webrtc/modules/audio_device/include/audio_device_defines.h - src - Git at Google

 /*
  *  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;

 // ----------------------------------------------------------------------------
 //  AudioDeviceObserver
 // ----------------------------------------------------------------------------

 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;

  protected:
   virtual ~AudioDeviceObserver() {}
 };

 // ----------------------------------------------------------------------------
 //  AudioTransport
 // ----------------------------------------------------------------------------

 class AudioTransport {
  public:
   virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
                                           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 size_t nSamples,
                                    const size_t nBytesPerSample,
                                    const uint8_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,
                               const int16_t* audio_data,
                               int sample_rate,
                               int 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_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_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_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)
       : 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) {
     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; }
   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();
   }
   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_;
   size_t frames_per_buffer_;
   size_t frames_per_10ms_buffer_;
 };

 }  // namespace webrtc

 #endif  // WEBRTC_AUDIO_DEVICE_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;

	// ----------------------------------------------------------------------------
	// AudioDeviceObserver
	// ----------------------------------------------------------------------------

	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;

	protected:
	virtual ~AudioDeviceObserver() {}
	};

	// ----------------------------------------------------------------------------
	// AudioTransport
	// ----------------------------------------------------------------------------

	class AudioTransport {
	public:
	virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
	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 size_t nSamples,
	const size_t nBytesPerSample,
	const uint8_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,
	const int16_t* audio_data,
	int sample_rate,
	int 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_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_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_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)
	: 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) {
	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; }
	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();
	}
	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_;
	size_t frames_per_buffer_;
	size_t frames_per_10ms_buffer_;
	};

	} // namespace webrtc

	#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H