Convert channel counts to size_t.

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.
  */
 
@@ skipping 26 matching lines @@
   virtual void OnWarningIsReported(const WarningCode warning) = 0;
 
  protected:
   virtual ~AudioDeviceObserver() {}
 };
 
 // ----------------------------------------------------------------------------
 //  AudioTransport
 // ----------------------------------------------------------------------------
 
 class AudioTransport {

[henrika_webrtc, 2015/12/29 09:00:57]

Note that this interface is used in Chrome and in Libjingle. I guess this change
might break things, and if so, is it really worth doing it?

See e.g.
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/webrtc...

https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjin...

and

https://code.google.com/p/chromium/codesearch#chromium/src/content/renderer/m...

and

https://code.google.com/p/chromium/codesearch#chromium/src/content/renderer/m...

  public:
   virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
                                           const size_t nSamples,
                                           const size_t nBytesPerSample,
-                                          const uint8_t nChannels,
+                                          const size_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 size_t nSamples,
                                    const size_t nBytesPerSample,
-                                   const uint8_t nChannels,
+                                   const size_t nChannels,
                                    const uint32_t samplesPerSec,
                                    void* audioSamples,
                                    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,
+                              size_t number_of_voe_channels,
                               const int16_t* audio_data,
                               int sample_rate,
-                              int number_of_channels,
+                              size_t number_of_channels,
                               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,
+                      size_t number_of_channels,
                       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,
+                               size_t number_of_channels,
                                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,
+                              size_t number_of_channels,

[henrika_webrtc, 2015/12/29 09:00:57]

I think this change will break Chrome here:

https://code.google.com/p/chromium/codesearch#chromium/src/content/renderer/m...

                               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.
   static const size_t kBitsPerSample = 16;
   AudioParameters()
       : sample_rate_(0),
         channels_(0),
         frames_per_buffer_(0),
         frames_per_10ms_buffer_(0) {}
-  AudioParameters(int sample_rate, int channels, size_t frames_per_buffer)
+  AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer)
       : sample_rate_(sample_rate),
         channels_(channels),
         frames_per_buffer_(frames_per_buffer),
         frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {}
-  void reset(int sample_rate, int channels, size_t frames_per_buffer) {
+  void reset(int sample_rate, size_t channels, size_t frames_per_buffer) {
     sample_rate_ = sample_rate;
     channels_ = channels;
     frames_per_buffer_ = frames_per_buffer;
     frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100);
   }
   size_t bits_per_sample() const { return kBitsPerSample; }
-  void reset(int sample_rate, int channels, double ms_per_buffer) {
+  void reset(int sample_rate, size_t channels, double ms_per_buffer) {
     reset(sample_rate, channels,
           static_cast<size_t>(sample_rate * ms_per_buffer + 0.5));
   }
-  void reset(int sample_rate, int channels) {
+  void reset(int sample_rate, size_t channels) {
     reset(sample_rate, channels, static_cast<size_t>(0));
   }
   int sample_rate() const { return sample_rate_; }
-  int channels() const { return channels_; }
+  size_t channels() const { return channels_; }
   size_t frames_per_buffer() const { return frames_per_buffer_; }
   size_t frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; }
   size_t GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; }
   size_t GetBytesPerBuffer() const {
     return frames_per_buffer_ * GetBytesPerFrame();
   }
   // The WebRTC audio device buffer (ADB) only requires that the sample rate
   // and number of channels are configured. Hence, to be "valid", only these
   // two attributes must be set.
   bool is_valid() const { return ((sample_rate_ > 0) && (channels_ > 0)); }
@@ skipping 10 matching lines @@
     return frames_per_buffer_ / (sample_rate_ / 1000.0);
   }
   double GetBufferSizeInSeconds() const {
     if (sample_rate_ == 0)
       return 0.0;
     return static_cast<double>(frames_per_buffer_) / (sample_rate_);
   }
 
  private:
   int sample_rate_;
-  int channels_;
+  size_t channels_;
   size_t frames_per_buffer_;
   size_t frames_per_10ms_buffer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_