Cleanup of iOS AudioDevice implementation

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 "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;
 
 // ----------------------------------------------------------------------------
 //  AudioDeviceObserver
 // ----------------------------------------------------------------------------
 
-class AudioDeviceObserver
-{
-public:
-    enum ErrorCode
-    {
-        kRecordingError = 0,
-        kPlayoutError = 1
-    };
-    enum WarningCode
-    {
-        kRecordingWarning = 0,
-        kPlayoutWarning = 1
-    };
+class AudioDeviceObserver {
+ public:
+  enum ErrorCode { kRecordingError = 0, kPlayoutError = 1 };
+  enum WarningCode { kRecordingWarning = 0, kPlayoutWarning = 1 };
 
-    virtual void OnErrorIsReported(const ErrorCode error) = 0;
-    virtual void OnWarningIsReported(const WarningCode warning) = 0;
+  virtual void OnErrorIsReported(const ErrorCode error) = 0;
+  virtual void OnWarningIsReported(const WarningCode warning) = 0;
 
-protected:
-    virtual ~AudioDeviceObserver() {}
+ protected:
+  virtual ~AudioDeviceObserver() {}
 };
 
 // ----------------------------------------------------------------------------
 //  AudioTransport
 // ----------------------------------------------------------------------------
 
-class AudioTransport
-{
-public:
-    virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
-                                            const uint32_t nSamples,
-                                            const uint8_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;
+class AudioTransport {
+ public:
+  virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                          const uint32_t nSamples,
+                                          const uint8_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,
-                                     const uint8_t nChannels,
-                                     const uint32_t samplesPerSec,
-                                     void* audioSamples,
-                                     uint32_t& nSamplesOut,
-                                     int64_t* elapsed_time_ms,
-                                     int64_t* ntp_time_ms) = 0;
+  virtual int32_t NeedMorePlayData(const uint32_t nSamples,
+                                   const uint8_t nBytesPerSample,
+                                   const uint8_t nChannels,
+                                   const uint32_t samplesPerSec,
+                                   void* audioSamples,
+                                   uint32_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,
-                                int audio_delay_milliseconds,
-                                int current_volume,
-                                bool key_pressed,
-                                bool need_audio_processing) { return 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,
+                              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) {}
+  // 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) {}
 
-    // 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) {}
+  // 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) {}
 
-    // 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,
-                                void* audio_data,
-                                int64_t* elapsed_time_ms,
-                                int64_t* ntp_time_ms) {}
+  // 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,
+                              void* audio_data,
+                              int64_t* elapsed_time_ms,
+                              int64_t* ntp_time_ms) {}
 
-protected:
-    virtual ~AudioTransport() {}
+ 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) {}
+  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);
+  }
+  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_; }
+  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 {
+    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_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H