Make AudioParameters a class instead of a struct

content/renderer/media/webrtc_audio_device_impl.h

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #pragma once
 
 #include <string>
 #include <vector>
@@ skipping 236 matching lines @@
   virtual int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const OVERRIDE;
 
   virtual int32_t ResetAudioDevice() OVERRIDE;
   virtual int32_t SetLoudspeakerStatus(bool enable) OVERRIDE;
   virtual int32_t GetLoudspeakerStatus(bool* enabled) const OVERRIDE;
 
   // Sets the session id.
   void SetSessionId(int session_id);
 
   // Accessors.
-  size_t input_buffer_size() const { return input_buffer_size_; }
-  size_t output_buffer_size() const { return output_buffer_size_; }
-  int input_channels() const { return input_channels_; }
-  int output_channels() const { return output_channels_; }
-  int input_sample_rate() const { return static_cast<int>(input_sample_rate_); }
+  size_t input_buffer_size() const {
+    return input_audio_parameters_.samples_per_packet();
+  }
+  size_t output_buffer_size() const {
+    return input_audio_parameters_.samples_per_packet();
+  }
+  int input_channels() const {
+    return input_audio_parameters_.channels();
+  }
+  int output_channels() const {
+    return output_audio_parameters_.channels();
+  }
+  int input_sample_rate() const {
+    return static_cast<int>(input_audio_parameters_.samples_per_second());

[tommi (sloooow) - chröme, 2012/03/10 10:11:32]

remove the cast?

[vrk (LEFT CHROMIUM), 2012/03/16 18:30:41]

Good catch! Done.

+  }
   int output_sample_rate() const {
-    return static_cast<int>(output_sample_rate_);
+    return static_cast<int>(output_audio_parameters_.samples_per_second());

[tommi (sloooow) - chröme, 2012/03/10 10:11:32]

remove the cast?

[vrk (LEFT CHROMIUM), 2012/03/16 18:30:41]

Done.

   }
   int input_delay_ms() const { return input_delay_ms_; }
   int output_delay_ms() const { return output_delay_ms_; }
   bool initialized() const { return initialized_; }
   bool playing() const { return playing_; }
   bool recording() const { return recording_; }
 
  private:
   // Make destructor private to ensure that we can only be deleted by Release().
   virtual ~WebRtcAudioDeviceImpl();
@@ skipping 14 matching lines @@
 
   // Provides access to the native audio output layer in the browser process.
   scoped_refptr<AudioDevice> audio_output_device_;
 
   // Weak reference to the audio callback.
   // The webrtc client defines |audio_transport_callback_| by calling
   // RegisterAudioCallback().
   webrtc::AudioTransport* audio_transport_callback_;
 
   // Cached values of utilized audio parameters. Platform dependent.
-  size_t input_buffer_size_;
-  size_t output_buffer_size_;
-  int input_channels_;
-  int output_channels_;
-  double input_sample_rate_;
-  double output_sample_rate_;
+  AudioParameters input_audio_parameters_;
+  AudioParameters output_audio_parameters_;

[tommi (sloooow) - chröme, 2012/03/10 10:11:32]

nice!

[vrk (LEFT CHROMIUM), 2012/03/16 18:30:41]

:)

 
   // Cached value of the current audio delay on the input/capture side.
   int input_delay_ms_;
 
   // Cached value of the current audio delay on the output/renderer side.
   int output_delay_ms_;
 
   // Buffers used for temporary storage during capture/render callbacks.
   // Allocated during initialization to save stack.
   scoped_array<int16> input_buffer_;
@@ skipping 13 matching lines @@
   int bytes_per_sample_;
 
   bool initialized_;
   bool playing_;
   bool recording_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_