.get());
       associate_send_channel_ = channel;
@@ skipping 132 matching lines @@
     rtc::scoped_ptr<NetworkPredictor> network_predictor_;
     // An associated send channel.
     rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
     ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_