Update audio code to use size_t more correctly, webrtc/modules/audio_device/

webrtc/modules/audio_device/include/audio_device_defines.h

 /*
  *  Copyright (c) 2011 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.
  */
 
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
 
+#include <stddef.h>
+
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 static const int kAdmMaxDeviceNameSize = 128;
 static const int kAdmMaxFileNameSize = 512;
 static const int kAdmMaxGuidSize = 128;
 
 static const int kAdmMinPlayoutBufferSizeMs = 10;
 static const int kAdmMaxPlayoutBufferSizeMs = 250;
@@ skipping 14 matching lines @@
   virtual ~AudioDeviceObserver() {}
 };
 
 // ----------------------------------------------------------------------------
 //  AudioTransport
 // ----------------------------------------------------------------------------
 
 class AudioTransport {
  public:
   virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
-                                          const uint32_t nSamples,
-                                          const uint8_t nBytesPerSample,
+                                          const size_t nSamples,
+                                          const size_t nBytesPerSample,
                                           const uint8_t nChannels,
                                           const uint32_t samplesPerSec,
                                           const uint32_t totalDelayMS,
                                           const int32_t clockDrift,
                                           const uint32_t currentMicLevel,
                                           const bool keyPressed,
                                           uint32_t& newMicLevel) = 0;
 
-  virtual int32_t NeedMorePlayData(const uint32_t nSamples,
-                                   const uint8_t nBytesPerSample,
+  virtual int32_t NeedMorePlayData(const size_t nSamples,
+                                   const size_t nBytesPerSample,
                                    const uint8_t nChannels,
                                    const uint32_t samplesPerSec,
                                    void* audioSamples,
-                                   uint32_t& nSamplesOut,
+                                   size_t& nSamplesOut,
                                    int64_t* elapsed_time_ms,
                                    int64_t* ntp_time_ms) = 0;
 
   // Method to pass captured data directly and unmixed to network channels.
   // |channel_ids| contains a list of VoE channels which are the
   // sinks to the capture data. |audio_delay_milliseconds| is the sum of
   // recording delay and playout delay of the hardware. |current_volume| is
   // in the range of [0, 255], representing the current microphone analog
   // volume. |key_pressed| is used by the typing detection.
   // |need_audio_processing| specify if the data needs to be processed by APM.
   // Currently WebRtc supports only one APM, and Chrome will make sure only
   // one stream goes through APM. When |need_audio_processing| is false, the
   // values of |audio_delay_milliseconds|, |current_volume| and |key_pressed|
   // will be ignored.
   // The return value is the new microphone volume, in the range of |0, 255].
   // When the volume does not need to be updated, it returns 0.
   // TODO(xians): Remove this interface after Chrome and Libjingle switches
   // to OnData().
   virtual int OnDataAvailable(const int voe_channels[],
                               int number_of_voe_channels,
                               const int16_t* audio_data,
                               int sample_rate,
                               int number_of_channels,
-                              int number_of_frames,
+                              size_t number_of_frames,
                               int audio_delay_milliseconds,
                               int current_volume,
                               bool key_pressed,
                               bool need_audio_processing) {
     return 0;
   }
 
   // Method to pass the captured audio data to the specific VoE channel.
   // |voe_channel| is the id of the VoE channel which is the sink to the
   // capture data.
   // TODO(xians): Remove this interface after Libjingle switches to
   // PushCaptureData().
   virtual void OnData(int voe_channel,
                       const void* audio_data,
                       int bits_per_sample,
                       int sample_rate,
                       int number_of_channels,
-                      int number_of_frames) {}
+                      size_t number_of_frames) {}
 
   // Method to push the captured audio data to the specific VoE channel.
   // The data will not undergo audio processing.
   // |voe_channel| is the id of the VoE channel which is the sink to the
   // capture data.
   // TODO(xians): Make the interface pure virtual after Libjingle
   // has its implementation.
   virtual void PushCaptureData(int voe_channel,
                                const void* audio_data,
                                int bits_per_sample,
                                int sample_rate,
                                int number_of_channels,
-                               int number_of_frames) {}
+                               size_t number_of_frames) {}
 
   // Method to pull mixed render audio data from all active VoE channels.
   // The data will not be passed as reference for audio processing internally.
   // TODO(xians): Support getting the unmixed render data from specific VoE
   // channel.
   virtual void PullRenderData(int bits_per_sample,
                               int sample_rate,
                               int number_of_channels,
-                              int number_of_frames,
+                              size_t number_of_frames,
                               void* audio_data,
                               int64_t* elapsed_time_ms,
                               int64_t* ntp_time_ms) {}
 
  protected:
   virtual ~AudioTransport() {}
 };
 
 // Helper class for storage of fundamental audio parameters such as sample rate,
 // number of channels, native buffer size etc.
 // Note that one audio frame can contain more than one channel sample and each
 // sample is assumed to be a 16-bit PCM sample. Hence, one audio frame in
 // stereo contains 2 * (16/8) = 4 bytes of data.
 class AudioParameters {
  public:
   // This implementation does only support 16-bit PCM samples.
   enum { kBitsPerSample = 16 };
   AudioParameters()
       : sample_rate_(0),
         channels_(0),
         frames_per_buffer_(0),
         frames_per_10ms_buffer_(0) {}
   AudioParameters(int sample_rate, int channels, int frames_per_buffer)
       : sample_rate_(sample_rate),
         channels_(channels),
         frames_per_buffer_(frames_per_buffer),
-        frames_per_10ms_buffer_(sample_rate / 100) {}
+        frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {}
   void reset(int sample_rate, int channels, int frames_per_buffer) {
     sample_rate_ = sample_rate;
     channels_ = channels;
     frames_per_buffer_ = frames_per_buffer;
-    frames_per_10ms_buffer_ = (sample_rate / 100);
+    frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100);
   }
   int bits_per_sample() const { return kBitsPerSample; }
   int sample_rate() const { return sample_rate_; }
   int channels() const { return channels_; }
   int frames_per_buffer() const { return frames_per_buffer_; }
-  int frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; }
+  size_t frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; }
   bool is_valid() const {
     return ((sample_rate_ > 0) && (channels_ > 0) && (frames_per_buffer_ > 0));
   }
   int GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; }
   int GetBytesPerBuffer() const {
     return frames_per_buffer_ * GetBytesPerFrame();
   }
-  int GetBytesPer10msBuffer() const {
+  size_t GetBytesPer10msBuffer() const {
     return frames_per_10ms_buffer_ * GetBytesPerFrame();
   }
   float GetBufferSizeInMilliseconds() const {
     if (sample_rate_ == 0)
       return 0.0f;
     return frames_per_buffer_ / (sample_rate_ / 1000.0f);
   }
 
  private:
   int sample_rate_;
   int channels_;
   int frames_per_buffer_;
-  int frames_per_10ms_buffer_;
+  size_t frames_per_10ms_buffer_;

[henrika_webrtc, 2015/07/24 10:43:04]

I have asked a similar question before but it is not clear to me why
frames_per_10ms_buffer_ should be size_t but not frames_per_buffer_. What is the
difference?

[Peter Kasting, 2015/07/27 23:28:02]

The answer is that fixing |frames_per_10ms_buffer_| was required to get the rest
of my CL to compile whereas fixing |frames_per_buffer_| wasn't.

In principle, both should be size_t.  This just doesn't bother to convert the
other variable yet.  I can stick that in my set of things to convert in the next
pass.

 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H