Refactor the AudioDevice for iOS and improve the performance and stability

webrtc/modules/audio_device/include/audio_device_defines.h

Index: webrtc/modules/audio_device/include/audio_device_defines.h
diff --git a/webrtc/modules/audio_device/include/audio_device_defines.h b/webrtc/modules/audio_device/include/audio_device_defines.h
index a14c77e4c6074a0166e7873b34bcc923fa9274e7..59f86c2156541ad25aa7ac6e29eefa2b841bcd03 100644
--- a/webrtc/modules/audio_device/include/audio_device_defines.h
+++ b/webrtc/modules/audio_device/include/audio_device_defines.h
@@ -8,8 +8,8 @@
  *  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
+#ifndef WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
+#define WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
 
 #include <stddef.h>
 
@@ -161,24 +161,41 @@ class AudioParameters {
     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 milliseconds_per_buffer) {

[tkchin_webrtc, 2015/09/01 20:54:50]

nit: ms_per_buffer is prob ok

[henrika_webrtc, 2015/09/03 13:44:41]

Done.

+    reset(sample_rate, channels,
+          static_cast<size_t>(sample_rate * milliseconds_per_buffer + 0.5));
+  }
+  void reset(int sample_rate, int channels) {
+    reset(sample_rate, channels, static_cast<size_t>(0));

[tkchin_webrtc, 2015/09/01 20:54:50]

nit: 0u?

[henrika_webrtc, 2015/09/03 13:44:41]

pkasting just landed this static_cast in a CL where he cleaned up how we use
size_t. Hope it is OK if I keep it as is since it was his choice to use this
method.

+  }
   int sample_rate() const { return sample_rate_; }
   int 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_; }
-  bool is_valid() const {
-    return ((sample_rate_ > 0) && (channels_ > 0) && (frames_per_buffer_ > 0));
-  }
   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)); }
+  // Most platforms also require that a native buffer size is defined.
+  // An audio parameter instance is considered to be "complete" if it is both
+  // "valid" (can be used by the ADB) and also has a native frame size.
+  bool is_complete() const { return (is_valid() && (frames_per_buffer_ > 0)); }

[tkchin_webrtc, 2015/09/01 20:54:50]

When is it useful to be valid but incomplete?

[henrika_webrtc, 2015/09/03 13:44:41]

Good question. Perhaps the current notation is not perfect but with a set that
is valid, you can call AudioDeviceIOS::UpdateAudioDeviceBuffer() which informs
the upper layers (WebRTC running at 10ms) about the sample rate and the number
of channels. In this state one can start to ask for audio from WebRTC and to
send recorded data to it.
But, the AudioParameters struct is not yet complete if it does not contain the
internal/native buffer size as well. This parts is only used by the lower layers
but I found it handy to add an extra check for it. Hence, 'complete' means
'valid for WebRTC communication' and possibility to kick of the native parts as
well. 

   size_t GetBytesPer10msBuffer() const {
     return frames_per_10ms_buffer_ * GetBytesPerFrame();
   }
-  float GetBufferSizeInMilliseconds() const {
+  double GetBufferSizeInMilliseconds() const {
+    if (sample_rate_ == 0)
+      return 0.0;
+    return frames_per_buffer_ / (sample_rate_ / 1000.0);
+  }
+  double GetBufferSizeInSeconds() const {
     if (sample_rate_ == 0)
-      return 0.0f;
-    return frames_per_buffer_ / (sample_rate_ / 1000.0f);
+      return 0.0;
+    return static_cast<double>(frames_per_buffer_) / (sample_rate_);
   }
 
  private:
@@ -190,4 +207,4 @@ class AudioParameters {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
+#endif  // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_